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[alsa-devel] [PATCH 0/8] Fixes for HDSPM, especially AES(32)
by Adrian Knoth 23 Feb '11

23 Feb '11

Hi! Florian Faber, Fredrik Lingvall and me have gone through an e-mail debug session during the past few weeks. Here's our result containing several fixes for RME cards. Cheers Adrian Knoth (8): ALSA: hpdsm - RME AES(32): Fix missing channel mappings ALSA: hdspm - Fix buffer handling on RME MADI/MADIface/AES(32) ALSA: hdspm - Restrict channel count on RME AES/AES32 ALSA: hdspm - Fix DS/QS output channel mappings on RME MADI/MADIface ALSA: hdspm - Remove input selector on MADIface ALSA: hdspm - fix sync check on AES32 ALSA: hdspm - prevent reading unitialized stack memory ALSA: hdspm - Fix lock/sync reporting on MADI and AES32 sound/pci/rme9652/hdspm.c | 130 +++++++++++++++++++++++++++++++++++--------- 1 files changed, 103 insertions(+), 27 deletions(-) -- 1.7.2.3

2 9

[alsa-devel] [PATCH 2/2] ALSA: HDA: Add ideapad quirk for two Dell machines
by David Henningsson 23 Feb '11

23 Feb '11

-- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] [PATCH 1/2] ALSA: HDA: Add a new Conexant codec 506e (20590)
by David Henningsson 23 Feb '11

23 Feb '11

I'm a little hesitant as of whether to send this one to stable or not, as the fallback of the 5066 parser can perform worse than the generic one. However, if we don't, we can't quirk them to work perfectly later. What do you think? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[PATCH 2/2] ALSA: HDA: Add ideapad quirk for two Dell machines
by David Henningsson 23 Feb '11

23 Feb '11

These two Dell machines have been reported working well with the ideapad model. BugLink: http://bugs.launchpad.net/bugs/723676 Cc: stable(a)kernel.org Tested-by: David Chen <david.chen(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 909ce9e..4d5004e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3114,6 +3114,8 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS), SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), -- 1.7.1 --------------070408030205050402080201--

1 0

Re: [alsa-devel] [PATCH] USB-sound: prevent kernel panic on disconnect
by Clemens Ladisch 23 Feb '11

23 Feb '11

Alan Stern wrote: > On Fri, 11 Feb 2011, Sarah Sharp wrote: > > I suspect it might be the audio driver, as someone on the alsa mailing > > list suggested the USB sound driver can't handle when one isochronous > > buffer in an URB has an error, but the URB status is 0. > > I don't think that can happen. The URB status won't be 0 unless all > the individual buffers have 0 status. In any case, the driver would then just copy garbage samples out of the buffer; this wouldn't affect the driver's data structures. > > From: Pierre-Louis Bossart <bossart.nospam(a)gmail.com> > > > > Note that this is only a work-around, it does not address the > > root cause of this inconsistency between urbs and PCM states. The > > dmesg below shows two calls to snd_urb_complete, the substream is > > NULL and the state is either running or stopped. This doesn't make > > any sense. > > ... > > ALSA urb.c:492: frame 0 active: -84 > > ALSA urb.c:197: cannot submit urb (err = -19) > > ALSA urb.c:186: NULL substream (subs->running 1) <- How is this possible? > > ALSA urb.c:186: NULL substream (subs->running 0) > > It's most likely a matter of the device being disconnected but the > device file still being open. subs->pcm_substream == NULL happens only when the device file _has_ been closed. > Perhaps not everything gets cleaned up the way it should when that > happens. There seems to be a race between snd_usb_pcm_close() and snd_usb_stream_disconnect(). I think a mutex taken by both functions should fix this; and all functions that check 'shutdown' probably need serializing. Regards, Clemens

4 11

[alsa-devel] [PATCH v4] ASoC: Add Freescale SGTL5000 codec support
by zhaoming.zeng＠freescale.com 23 Feb '11

23 Feb '11

From: Zeng Zhaoming <zhaoming.zeng(a)freescale.com> Add Freescale SGTL5000 codec support. Supported features: - line-in and mic input - headphone and line-out output - line-in bypass ADC and DAC to headphone - 16, 20, 24, 32 bit audio - 8 ~ 96k sample rates Signed-off-by: Zeng Zhaoming <zhaoming.zeng(a)freescale.com> --- changes since v3: 1. Rebase to sound-2.6 git's for-2.6.39 branch 2. Add feature discription in git commit message. 3. Change sgtl5000.h license to GPLv2 4. Change to using clamp() in custom pcm volume get() and put() routines 5. Change to using snd_soc_write() instead of snd_soc_update_bits_lock() in put(). 6. Add power down event to mic bias. 7. Define Capture Attenuate Switch as a TLV control 8. Implement an internal vddd regulator and enable it when vddd not provided externally. 9. Some style modification to make the code clearer. 10.Not modify memcpy((void *)) because my toolchain (gcc 4.4.4) complains about the type mismatch. Thanks for review of Arnaud, Mark and Timur. --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sgtl5000.c | 1491 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sgtl5000.h | 402 ++++++++++++ 4 files changed, 1900 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e239345..c04da18 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 + select SND_SOC_SGTL5000 if I2C select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C @@ -182,6 +183,10 @@ config SND_SOC_MAX98088 config SND_SOC_PCM3008 tristate +#Freescale SGTL5000 codec +config SND_SOC_SGTL5000 + tristate + config SND_SOC_SN95031 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ae10507..3bbb08c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -19,6 +19,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o @@ -103,6 +104,7 @@ obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c new file mode 100644 index 0000000..fe51764 --- /dev/null +++ b/sound/soc/codecs/sgtl5000.c @@ -0,0 +1,1491 @@ +/* + * sgtl5000.c -- SGTL5000 ALSA SoC Audio driver + * + * Copyright 2010-2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <linux/regulator/driver.h> +#include <linux/regulator/machine.h> +#include <linux/regulator/consumer.h> +#include <sound/core.h> +#include <sound/tlv.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <mach/hardware.h> + +#include "sgtl5000.h" + +#define SGTL5000_DAP_REG_OFFSET 0x0100 +#define SGTL5000_MAX_REG_OFFSET 0x013A + +/* default value of sgtl5000 registers except DAP */ +static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] = { + 0xa011, /* 0x0000, CHIP_ID. 11 stand for revison 17 */ + 0x0000, /* 0x0002, CHIP_DIG_POWER. */ + 0x0008, /* 0x0004, CHIP_CKL_CTRL */ + 0x0010, /* 0x0006, CHIP_I2S_CTRL */ + 0x0000, /* 0x0008, reserved */ + 0x0008, /* 0x000A, CHIP_SSS_CTRL */ + 0x0000, /* 0x000C, reserved */ + 0x020c, /* 0x000E, CHIP_ADCDAC_CTRL */ + 0x3c3c, /* 0x0010, CHIP_DAC_VOL */ + 0x0000, /* 0x0012, reserved */ + 0x015f, /* 0x0014, CHIP_PAD_STRENGTH */ + 0x0000, /* 0x0016, reserved */ + 0x0000, /* 0x0018, reserved */ + 0x0000, /* 0x001A, reserved */ + 0x0000, /* 0x001E, reserved */ + 0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */ + 0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */ + 0x0111, /* 0x0024, CHIP_ANN_CTRL */ + 0x0000, /* 0x0026, CHIP_LINREG_CTRL */ + 0x0000, /* 0x0028, CHIP_REF_CTRL */ + 0x0000, /* 0x002A, CHIP_MIC_CTRL */ + 0x0000, /* 0x002C, CHIP_LINE_OUT_CTRL */ + 0x0404, /* 0x002E, CHIP_LINE_OUT_VOL */ + 0x7060, /* 0x0030, CHIP_ANA_POWER */ + 0x5000, /* 0x0032, CHIP_PLL_CTRL */ + 0x0000, /* 0x0034, CHIP_CLK_TOP_CTRL */ + 0x0000, /* 0x0036, CHIP_ANA_STATUS */ + 0x0000, /* 0x0038, reserved */ + 0x0000, /* 0x003A, CHIP_ANA_TEST2 */ + 0x0000, /* 0x003C, CHIP_SHORT_CTRL */ + 0x0000, /* reserved */ +}; + +/* default value of dap registers */ +static const u16 sgtl5000_dap_regs[] = { + 0x0000, /* 0x0100, DAP_CONTROL */ + 0x0000, /* 0x0102, DAP_PEQ */ + 0x0040, /* 0x0104, DAP_BASS_ENHANCE */ + 0x051f, /* 0x0106, DAP_BASS_ENHANCE_CTRL */ + 0x0000, /* 0x0108, DAP_AUDIO_EQ */ + 0x0040, /* 0x010A, DAP_SGTL_SURROUND */ + 0x0000, /* 0x010C, DAP_FILTER_COEF_ACCESS */ + 0x0000, /* 0x010E, DAP_COEF_WR_B0_MSB */ + 0x0000, /* 0x0110, DAP_COEF_WR_B0_LSB */ + 0x0000, /* 0x0112, reserved */ + 0x0000, /* 0x0114, reserved */ + 0x002f, /* 0x0116, DAP_AUDIO_EQ_BASS_BAND0 */ + 0x002f, /* 0x0118, DAP_AUDIO_EQ_BAND0 */ + 0x002f, /* 0x011A, DAP_AUDIO_EQ_BAND2 */ + 0x002f, /* 0x011C, DAP_AUDIO_EQ_BAND3 */ + 0x002f, /* 0x011E, DAP_AUDIO_EQ_TREBLE_BAND4 */ + 0x8000, /* 0x0120, DAP_MAIN_CHAN */ + 0x0000, /* 0x0122, DAP_MIX_CHAN */ + 0x0510, /* 0x0124, DAP_AVC_CTRL */ + 0x1473, /* 0x0126, DAP_AVC_THRESHOLD */ + 0x0028, /* 0x0128, DAP_AVC_ATTACK */ + 0x0050, /* 0x012A, DAP_AVC_DECAY */ + 0x0000, /* 0x012C, DAP_COEF_WR_B1_MSB */ + 0x0000, /* 0x012E, DAP_COEF_WR_B1_LSB */ + 0x0000, /* 0x0130, DAP_COEF_WR_B2_MSB */ + 0x0000, /* 0x0132, DAP_COEF_WR_B2_LSB */ + 0x0000, /* 0x0134, DAP_COEF_WR_A1_MSB */ + 0x0000, /* 0x0136, DAP_COEF_WR_A1_LSB */ + 0x0000, /* 0x0138, DAP_COEF_WR_A2_MSB */ + 0x0000, /* 0x013A, DAP_COEF_WR_A2_LSB */ +}; + +/* regulator supplies for sgtl5000, VDDD is an option external supply */ +enum sgtl5000_regulator_supplies { + VDDA, + VDDIO, + VDDD, + SGTL5000_SUPPLY_NUM +}; + +/* vddd is optinal supply */ +static const char *supply_names[SGTL5000_SUPPLY_NUM] = { + "VDDA", + "VDDIO", + "VDDD" +}; + +#define LDO_CONSUMER_NAME "VDDD_LDO" +#define LDO_VOLTAGE 1200000 + +static struct regulator_consumer_supply ldo_consumer[] = { + REGULATOR_SUPPLY(LDO_CONSUMER_NAME, NULL), +}; + +struct regulator_init_data ldo_init_data = { + .constraints = { + .min_uV = 850000, + .max_uV = 1600000, + .valid_modes_mask = REGULATOR_MODE_NORMAL, + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, + .num_consumer_supplies = 1, + .consumer_supplies = &ldo_consumer[0], +}; + +/* + * sgtl5000 internal ldo regulator, + * enabled when VDDD not provided + */ +struct ldo_regulator { + struct regulator_desc desc; + struct regulator_dev *dev; + int voltage; + void *codec_data; + bool enabled; +}; + +/* sgtl5000 private structure in codec */ +struct sgtl5000_priv { + int sysclk; /* sysclk rate */ + int master; /* i2s master or not */ + int fmt; /* i2s data format */ + struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM]; + struct ldo_regulator *ldo; +}; + +/* + * mic_bias power on/off share the same register bits with + * output impedance of mic bias, when power on mic bias, we + * need reclaim it to impedance value. + * 0x0 = Powered off + * 0x1 = 2Kohm + * 0x2 = 4Kohm + * 0x3 = 8Kohm + */ +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* change mic bias resistor to 4Kohm */ + snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_4k, SGTL5000_BIAS_R_4k); + break; + + case SND_SOC_DAPM_PRE_PMD: + /* + * SGTL5000_BIAS_R_8k as mask to clean the two bits + * of mic bias and output impedance + */ + snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_8k, 0); + break; + } + return 0; +} + +/* + * using codec assist to small pop, hp_powerup or lineout_powerup + * should stay setting until vag_powerup is fully ramped down, + * vag fully ramped down require 400ms. + */ +static int small_pop_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, 0); + msleep(400); + break; + default: + break; + } + + return 0; +} + +/* input sources for ADC */ +static const char *adc_mux_text[] = { + "MIC_IN", "LINE_IN" +}; + +static const struct soc_enum adc_enum = +SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 2, 2, adc_mux_text); + +static const struct snd_kcontrol_new adc_mux = +SOC_DAPM_ENUM("Capture Mux", adc_enum); + +/* input sources for DAC */ +static const char *dac_mux_text[] = { + "DAC", "LINE_IN" +}; + +static const struct soc_enum dac_enum = +SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 6, 2, dac_mux_text); + +static const struct snd_kcontrol_new dac_mux = +SOC_DAPM_ENUM("Headphone Mux", dac_enum); + +static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINE_IN"), + SND_SOC_DAPM_INPUT("MIC_IN"), + + SND_SOC_DAPM_OUTPUT("HP_OUT"), + SND_SOC_DAPM_OUTPUT("LINE_OUT"), + + SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0, + small_pop_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0, + small_pop_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), + SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), + + /* aif for i2s input */ + SND_SOC_DAPM_AIF_IN("AIFIN", "Playback", + 0, SGTL5000_CHIP_DIG_POWER, + 0, 0), + + /* aif for i2s output */ + SND_SOC_DAPM_AIF_OUT("AIFOUT", "Capture", + 0, SGTL5000_CHIP_DIG_POWER, + 1, 0), + + SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), + + SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), +}; + +/* routes for sgtl5000 */ +static const struct snd_soc_dapm_route audio_map[] = { + {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ + {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ + + {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ + {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ + + {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ + {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ + {"LO", NULL, "DAC"}, /* dac --> line_out */ + + {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ + {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ + + {"LINE_OUT", NULL, "LO"}, + {"HP_OUT", NULL, "HP"}, +}; + +/* custom function to fetch info of PCM playback volume */ +static int dac_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xfc - 0x3c; + return 0; +} + +/* + * custom function to get of PCM playback volume + * + * dac volume register + * 15-------------8-7--------------0 + * | R channel vol | L channel vol | + * ------------------------------- + * + * PCM volume with 0.5017 dB steps from 0 to -90 dB + * + * register values map to dB + * 0x3B and less = Reserved + * 0x3C = 0 dB + * 0x3D = -0.5 dB + * 0xF0 = -90 dB + * 0xFC and greater = Muted + * + * register value map to userspace value + * + * register value 0x3c(0dB) 0xf0(-90dB)0xfc + * ------------------------------ + * userspace value 0xc0 0 + */ +static int dac_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg, l, r; + + reg = snd_soc_read(codec, SGTL5000_CHIP_DAC_VOL); + + /* get left channel volume */ + l = (reg & SGTL5000_DAC_VOL_LEFT_MASK) >> SGTL5000_DAC_VOL_LEFT_SHIFT; + + /* get right channel volume */ + r = (reg & SGTL5000_DAC_VOL_RIGHT_MASK) >> SGTL5000_DAC_VOL_RIGHT_SHIFT; + + /* make sure value fall in (0x3c,0xfc) */ + l = clamp(l, 0x3c, 0xfc); + r = clamp(r, 0x3c, 0xfc); + + /* invert it and map to userspace value */ + l = 0xfc - l; + r = 0xfc - r; + + ucontrol->value.integer.value[0] = l; + ucontrol->value.integer.value[1] = r; + + return 0; +} + +/* + * custom function to put of PCM playback volume + * + * dac volume register + * 15-------------8-7--------------0 + * | R channel vol | L channel vol | + * ------------------------------- + * + * PCM volume with 0.5017 dB steps from 0 to -90 dB + * + * register values map to dB + * 0x3B and less = Reserved + * 0x3C = 0 dB + * 0x3D = -0.5 dB + * 0xF0 = -90 dB + * 0xFC and greater = Muted + * + * userspace value map to register value + * + * userspace value 0xc0 0 + * ------------------------------ + * register value 0x3c(0dB) 0xf0(-90dB)0xfc + */ +static int dac_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg, l, r; + + l = ucontrol->value.integer.value[0]; + r = ucontrol->value.integer.value[1]; + + /* make sure userspace volume fall in (0, 0xfc-0x3c) */ + l = clamp(l, 0, 0xfc - 0x3c); + r = clamp(r, 0, 0xfc - 0x3c); + + /* invert it, get the value can be set to register */ + l = 0xfc - l; + r = 0xfc - r; + + /* shift to get the register value */ + reg = l << SGTL5000_DAC_VOL_LEFT_SHIFT | + r << SGTL5000_DAC_VOL_RIGHT_SHIFT; + + snd_soc_write(codec, SGTL5000_CHIP_DAC_VOL, reg); + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0); + +/* tlv for mic gain, 0db 20db 30db 40db */ +static const unsigned int mic_gain_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0), +}; + +/* tlv for hp volume, -51.5db to 12.0db, step .5db */ +static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); + +static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { + /* SOC_DOUBLE_S8_TLV with invert */ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = dac_info_volsw, + .get = dac_get_volsw, + .put = dac_put_volsw, + }, + + SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0), + SOC_SINGLE_TLV("Capture Attenuate Switch (-6dB)", + SGTL5000_CHIP_ANA_ADC_CTRL, + 8, 2, 0, capture_6db_attenuate), + SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0), + + SOC_DOUBLE_TLV("Headphone Playback Volume", + SGTL5000_CHIP_ANA_HP_CTRL, + 0, 8, + 0x7f, 1, + headphone_volume), + SOC_SINGLE("Headphone Playback ZC Switch", SGTL5000_CHIP_ANA_CTRL, + 5, 1, 0), + + SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, + 0, 4, 0, mic_gain_tlv), +}; + +/* mute the codec used by alsa core */ +static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 adcdac_ctrl = SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT; + + snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, + adcdac_ctrl, mute ? adcdac_ctrl : 0); + + return 0; +} + +/* set codec format */ +static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + u16 i2sctl = 0; + + sgtl5000->master = 0; + /* + * i2s clock and frame master setting. + * ONLY support: + * - clock and frame slave, + * - clock and frame master + */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBM_CFM: + i2sctl |= SGTL5000_I2S_MASTER; + sgtl5000->master = 1; + break; + default: + return -EINVAL; + } + + /* setting i2s data format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + i2sctl |= SGTL5000_I2S_MODE_PCM; + break; + case SND_SOC_DAIFMT_DSP_B: + i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_LRALIGN; + break; + case SND_SOC_DAIFMT_I2S: + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + break; + case SND_SOC_DAIFMT_RIGHT_J: + i2sctl |= SGTL5000_I2S_MODE_RJ; + i2sctl |= SGTL5000_I2S_LRPOL; + break; + case SND_SOC_DAIFMT_LEFT_J: + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_LRALIGN; + break; + default: + return -EINVAL; + } + + sgtl5000->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + /* Clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + i2sctl |= SGTL5000_I2S_SCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, SGTL5000_CHIP_I2S_CTRL, i2sctl); + + return 0; +} + +/* set codec sysclk */ +static int sgtl5000_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case SGTL5000_SYSCLK: + sgtl5000->sysclk = freq; + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * set clock according to i2s frame clock, + * sgtl5000 provide 2 clock sources. + * 1. sys_mclk. sample freq can only configure to + * 1/256, 1/384, 1/512 of sys_mclk. + * 2. pll. can derive any audio clocks. + * + * clock setting rules: + * 1. in slave mode, only sys_mclk can use. + * 2. as constraint by sys_mclk, sample freq should + * set to 32k, 44.1k and above. + * 3. using sys_mclk prefer to pll to save power. + */ +static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int clk_ctl = 0; + int sys_fs; /* sample freq */ + + /* + * sample freq should be divided by frame clock, + * if frame clock lower than 44.1khz, sample feq should set to + * 32khz or 44.1khz. + */ + switch (frame_rate) { + case 8000: + case 16000: + sys_fs = 32000; + break; + case 11025: + case 22050: + sys_fs = 44100; + break; + default: + sys_fs = frame_rate; + break; + } + + /* set devided factor of frame clock */ + switch (sys_fs / frame_rate) { + case 4: + clk_ctl |= SGTL5000_RATE_MODE_DIV_4 << SGTL5000_RATE_MODE_SHIFT; + break; + case 2: + clk_ctl |= SGTL5000_RATE_MODE_DIV_2 << SGTL5000_RATE_MODE_SHIFT; + break; + case 1: + clk_ctl |= SGTL5000_RATE_MODE_DIV_1 << SGTL5000_RATE_MODE_SHIFT; + break; + default: + return -EINVAL; + } + + /* set the sys_fs accroding to frame rate */ + switch (sys_fs) { + case 32000: + clk_ctl |= SGTL5000_SYS_FS_32k << SGTL5000_SYS_FS_SHIFT; + break; + case 44100: + clk_ctl |= SGTL5000_SYS_FS_44_1k << SGTL5000_SYS_FS_SHIFT; + break; + case 48000: + clk_ctl |= SGTL5000_SYS_FS_48k << SGTL5000_SYS_FS_SHIFT; + break; + case 96000: + clk_ctl |= SGTL5000_SYS_FS_96k << SGTL5000_SYS_FS_SHIFT; + break; + default: + dev_err(codec->dev, "frame rate %d not supported\n", + frame_rate); + return -EINVAL; + } + + /* + * calculate the divider of mclk/sample_freq, + * factor of freq =96k can only be 256, since mclk in range (12m,27m) + */ + switch (sgtl5000->sysclk / sys_fs) { + case 256: + clk_ctl |= SGTL5000_MCLK_FREQ_256FS << + SGTL5000_MCLK_FREQ_SHIFT; + break; + case 384: + clk_ctl |= SGTL5000_MCLK_FREQ_384FS << + SGTL5000_MCLK_FREQ_SHIFT; + break; + case 512: + clk_ctl |= SGTL5000_MCLK_FREQ_512FS << + SGTL5000_MCLK_FREQ_SHIFT; + break; + default: + /* if mclk not satisify the divider, use pll */ + if (sgtl5000->master) + clk_ctl |= SGTL5000_MCLK_FREQ_PLL << + SGTL5000_MCLK_FREQ_SHIFT; + else { + pr_err("%s: PLL not supported in slave mode\n", + __func__); + return -EINVAL; + } + } + + /* if using pll, please check manual 6.4.2 for detail */ + if ((clk_ctl & SGTL5000_MCLK_FREQ_MASK) == SGTL5000_MCLK_FREQ_PLL) { + u64 out, t; + int div2; + int pll_ctl; + unsigned int in, int_div, frac_div; + + if (sgtl5000->sysclk > 17000000) { + div2 = 1; + in = sgtl5000->sysclk / 2; + } else { + div2 = 0; + in = sgtl5000->sysclk; + } + if (sys_fs == 44100) + out = 180633600; + else + out = 196608000; + t = do_div(out, in); + int_div = out; + t *= 2048; + do_div(t, in); + frac_div = t; + pll_ctl = int_div << SGTL5000_PLL_INT_DIV_SHIFT | + frac_div << SGTL5000_PLL_FRAC_DIV_SHIFT; + + snd_soc_write(codec, SGTL5000_CHIP_PLL_CTRL, pll_ctl); + if (div2) + snd_soc_update_bits(codec, + SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INPUT_FREQ_DIV2, + SGTL5000_INPUT_FREQ_DIV2); + else + snd_soc_update_bits(codec, + SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INPUT_FREQ_DIV2, + 0); + + /* power up pll */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP); + } else { + /* power down pll */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, + 0); + } + + /* if using pll, clk_ctrl must be set after pll power up */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); + + return 0; +} + +/* + * Set PCM DAI bit size and sample rate. + * input: params_rate, params_fmt + */ +static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int channels = params_channels(params); + int i2s_ctl = 0; + int stereo; + int ret; + + /* sysclk should already set */ + if (!sgtl5000->sysclk) { + dev_err(codec->dev, "%s: set sysclk first!\n", __func__); + return -EFAULT; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + stereo = SGTL5000_DAC_STEREO; + else + stereo = SGTL5000_ADC_STEREO; + + /* set mono to save power */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, stereo, + channels == 1 ? 0 : stereo); + + /* set codec clock base on lrclk */ + ret = sgtl5000_set_clock(codec, params_rate(params)); + if (ret) + return ret; + + /* set i2s data format */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) + return -EINVAL; + i2s_ctl |= SGTL5000_I2S_DLEN_16 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_32FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + i2s_ctl |= SGTL5000_I2S_DLEN_20 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + i2s_ctl |= SGTL5000_I2S_DLEN_24 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S32_LE: + if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) + return -EINVAL; + i2s_ctl |= SGTL5000_I2S_DLEN_32 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, i2s_ctl, i2s_ctl); + + return 0; +} + +static int ldo_regulator_is_enabled(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + + return ldo->enabled; +} + +static int ldo_regulator_enable(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data; + int reg; + + if (ldo_regulator_is_enabled(dev)) + return 0; + + /* set regulator value firstly */ + reg = (1600 - ldo->voltage / 1000) / 50; + reg = clamp(reg, 0x0, 0xf); + + /* amend the voltage value, unit: uV */ + ldo->voltage = (1600 - reg * 50) * 1000; + + /* set voltage to register */ + snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, + (0x1 << 4) - 1, reg); + + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINEREG_D_POWERUP, + SGTL5000_LINEREG_D_POWERUP); + + /* when internal ldo enabled, simple digital power can be disabled */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINREG_SIMPLE_POWERUP, + 0); + + ldo->enabled = 1; + return 0; +} + +static int ldo_regulator_disable(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data; + + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINEREG_D_POWERUP, + 0); + + /* clear voltage info */ + snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, + (0x1 << 4) - 1, 0); + + ldo->enabled = 0; + + return 0; +} + +static int ldo_regulator_get_voltage(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + + return ldo->voltage; +} + +static struct regulator_ops ldo_regulator_ops = { + .is_enabled = ldo_regulator_is_enabled, + .enable = ldo_regulator_enable, + .disable = ldo_regulator_disable, + .get_voltage = ldo_regulator_get_voltage, +}; + +static int ldo_regulator_register(struct snd_soc_codec *codec, + struct regulator_init_data *init_data, + int voltage) +{ + struct ldo_regulator *ldo; + + ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL); + + if (!ldo) { + dev_err(codec->dev, "failed to allocate ldo_regulator\n"); + return -ENOMEM; + } + + ldo->desc.name = kstrdup(dev_name(codec->dev), GFP_KERNEL); + if (!ldo->desc.name) { + dev_err(codec->dev, "failed to allocate decs name memory\n"); + return -ENOMEM; + } + + ldo->desc.type = REGULATOR_VOLTAGE; + ldo->desc.owner = THIS_MODULE; + ldo->desc.ops = &ldo_regulator_ops; + ldo->desc.n_voltages = 1; + + ldo->codec_data = codec; + ldo->voltage = voltage; + + ldo->dev = regulator_register(&ldo->desc, codec->dev, + init_data, ldo); + if (IS_ERR(ldo->dev)) { + dev_err(codec->dev, "failed to register regulator\n"); + kfree(ldo->desc.name); + kfree(ldo); + + return PTR_ERR(ldo->dev); + } + + return 0; +} + +static int ldo_regulator_remove(struct snd_soc_codec *codec) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + struct ldo_regulator *ldo = sgtl5000->ldo; + + if (!ldo) + return 0; + + regulator_unregister(ldo->dev); + kfree(ldo->desc.name); + kfree(ldo); + + return 0; +} + +/* + * set dac bias + * common state changes: + * startup: + * off --> standby --> prepare --> on + * standby --> prepare --> on + * + * stop: + * on --> prepare --> standby + */ +static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable( + ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) + return ret; + udelay(10); + } + + break; + case SND_SOC_BIAS_OFF: + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +#define SGTL5000_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai_ops sgtl5000_ops = { + .hw_params = sgtl5000_pcm_hw_params, + .digital_mute = sgtl5000_digital_mute, + .set_fmt = sgtl5000_set_dai_fmt, + .set_sysclk = sgtl5000_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver sgtl5000_dai = { + .name = "sgtl5000", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + /* + * only support 8~48K + 96K, + * TODO modify hw_param to support more + */ + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000, + .formats = SGTL5000_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000, + .formats = SGTL5000_FORMATS, + }, + .ops = &sgtl5000_ops, + .symmetric_rates = 1, +}; + +static int sgtl5000_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case SGTL5000_CHIP_ID: + case SGTL5000_CHIP_ADCDAC_CTRL: + case SGTL5000_CHIP_ANA_STATUS: + return 1; + } + + return 0; +} + +#ifdef CONFIG_SUSPEND +static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +/* + * restore all sgtl5000 registers, + * since a big hole between dap and regular registers, + * we will restore them respectively. + */ +static int sgtl5000_restore_regs(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i; + int regular_regs = SGTL5000_CHIP_SHORT_CTRL >> 1; + + /* restore regular registers */ + for (i = 0; i < regular_regs; i++) { + int reg = i << 1; + + /* this regs depends on the others */ + if (reg == SGTL5000_CHIP_ANA_POWER || + reg == SGTL5000_CHIP_CLK_CTRL || + reg == SGTL5000_CHIP_LINREG_CTRL || + reg == SGTL5000_CHIP_LINE_OUT_CTRL || + reg == SGTL5000_CHIP_CLK_CTRL) + continue; + + snd_soc_write(codec, reg, cache[i]); + } + + /* restore dap registers */ + for (i = SGTL5000_DAP_REG_OFFSET >> 1; + i < SGTL5000_MAX_REG_OFFSET >> 1; i++) { + int reg = i << 1; + + snd_soc_write(codec, reg, cache[i]); + } + + /* + * restore power and other regs accroding + * to set_power() and set_clock() + */ + snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, + cache[SGTL5000_CHIP_LINREG_CTRL >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, + cache[SGTL5000_CHIP_ANA_POWER >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, + cache[SGTL5000_CHIP_CLK_CTRL >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, + cache[SGTL5000_CHIP_REF_CTRL >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, + cache[SGTL5000_CHIP_LINE_OUT_CTRL >> 1]); + return 0; +} + +static int sgtl5000_resume(struct snd_soc_codec *codec) +{ + /* Bring the codec back up to standby to enable regulators */ + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Restore registers by cached in memory */ + sgtl5000_restore_regs(codec); + return 0; +} +#else +#define sgtl5000_suspend NULL +#define sgtl5000_resume NULL +#endif /* CONFIG_SUSPEND */ + +/* + * sgtl5000 has 3 internal power supplies: + * 1. VAG, normally set to vdda/2 + * 2. chargepump, set to different value + * according to voltage of vdda and vddio + * 3. line out VAG, normally set to vddio/2 + * + * and should be set accroding to: + * 1. vddd provided by external or not + * 2. vdda and vddio voltage value. > 3.1v or not + * 3. chip revision >=0x11 or not. If >=0x11, not use external vddd. + */ +static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) +{ + int vddd, vdda, vddio; + u16 ana_pwr, lreg_ctrl; + int vag; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + vdda = regulator_get_voltage(sgtl5000->supplies[VDDA].consumer); + vddio = regulator_get_voltage(sgtl5000->supplies[VDDIO].consumer); + vddd = regulator_get_voltage(sgtl5000->supplies[VDDD].consumer); + + vdda = vdda / 1000; + vddio = vddio / 1000; + vddd = vddd / 1000; + + if (vdda <= 0 || vddio <= 0 || vddd < 0) { + dev_err(codec->dev, "regulator voltage not set correctly\n"); + + return -EINVAL; + } + + /* accroding to datasheet, maximum voltage of supplies */ + if (vdda > 3600 || vddio > 3600 || vddd > 1980) { + dev_err(codec->dev, + "exceed max voltage vdda %dmv vddio %dma vddd %dma\n", + vdda, vddio, vddd); + + return -EINVAL; + } + + /* reset value */ + ana_pwr = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER); + ana_pwr |= SGTL5000_DAC_STEREO | + SGTL5000_ADC_STEREO | + SGTL5000_REFTOP_POWERUP; + lreg_ctrl = snd_soc_read(codec, SGTL5000_CHIP_LINREG_CTRL); + + if (vddio < 3100 && vdda < 3100) { + /* enable internal oscillator used for charge pump */ + snd_soc_update_bits(codec, SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INT_OSC_EN, + SGTL5000_INT_OSC_EN); + /* Enable VDDC charge pump */ + ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP; + } else if (vddio >= 3100 && vdda >= 3100) { + /* + * if vddio and vddd > 3.1v, + * charge pump should be clean before set ana_pwr + */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VDDC_CHRGPMP_POWERUP, 0); + + /* VDDC use VDDIO rail */ + lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD; + lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO << + SGTL5000_VDDC_MAN_ASSN_SHIFT; + } + + snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr); + + /* set voltage to register */ + snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, + (0x1 << 4) - 1, 0x8); + + /* + * if vddd linear reg has been enabled, + * simple digital supply should be clear to get + * proper VDDD voltage. + */ + if (ana_pwr & SGTL5000_LINEREG_D_POWERUP) + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINREG_SIMPLE_POWERUP, + 0); + else + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINREG_SIMPLE_POWERUP | + SGTL5000_STARTUP_POWERUP, + 0); + + /* + * set ADC/DAC VAG to vdda / 2, + * should stay in range (0.8v, 1.575v) + */ + vag = vdda / 2; + if (vag <= SGTL5000_ANA_GND_BASE) + vag = 0; + else if (vag >= SGTL5000_ANA_GND_BASE + SGTL5000_ANA_GND_STP * + (SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT)) + vag = SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT; + else + vag = (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP; + + snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, + vag << SGTL5000_ANA_GND_SHIFT, + vag << SGTL5000_ANA_GND_SHIFT); + + /* set line out VAG to vddio / 2, in range (0.8v, 1.675v) */ + vag = vddio / 2; + if (vag <= SGTL5000_LINE_OUT_GND_BASE) + vag = 0; + else if (vag >= SGTL5000_LINE_OUT_GND_BASE + + SGTL5000_LINE_OUT_GND_STP * SGTL5000_LINE_OUT_GND_MAX) + vag = SGTL5000_LINE_OUT_GND_MAX; + else + vag = (vag - SGTL5000_LINE_OUT_GND_BASE) / + SGTL5000_LINE_OUT_GND_STP; + + snd_soc_update_bits(codec, SGTL5000_CHIP_LINE_OUT_CTRL, + vag << SGTL5000_LINE_OUT_GND_SHIFT | + SGTL5000_LINE_OUT_CURRENT_360u << + SGTL5000_LINE_OUT_CURRENT_SHIFT, + vag << SGTL5000_LINE_OUT_GND_SHIFT | + SGTL5000_LINE_OUT_CURRENT_360u << + SGTL5000_LINE_OUT_CURRENT_SHIFT); + + return 0; +} + +static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) +{ + u16 reg; + int ret; + int rev, i; + int external_vddd = 0; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) + sgtl5000->supplies[i].supply = supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (!ret) + external_vddd = 1; + else { + /* set internal ldo to 1.2v */ + int voltage = LDO_VOLTAGE; + + sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME; + + ret = ldo_regulator_register(codec, &ldo_init_data, voltage); + if (ret) { + dev_err(codec->dev, + "Failed to register vddd internal supplies: %d\n", + ret); + return ret; + } + ret = regulator_bulk_get(codec->dev, + ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + + if (ret) { + ldo_regulator_remove(codec); + dev_err(codec->dev, + "Failed to request supplies: %d\n", ret); + + return ret; + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) + goto err_regulator_free; + + /* wait for all power rails bring up */ + udelay(10); + + /* read chip information */ + reg = snd_soc_read(codec, SGTL5000_CHIP_ID); + if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != + SGTL5000_PARTID_PART_ID) { + dev_err(codec->dev, + "Device with ID register %x is not a sgtl5000\n", reg); + ret = -ENODEV; + goto err_regulator_disable; + } + + rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + dev_info(codec->dev, "sgtl5000 revision %d\n", rev); + + /* + * workaround for revision 0x11 and later, + * roll back to use internal LDO + */ + if (external_vddd && rev >= 0x11) { + int voltage = LDO_VOLTAGE; + /* disable all regulator first */ + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + /* free VDDD regulator */ + regulator_put(sgtl5000->supplies[VDDD].consumer); + + sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME; + + ret = ldo_regulator_register(codec, &ldo_init_data, voltage); + if (ret) + return ret; + + ret = regulator_bulk_get(codec->dev, + ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) { + ldo_regulator_remove(codec); + dev_err(codec->dev, + "Failed to request supplies: %d\n", ret); + + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) + goto err_regulator_free; + } + + return 0; + +err_regulator_disable: + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); +err_regulator_free: + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (external_vddd) + ldo_regulator_remove(codec); + return ret; + +} +static int sgtl5000_probe(struct snd_soc_codec *codec) +{ + int ret; + + /* setup i2c data ops */ + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + ret = sgtl5000_enable_regulators(codec); + if (ret) + return ret; + + /* power up sgtl5000 */ + ret = sgtl5000_set_power_regs(codec); + if (ret) + return ret; + + /* enable small pop, introduce 400ms delay in turning off */ + snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, + SGTL5000_SMALL_POP, + SGTL5000_SMALL_POP); + + /* disable short cut detector */ + snd_soc_write(codec, SGTL5000_CHIP_SHORT_CTRL, 0); + + /* + * set i2s as default input of sound switch + * TODO: add sound switch to control and dapm widge. + */ + snd_soc_write(codec, SGTL5000_CHIP_SSS_CTRL, + SGTL5000_DAC_SEL_I2S_IN << SGTL5000_DAC_SEL_SHIFT); + snd_soc_write(codec, SGTL5000_CHIP_DIG_POWER, + SGTL5000_ADC_EN | SGTL5000_DAC_EN); + + /* enable dac volume ramp by default */ + snd_soc_write(codec, SGTL5000_CHIP_ADCDAC_CTRL, + SGTL5000_DAC_VOL_RAMP_EN | + SGTL5000_DAC_MUTE_RIGHT | + SGTL5000_DAC_MUTE_LEFT); + + snd_soc_write(codec, SGTL5000_CHIP_PAD_STRENGTH, 0x015f); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_CTRL, + SGTL5000_HP_ZCD_EN | + SGTL5000_ADC_ZCD_EN); + + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + + /* + * disable DAP + * TODO: + * Enable DAP in kcontrol and dapm. + */ + snd_soc_write(codec, SGTL5000_DAP_CTRL, 0); + + /* leading to standby state */ + ret = sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + return ret; + + snd_soc_add_controls(codec, sgtl5000_snd_controls, + ARRAY_SIZE(sgtl5000_snd_controls)); + + snd_soc_dapm_new_controls(&codec->dapm, sgtl5000_dapm_widgets, + ARRAY_SIZE(sgtl5000_dapm_widgets)); + + snd_soc_dapm_add_routes(&codec->dapm, audio_map, + ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(&codec->dapm); + + return 0; +} + +static int sgtl5000_remove(struct snd_soc_codec *codec) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_dapm_free(&codec->dapm); + + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + + ldo_regulator_remove(codec); + + return 0; +} + +struct snd_soc_codec_driver sgtl5000_driver = { + .probe = sgtl5000_probe, + .remove = sgtl5000_remove, + .suspend = sgtl5000_suspend, + .resume = sgtl5000_resume, + .set_bias_level = sgtl5000_set_bias_level, + .reg_cache_size = ARRAY_SIZE(sgtl5000_regs), + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, + .reg_cache_default = sgtl5000_regs, + .volatile_register = sgtl5000_volatile_register, +}; + +static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct sgtl5000_priv *sgtl5000; + int ret; + + sgtl5000 = kzalloc(sizeof(struct sgtl5000_priv), GFP_KERNEL); + if (!sgtl5000) + return -ENOMEM; + + /* + * copy DAP default values to default value array. + * sgtl5000 register space has a big hole, merge it + * at init phase makes life easy. + * FIXME: should we drop 'const' of sgtl5000_regs? + */ + memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)), + sgtl5000_dap_regs, + SGTL5000_MAX_REG_OFFSET - SGTL5000_DAP_REG_OFFSET); + + i2c_set_clientdata(client, sgtl5000); + + ret = snd_soc_register_codec(&client->dev, + &sgtl5000_driver, &sgtl5000_dai, 1); + if (ret) { + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + kfree(sgtl5000); + return ret; + } + + return 0; +} + +static __devexit int sgtl5000_i2c_remove(struct i2c_client *client) +{ + struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + kfree(sgtl5000); + return 0; +} + +static const struct i2c_device_id sgtl5000_id[] = { + {"sgtl5000", 0}, + {}, +}; + +MODULE_DEVICE_TABLE(i2c, sgtl5000_id); + +static struct i2c_driver sgtl5000_i2c_driver = { + .driver = { + .name = "sgtl5000", + .owner = THIS_MODULE, + }, + .probe = sgtl5000_i2c_probe, + .remove = __devexit_p(sgtl5000_i2c_remove), + .id_table = sgtl5000_id, +}; + +static int __init sgtl5000_modinit(void) +{ + return i2c_add_driver(&sgtl5000_i2c_driver); +} +module_init(sgtl5000_modinit); + +static void __exit sgtl5000_exit(void) +{ + i2c_del_driver(&sgtl5000_i2c_driver); +} +module_exit(sgtl5000_exit); + +MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Zeng Zhaoming <zhaoming.zeng(a)freescale.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h new file mode 100644 index 0000000..c29312a --- /dev/null +++ b/sound/soc/codecs/sgtl5000.h @@ -0,0 +1,402 @@ +/* + * sgtl5000.h - SGTL5000 audio codec interface + * + * Copyright 2010-2011 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _SGTL5000_H +#define _SGTL5000_H + +#include <linux/i2c.h> + +/* + * Register values. + */ +#define SGTL5000_CHIP_ID 0x0000 +#define SGTL5000_CHIP_DIG_POWER 0x0002 +#define SGTL5000_CHIP_CLK_CTRL 0x0004 +#define SGTL5000_CHIP_I2S_CTRL 0x0006 +#define SGTL5000_CHIP_SSS_CTRL 0x000a +#define SGTL5000_CHIP_ADCDAC_CTRL 0x000e +#define SGTL5000_CHIP_DAC_VOL 0x0010 +#define SGTL5000_CHIP_PAD_STRENGTH 0x0014 +#define SGTL5000_CHIP_ANA_ADC_CTRL 0x0020 +#define SGTL5000_CHIP_ANA_HP_CTRL 0x0022 +#define SGTL5000_CHIP_ANA_CTRL 0x0024 +#define SGTL5000_CHIP_LINREG_CTRL 0x0026 +#define SGTL5000_CHIP_REF_CTRL 0x0028 +#define SGTL5000_CHIP_MIC_CTRL 0x002a +#define SGTL5000_CHIP_LINE_OUT_CTRL 0x002c +#define SGTL5000_CHIP_LINE_OUT_VOL 0x002e +#define SGTL5000_CHIP_ANA_POWER 0x0030 +#define SGTL5000_CHIP_PLL_CTRL 0x0032 +#define SGTL5000_CHIP_CLK_TOP_CTRL 0x0034 +#define SGTL5000_CHIP_ANA_STATUS 0x0036 +#define SGTL5000_CHIP_SHORT_CTRL 0x003c +#define SGTL5000_CHIP_ANA_TEST2 0x003a +#define SGTL5000_DAP_CTRL 0x0100 +#define SGTL5000_DAP_PEQ 0x0102 +#define SGTL5000_DAP_BASS_ENHANCE 0x0104 +#define SGTL5000_DAP_BASS_ENHANCE_CTRL 0x0106 +#define SGTL5000_DAP_AUDIO_EQ 0x0108 +#define SGTL5000_DAP_SURROUND 0x010a +#define SGTL5000_DAP_FLT_COEF_ACCESS 0x010c +#define SGTL5000_DAP_COEF_WR_B0_MSB 0x010e +#define SGTL5000_DAP_COEF_WR_B0_LSB 0x0110 +#define SGTL5000_DAP_EQ_BASS_BAND0 0x0116 +#define SGTL5000_DAP_EQ_BASS_BAND1 0x0118 +#define SGTL5000_DAP_EQ_BASS_BAND2 0x011a +#define SGTL5000_DAP_EQ_BASS_BAND3 0x011c +#define SGTL5000_DAP_EQ_BASS_BAND4 0x011e +#define SGTL5000_DAP_MAIN_CHAN 0x0120 +#define SGTL5000_DAP_MIX_CHAN 0x0122 +#define SGTL5000_DAP_AVC_CTRL 0x0124 +#define SGTL5000_DAP_AVC_THRESHOLD 0x0126 +#define SGTL5000_DAP_AVC_ATTACK 0x0128 +#define SGTL5000_DAP_AVC_DECAY 0x012a +#define SGTL5000_DAP_COEF_WR_B1_MSB 0x012c +#define SGTL5000_DAP_COEF_WR_B1_LSB 0x012e +#define SGTL5000_DAP_COEF_WR_B2_MSB 0x0130 +#define SGTL5000_DAP_COEF_WR_B2_LSB 0x0132 +#define SGTL5000_DAP_COEF_WR_A1_MSB 0x0134 +#define SGTL5000_DAP_COEF_WR_A1_LSB 0x0136 +#define SGTL5000_DAP_COEF_WR_A2_MSB 0x0138 +#define SGTL5000_DAP_COEF_WR_A2_LSB 0x013a + +/* + * Field Definitions. + */ + +/* + * SGTL5000_CHIP_ID + */ +#define SGTL5000_PARTID_MASK 0xff00 +#define SGTL5000_PARTID_SHIFT 8 +#define SGTL5000_PARTID_WIDTH 8 +#define SGTL5000_PARTID_PART_ID 0xa0 +#define SGTL5000_REVID_MASK 0x00ff +#define SGTL5000_REVID_SHIFT 0 +#define SGTL5000_REVID_WIDTH 8 + +/* + * SGTL5000_CHIP_DIG_POWER + */ +#define SGTL5000_ADC_EN 0x0040 +#define SGTL5000_DAC_EN 0x0020 +#define SGTL5000_DAP_POWERUP 0x0010 +#define SGTL5000_I2S_OUT_POWERUP 0x0002 +#define SGTL5000_I2S_IN_POWERUP 0x0001 + +/* + * SGTL5000_CHIP_CLK_CTRL + */ +#define SGTL5000_RATE_MODE_MASK 0x0030 +#define SGTL5000_RATE_MODE_SHIFT 4 +#define SGTL5000_RATE_MODE_WIDTH 2 +#define SGTL5000_RATE_MODE_DIV_1 0 +#define SGTL5000_RATE_MODE_DIV_2 1 +#define SGTL5000_RATE_MODE_DIV_4 2 +#define SGTL5000_RATE_MODE_DIV_6 3 +#define SGTL5000_SYS_FS_MASK 0x000c +#define SGTL5000_SYS_FS_SHIFT 2 +#define SGTL5000_SYS_FS_WIDTH 2 +#define SGTL5000_SYS_FS_32k 0x0 +#define SGTL5000_SYS_FS_44_1k 0x1 +#define SGTL5000_SYS_FS_48k 0x2 +#define SGTL5000_SYS_FS_96k 0x3 +#define SGTL5000_MCLK_FREQ_MASK 0x0003 +#define SGTL5000_MCLK_FREQ_SHIFT 0 +#define SGTL5000_MCLK_FREQ_WIDTH 2 +#define SGTL5000_MCLK_FREQ_256FS 0x0 +#define SGTL5000_MCLK_FREQ_384FS 0x1 +#define SGTL5000_MCLK_FREQ_512FS 0x2 +#define SGTL5000_MCLK_FREQ_PLL 0x3 + +/* + * SGTL5000_CHIP_I2S_CTRL + */ +#define SGTL5000_I2S_SCLKFREQ_MASK 0x0100 +#define SGTL5000_I2S_SCLKFREQ_SHIFT 8 +#define SGTL5000_I2S_SCLKFREQ_WIDTH 1 +#define SGTL5000_I2S_SCLKFREQ_64FS 0x0 +#define SGTL5000_I2S_SCLKFREQ_32FS 0x1 /* Not for RJ mode */ +#define SGTL5000_I2S_MASTER 0x0080 +#define SGTL5000_I2S_SCLK_INV 0x0040 +#define SGTL5000_I2S_DLEN_MASK 0x0030 +#define SGTL5000_I2S_DLEN_SHIFT 4 +#define SGTL5000_I2S_DLEN_WIDTH 2 +#define SGTL5000_I2S_DLEN_32 0x0 +#define SGTL5000_I2S_DLEN_24 0x1 +#define SGTL5000_I2S_DLEN_20 0x2 +#define SGTL5000_I2S_DLEN_16 0x3 +#define SGTL5000_I2S_MODE_MASK 0x000c +#define SGTL5000_I2S_MODE_SHIFT 2 +#define SGTL5000_I2S_MODE_WIDTH 2 +#define SGTL5000_I2S_MODE_I2S_LJ 0x0 +#define SGTL5000_I2S_MODE_RJ 0x1 +#define SGTL5000_I2S_MODE_PCM 0x2 +#define SGTL5000_I2S_LRALIGN 0x0002 +#define SGTL5000_I2S_LRPOL 0x0001 /* set for which mode */ + +/* + * SGTL5000_CHIP_SSS_CTRL + */ +#define SGTL5000_DAP_MIX_LRSWAP 0x4000 +#define SGTL5000_DAP_LRSWAP 0x2000 +#define SGTL5000_DAC_LRSWAP 0x1000 +#define SGTL5000_I2S_OUT_LRSWAP 0x0400 +#define SGTL5000_DAP_MIX_SEL_MASK 0x0300 +#define SGTL5000_DAP_MIX_SEL_SHIFT 8 +#define SGTL5000_DAP_MIX_SEL_WIDTH 2 +#define SGTL5000_DAP_MIX_SEL_ADC 0x0 +#define SGTL5000_DAP_MIX_SEL_I2S_IN 0x1 +#define SGTL5000_DAP_SEL_MASK 0x00c0 +#define SGTL5000_DAP_SEL_SHIFT 6 +#define SGTL5000_DAP_SEL_WIDTH 2 +#define SGTL5000_DAP_SEL_ADC 0x0 +#define SGTL5000_DAP_SEL_I2S_IN 0x1 +#define SGTL5000_DAC_SEL_MASK 0x0030 +#define SGTL5000_DAC_SEL_SHIFT 4 +#define SGTL5000_DAC_SEL_WIDTH 2 +#define SGTL5000_DAC_SEL_ADC 0x0 +#define SGTL5000_DAC_SEL_I2S_IN 0x1 +#define SGTL5000_DAC_SEL_DAP 0x3 +#define SGTL5000_I2S_OUT_SEL_MASK 0x0003 +#define SGTL5000_I2S_OUT_SEL_SHIFT 0 +#define SGTL5000_I2S_OUT_SEL_WIDTH 2 +#define SGTL5000_I2S_OUT_SEL_ADC 0x0 +#define SGTL5000_I2S_OUT_SEL_I2S_IN 0x1 +#define SGTL5000_I2S_OUT_SEL_DAP 0x3 + +/* + * SGTL5000_CHIP_ADCDAC_CTRL + */ +#define SGTL5000_VOL_BUSY_DAC_RIGHT 0x2000 +#define SGTL5000_VOL_BUSY_DAC_LEFT 0x1000 +#define SGTL5000_DAC_VOL_RAMP_EN 0x0200 +#define SGTL5000_DAC_VOL_RAMP_EXPO 0x0100 +#define SGTL5000_DAC_MUTE_RIGHT 0x0008 +#define SGTL5000_DAC_MUTE_LEFT 0x0004 +#define SGTL5000_ADC_HPF_FREEZE 0x0002 +#define SGTL5000_ADC_HPF_BYPASS 0x0001 + +/* + * SGTL5000_CHIP_DAC_VOL + */ +#define SGTL5000_DAC_VOL_RIGHT_MASK 0xff00 +#define SGTL5000_DAC_VOL_RIGHT_SHIFT 8 +#define SGTL5000_DAC_VOL_RIGHT_WIDTH 8 +#define SGTL5000_DAC_VOL_LEFT_MASK 0x00ff +#define SGTL5000_DAC_VOL_LEFT_SHIFT 0 +#define SGTL5000_DAC_VOL_LEFT_WIDTH 8 + +/* + * SGTL5000_CHIP_PAD_STRENGTH + */ +#define SGTL5000_PAD_I2S_LRCLK_MASK 0x0300 +#define SGTL5000_PAD_I2S_LRCLK_SHIFT 8 +#define SGTL5000_PAD_I2S_LRCLK_WIDTH 2 +#define SGTL5000_PAD_I2S_SCLK_MASK 0x00c0 +#define SGTL5000_PAD_I2S_SCLK_SHIFT 6 +#define SGTL5000_PAD_I2S_SCLK_WIDTH 2 +#define SGTL5000_PAD_I2S_DOUT_MASK 0x0030 +#define SGTL5000_PAD_I2S_DOUT_SHIFT 4 +#define SGTL5000_PAD_I2S_DOUT_WIDTH 2 +#define SGTL5000_PAD_I2C_SDA_MASK 0x000c +#define SGTL5000_PAD_I2C_SDA_SHIFT 2 +#define SGTL5000_PAD_I2C_SDA_WIDTH 2 +#define SGTL5000_PAD_I2C_SCL_MASK 0x0003 +#define SGTL5000_PAD_I2C_SCL_SHIFT 0 +#define SGTL5000_PAD_I2C_SCL_WIDTH 2 + +/* + * SGTL5000_CHIP_ANA_ADC_CTRL + */ +#define SGTL5000_ADC_VOL_M6DB 0x0100 +#define SGTL5000_ADC_VOL_RIGHT_MASK 0x00f0 +#define SGTL5000_ADC_VOL_RIGHT_SHIFT 4 +#define SGTL5000_ADC_VOL_RIGHT_WIDTH 4 +#define SGTL5000_ADC_VOL_LEFT_MASK 0x000f +#define SGTL5000_ADC_VOL_LEFT_SHIFT 0 +#define SGTL5000_ADC_VOL_LEFT_WIDTH 4 + +/* + * SGTL5000_CHIP_ANA_HP_CTRL + */ +#define SGTL5000_HP_VOL_RIGHT_MASK 0x7f00 +#define SGTL5000_HP_VOL_RIGHT_SHIFT 8 +#define SGTL5000_HP_VOL_RIGHT_WIDTH 7 +#define SGTL5000_HP_VOL_LEFT_MASK 0x007f +#define SGTL5000_HP_VOL_LEFT_SHIFT 0 +#define SGTL5000_HP_VOL_LEFT_WIDTH 7 + +/* + * SGTL5000_CHIP_ANA_CTRL + */ +#define SGTL5000_LINE_OUT_MUTE 0x0100 +#define SGTL5000_HP_SEL_MASK 0x0040 +#define SGTL5000_HP_SEL_SHIFT 6 +#define SGTL5000_HP_SEL_WIDTH 1 +#define SGTL5000_HP_SEL_DAC 0x0 +#define SGTL5000_HP_SEL_LINE_IN 0x1 +#define SGTL5000_HP_ZCD_EN 0x0020 +#define SGTL5000_HP_MUTE 0x0010 +#define SGTL5000_ADC_SEL_MASK 0x0004 +#define SGTL5000_ADC_SEL_SHIFT 2 +#define SGTL5000_ADC_SEL_WIDTH 1 +#define SGTL5000_ADC_SEL_MIC 0x0 +#define SGTL5000_ADC_SEL_LINE_IN 0x1 +#define SGTL5000_ADC_ZCD_EN 0x0002 +#define SGTL5000_ADC_MUTE 0x0001 + +/* + * SGTL5000_CHIP_LINREG_CTRL + */ +#define SGTL5000_VDDC_MAN_ASSN_MASK 0x0040 +#define SGTL5000_VDDC_MAN_ASSN_SHIFT 6 +#define SGTL5000_VDDC_MAN_ASSN_WIDTH 1 +#define SGTL5000_VDDC_MAN_ASSN_VDDA 0x0 +#define SGTL5000_VDDC_MAN_ASSN_VDDIO 0x1 +#define SGTL5000_VDDC_ASSN_OVRD 0x0020 +#define SGTL5000_LINREG_VDDD_MASK 0x000f +#define SGTL5000_LINREG_VDDD_SHIFT 0 +#define SGTL5000_LINREG_VDDD_WIDTH 4 + +/* + * SGTL5000_CHIP_REF_CTRL + */ +#define SGTL5000_ANA_GND_MASK 0x01f0 +#define SGTL5000_ANA_GND_SHIFT 4 +#define SGTL5000_ANA_GND_WIDTH 5 +#define SGTL5000_ANA_GND_BASE 800 /* mv */ +#define SGTL5000_ANA_GND_STP 25 /*mv */ +#define SGTL5000_BIAS_CTRL_MASK 0x000e +#define SGTL5000_BIAS_CTRL_SHIFT 1 +#define SGTL5000_BIAS_CTRL_WIDTH 3 +#define SGTL5000_SMALL_POP 0x0001 + +/* + * SGTL5000_CHIP_MIC_CTRL + */ +#define SGTL5000_BIAS_R_MASK 0x0200 +#define SGTL5000_BIAS_R_SHIFT 8 +#define SGTL5000_BIAS_R_WIDTH 2 +#define SGTL5000_BIAS_R_off 0x0 +#define SGTL5000_BIAS_R_2K 0x1 +#define SGTL5000_BIAS_R_4k 0x2 +#define SGTL5000_BIAS_R_8k 0x3 +#define SGTL5000_BIAS_VOLT_MASK 0x0070 +#define SGTL5000_BIAS_VOLT_SHIFT 4 +#define SGTL5000_BIAS_VOLT_WIDTH 3 +#define SGTL5000_MIC_GAIN_MASK 0x0003 +#define SGTL5000_MIC_GAIN_SHIFT 0 +#define SGTL5000_MIC_GAIN_WIDTH 2 + +/* + * SGTL5000_CHIP_LINE_OUT_CTRL + */ +#define SGTL5000_LINE_OUT_CURRENT_MASK 0x0f00 +#define SGTL5000_LINE_OUT_CURRENT_SHIFT 8 +#define SGTL5000_LINE_OUT_CURRENT_WIDTH 4 +#define SGTL5000_LINE_OUT_CURRENT_180u 0x0 +#define SGTL5000_LINE_OUT_CURRENT_270u 0x1 +#define SGTL5000_LINE_OUT_CURRENT_360u 0x3 +#define SGTL5000_LINE_OUT_CURRENT_450u 0x7 +#define SGTL5000_LINE_OUT_CURRENT_540u 0xf +#define SGTL5000_LINE_OUT_GND_MASK 0x003f +#define SGTL5000_LINE_OUT_GND_SHIFT 0 +#define SGTL5000_LINE_OUT_GND_WIDTH 6 +#define SGTL5000_LINE_OUT_GND_BASE 800 /* mv */ +#define SGTL5000_LINE_OUT_GND_STP 25 +#define SGTL5000_LINE_OUT_GND_MAX 0x23 + +/* + * SGTL5000_CHIP_LINE_OUT_VOL + */ +#define SGTL5000_LINE_OUT_VOL_RIGHT_MASK 0x1f00 +#define SGTL5000_LINE_OUT_VOL_RIGHT_SHIFT 8 +#define SGTL5000_LINE_OUT_VOL_RIGHT_WIDTH 5 +#define SGTL5000_LINE_OUT_VOL_LEFT_MASK 0x001f +#define SGTL5000_LINE_OUT_VOL_LEFT_SHIFT 0 +#define SGTL5000_LINE_OUT_VOL_LEFT_WIDTH 5 + +/* + * SGTL5000_CHIP_ANA_POWER + */ +#define SGTL5000_DAC_STEREO 0x4000 +#define SGTL5000_LINREG_SIMPLE_POWERUP 0x2000 +#define SGTL5000_STARTUP_POWERUP 0x1000 +#define SGTL5000_VDDC_CHRGPMP_POWERUP 0x0800 +#define SGTL5000_PLL_POWERUP 0x0400 +#define SGTL5000_LINEREG_D_POWERUP 0x0200 +#define SGTL5000_VCOAMP_POWERUP 0x0100 +#define SGTL5000_VAG_POWERUP 0x0080 +#define SGTL5000_ADC_STEREO 0x0040 +#define SGTL5000_REFTOP_POWERUP 0x0020 +#define SGTL5000_HP_POWERUP 0x0010 +#define SGTL5000_DAC_POWERUP 0x0008 +#define SGTL5000_CAPLESS_HP_POWERUP 0x0004 +#define SGTL5000_ADC_POWERUP 0x0002 +#define SGTL5000_LINE_OUT_POWERUP 0x0001 + +/* + * SGTL5000_CHIP_PLL_CTRL + */ +#define SGTL5000_PLL_INT_DIV_MASK 0xf800 +#define SGTL5000_PLL_INT_DIV_SHIFT 11 +#define SGTL5000_PLL_INT_DIV_WIDTH 5 +#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700 +#define SGTL5000_PLL_FRAC_DIV_SHIFT 0 +#define SGTL5000_PLL_FRAC_DIV_WIDTH 11 + +/* + * SGTL5000_CHIP_CLK_TOP_CTRL + */ +#define SGTL5000_INT_OSC_EN 0x0800 +#define SGTL5000_INPUT_FREQ_DIV2 0x0008 + +/* + * SGTL5000_CHIP_ANA_STATUS + */ +#define SGTL5000_HP_LRSHORT 0x0200 +#define SGTL5000_CAPLESS_SHORT 0x0100 +#define SGTL5000_PLL_LOCKED 0x0010 + +/* + * SGTL5000_CHIP_SHORT_CTRL + */ +#define SGTL5000_LVLADJR_MASK 0x7000 +#define SGTL5000_LVLADJR_SHIFT 12 +#define SGTL5000_LVLADJR_WIDTH 3 +#define SGTL5000_LVLADJL_MASK 0x0700 +#define SGTL5000_LVLADJL_SHIFT 8 +#define SGTL5000_LVLADJL_WIDTH 3 +#define SGTL5000_LVLADJC_MASK 0x0070 +#define SGTL5000_LVLADJC_SHIFT 4 +#define SGTL5000_LVLADJC_WIDTH 3 +#define SGTL5000_LR_SHORT_MOD_MASK 0x000c +#define SGTL5000_LR_SHORT_MOD_SHIFT 2 +#define SGTL5000_LR_SHORT_MOD_WIDTH 2 +#define SGTL5000_CM_SHORT_MOD_MASK 0x0003 +#define SGTL5000_CM_SHORT_MOD_SHIFT 0 +#define SGTL5000_CM_SHORT_MOD_WIDTH 2 + +/* + *SGTL5000_CHIP_ANA_TEST2 + */ +#define SGTL5000_MONO_DAC 0x1000 + +/* + * SGTL5000_DAP_CTRL + */ +#define SGTL5000_DAP_MIX_EN 0x0010 +#define SGTL5000_DAP_EN 0x0001 + +#define SGTL5000_SYSCLK 0x00 +#define SGTL5000_LRCLK 0x01 + +#endif -- 1.7.0.4

6 7

[alsa-devel] [PATCH 1/2] ASoC: Fix WM8958 default microphone detection argument ordering
by Mark Brown 22 Feb '11

22 Feb '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 5 ++--- 1 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 4df047e..5f16870 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3012,11 +3012,10 @@ static void wm8958_default_micdet(u16 status, void *data) report |= SND_JACK_BTN_5; done: - snd_soc_jack_report(wm8994->micdet[0].jack, + snd_soc_jack_report(wm8994->micdet[0].jack, report, SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_BTN_4 | SND_JACK_BTN_5 | - SND_JACK_MICROPHONE | SND_JACK_VIDEOOUT, - report); + SND_JACK_MICROPHONE | SND_JACK_VIDEOOUT); } /** -- 1.7.2.3

2 2

[alsa-devel] [PATCH] ASoC: Support configuration of WM8958 microphone bias analogue parameters
by Mark Brown 22 Feb '11

22 Feb '11

The WM8958 has a different microphone bias architecture to WM8994 so needs different configuration to WM8994. Support this in platform data. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/mfd/wm8994/pdata.h | 7 +++++-- include/linux/mfd/wm8994/registers.h | 2 ++ sound/soc/codecs/wm8994.c | 7 +++++++ 3 files changed, 14 insertions(+), 2 deletions(-) diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index 0686946..466b1c7 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -108,13 +108,16 @@ struct wm8994_pdata { */ int micdet_irq; - /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ + /* WM8994 microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ unsigned int micbias1_lvl:1; unsigned int micbias2_lvl:1; - /* Jack detect threashold levels, see datasheet for values */ + /* WM8994 jack detect threashold levels, see datasheet for values */ unsigned int jd_scthr:2; unsigned int jd_thr:2; + + /* WM8958 microphone bias configuration */ + int micbias[2]; }; #endif diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index be072fa..f3ee842 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -63,6 +63,8 @@ #define WM8994_MICBIAS 0x3A #define WM8994_LDO_1 0x3B #define WM8994_LDO_2 0x3C +#define WM8958_MICBIAS1 0x3D +#define WM8958_MICBIAS2 0x3E #define WM8994_CHARGE_PUMP_1 0x4C #define WM8958_CHARGE_PUMP_2 0x4D #define WM8994_CLASS_W_1 0x51 diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index ce192cc..4df047e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2859,6 +2859,13 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) else snd_soc_add_controls(wm8994->codec, wm8994_eq_controls, ARRAY_SIZE(wm8994_eq_controls)); + + for (i = 0; i < ARRAY_SIZE(pdata->micbias); i++) { + if (pdata->micbias[i]) { + snd_soc_write(codec, WM8958_MICBIAS1 + i, + pdata->micbias[i] & 0xffff); + } + } } /** -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Correct definition of WM8903_VMID_RES_5K
by Mark Brown 22 Feb '11

22 Feb '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h index 7d96646..ed4cb5b 100644 --- a/sound/soc/codecs/wm8903.h +++ b/sound/soc/codecs/wm8903.h @@ -183,7 +183,7 @@ extern int wm8903_mic_detect(struct snd_soc_codec *codec, #define WM8903_VMID_RES_50K 2 #define WM8903_VMID_RES_250K 3 -#define WM8903_VMID_RES_5K 4 +#define WM8903_VMID_RES_5K 6 /* * R8 (0x08) - Analogue DAC 0 -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: omap: rx51: Add FM transmitter support
by Jarkko Nikula 22 Feb '11

22 Feb '11

Si4713 FM transmitter on Nokia RX-51/N900 is connected to same Line out signals of TLV320AIC34 than TPA6130 headphone amplifier. This patch adds route to transmitter and "FM Transmitter" control to keep route active when needed. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- - Liam: This goes on top of previous rx51 update (I didn't find those from your tree): http://mailman.alsa-project.org/pipermail/alsa-devel/2011-February/036972.h… - For those who are keen to try you need following patch in order to be able to activate the FM transmitter: http://marc.info/?l=linux-omap&m=129827572706543&w=2 Otherwise there is no any build or runtime dependency to Si4713 driver. --- sound/soc/omap/rx51.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index e28f089..055e447 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -251,6 +251,7 @@ static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMic", NULL), SND_SOC_DAPM_HP("Headphone Jack", rx51_hp_event), SND_SOC_DAPM_MIC("HS Mic", NULL), + SND_SOC_DAPM_LINE("FM Transmitter", NULL), }; static const struct snd_soc_dapm_widget aic34_dapm_widgetsb[] = { @@ -262,6 +263,8 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Ext Spk", NULL, "HPROUT"}, {"Headphone Jack", NULL, "LLOUT"}, {"Headphone Jack", NULL, "RLOUT"}, + {"FM Transmitter", NULL, "LLOUT"}, + {"FM Transmitter", NULL, "RLOUT"}, {"DMic Rate 64", NULL, "Mic Bias 2V"}, {"Mic Bias 2V", NULL, "DMic"}, @@ -292,6 +295,7 @@ static const struct snd_kcontrol_new aic34_rx51_controls[] = { rx51_get_input, rx51_set_input), SOC_ENUM_EXT("Jack Function", rx51_enum[2], rx51_get_jack, rx51_set_jack), + SOC_DAPM_PIN_SWITCH("FM Transmitter"), }; static const struct snd_kcontrol_new aic34_rx51_controlsb[] = { -- 1.7.2.3

4 4

[alsa-devel] [PATCH] ASoC: Mark WM8958 microphone bias registers as readable
by Mark Brown 22 Feb '11

22 Feb '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994-tables.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index 42b3248..61dfd91 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -62,8 +62,8 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x00FF, 0x00FF }, /* R58 - MICBIAS */ { 0x000F, 0x000F }, /* R59 - LDO 1 */ { 0x0007, 0x0007 }, /* R60 - LDO 2 */ - { 0x0000, 0x0000 }, /* R61 */ - { 0x0000, 0x0000 }, /* R62 */ + { 0xFFFF, 0xFFFF }, /* R61 */ + { 0xFFFF, 0xFFFF }, /* R62 */ { 0x0000, 0x0000 }, /* R63 */ { 0x0000, 0x0000 }, /* R64 */ { 0x0000, 0x0000 }, /* R65 */ -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Hook wm_hubs micbiases up to CLK_SYS
by Mark Brown 22 Feb '11

22 Feb '11

The microphone detection functionality requires a clock to work. In any non-detection case where the MICBIAS is enabled CLK_SYS will be needed anyway so there is no negative impact on power consumption. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index a5c1556..7b6b3c1 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -675,6 +675,9 @@ SND_SOC_DAPM_OUTPUT("LINEOUT2N"), }; static const struct snd_soc_dapm_route analogue_routes[] = { + { "MICBIAS1", NULL, "CLK_SYS" }, + { "MICBIAS2", NULL, "CLK_SYS" }, + { "IN1L PGA", "IN1LP Switch", "IN1LP" }, { "IN1L PGA", "IN1LN Switch", "IN1LN" }, -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Mark WM8958 microphone detection registers readable
by Mark Brown 22 Feb '11

22 Feb '11

So they show up in codec_reg. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994-tables.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index 68e9b02..42b3248 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -209,9 +209,9 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x0000, 0x0000 }, /* R205 */ { 0x0000, 0x0000 }, /* R206 */ { 0x0000, 0x0000 }, /* R207 */ - { 0x0000, 0x0000 }, /* R208 */ - { 0x0000, 0x0000 }, /* R209 */ - { 0x0000, 0x0000 }, /* R210 */ + { 0xFFFF, 0xFFFF }, /* R208 */ + { 0xFFFF, 0xFFFF }, /* R209 */ + { 0xFFFF, 0xFFFF }, /* R210 */ { 0x0000, 0x0000 }, /* R211 */ { 0x0000, 0x0000 }, /* R212 */ { 0x0000, 0x0000 }, /* R213 */ -- 1.7.2.3

2 1

[alsa-devel] [PATCH 1/2] ASoC: Remove card from snd_soc_dapm_set_bias_level()
by Mark Brown 22 Feb '11

22 Feb '11

We can get the card from the DAPM context so don't bother passing it as an argument. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 22 +++++++++++----------- 1 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3bee4b2..1a932bb 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -125,17 +125,17 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( /** * snd_soc_dapm_set_bias_level - set the bias level for the system - * @card: audio device + * @dapm: DAPM context * @level: level to configure * * Configure the bias (power) levels for the SoC audio device. * * Returns 0 for success else error. */ -static int snd_soc_dapm_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, +static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { + struct snd_soc_card *card = dapm->card; int ret = 0; switch (level) { @@ -1112,7 +1112,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) list_for_each_entry(d, &dapm->card->dapm_list, list) { if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_dapm_set_bias_level(card, d, + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); if (ret != 0) dev_err(d->dev, @@ -1122,7 +1122,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) /* If we're changing to all on or all off then prepare */ if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) || (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) { - ret = snd_soc_dapm_set_bias_level(card, d, + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE); if (ret != 0) dev_err(d->dev, @@ -1141,7 +1141,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) list_for_each_entry(d, &dapm->card->dapm_list, list) { /* If we just powered the last thing off drop to standby bias */ if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) { - ret = snd_soc_dapm_set_bias_level(card, d, + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); if (ret != 0) dev_err(d->dev, @@ -1152,7 +1152,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) /* If we're in standby and can support bias off then do that */ if (d->bias_level == SND_SOC_BIAS_STANDBY && d->idle_bias_off) { - ret = snd_soc_dapm_set_bias_level(card, d, + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_OFF); if (ret != 0) dev_err(d->dev, @@ -1161,7 +1161,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) /* If we just powered up then move to active bias */ if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) { - ret = snd_soc_dapm_set_bias_level(card, d, + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_ON); if (ret != 0) dev_err(d->dev, @@ -2439,9 +2439,9 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) * standby. */ if (powerdown) { - snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_PREPARE); + snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE); dapm_seq_run(dapm, &down_list, 0, false); - snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_STANDBY); + snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY); } } @@ -2454,7 +2454,7 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card) list_for_each_entry(codec, &card->codec_dev_list, list) { soc_dapm_shutdown_codec(&codec->dapm); - snd_soc_dapm_set_bias_level(card, &codec->dapm, SND_SOC_BIAS_OFF); + snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF); } } -- 1.7.2.3

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[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 22 Feb '11

22 Feb '11

The following changes since commit 6ed8f1485fc82d44ac464bc84a7dcdddd1fa096f: ASoC: Improve WM8994 digital power sequencing (2011-02-03 20:17:13 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 These are all driver specific with the exception of the two commits from me. These are both small and focused changes which fix obvious issues for some systems, both being things that the framework should clearly have been doing but had omitted to do. Dimitris Papastamos (2): ASoC: WM8994: Improve robustness in some use cases ASoC: WM8994: Improve playback robustness Janusz Krzysztofik (1): ASoC: CX20442: fix wrong reg_cache_default content Mark Brown (2): ASoC: Sync initial widget state with hardware ASoC: Ensure supplies are maintained for force enabled widgets Stephen Warren (2): ASoC: WM8903: Fix mic detection register definitions ASoC: WM8903: Fix mic detection enable logic include/sound/wm8903.h | 10 +-- sound/soc/codecs/cx20442.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8994.c | 197 ++++++++++++++++++++++++++++++++++++++++--- sound/soc/soc-dapm.c | 23 +++++- 5 files changed, 210 insertions(+), 24 deletions(-)

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[alsa-devel] ALC270 alsa
by Commander 21 Feb '11

21 Feb '11

Hi Need help with my laptop to get any sound from it. Alsa script = http://www.alsa-project.org/db/?f=2ce0f50d2d7973d17c8cbdf4ee95f2860365c005 Tried various things however nothing seems to work. speaker-test -c 2 does not output any sound. Best Regards Artur

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[alsa-devel] [PATCH v7 0/2] input: Add support for ECI (multimedia) accessories
by tapio.vihuri＠nokia.com 21 Feb '11

21 Feb '11

From: Tapio Vihuri <tapio.vihuri(a)nokia.com> Thank you for comments. Major reformat inspired by adxl34x implementation. There is no more separate input and controller driver. They are combined into one, and controller part is selectable via Kconfig. Driver is now called simply "ECI input driver" I'm planning to introduce second controller later. Both controllers are i2c-devices, but internal way of working is totally different. Created against linux-2.6.38-rc4 at: 795abaf klist: Fix object alignment on 64-bit. Please review. --- v6 message --- Thank you for comments. ECI input driver changes: - Add debuging config option with adequate "help" in Kconfig - Clean out debug messages - Clean out error messages - Force all buttons up event when unplugging accessory ECI controller driver changes: - Remove unused debuging config option Created against linux-2.6.38-rc2 at: 70d1f36 Merge branch 'for-linus' of git://git.infradead.org/ubi-2.6 Please review. --- v5 message --- Thank you for comments. ECI input driver changes: - Use sparse_keymap_setup() properly, remove unneeded input set_bits ECI controller driver changes: - Added "button" debugfs entry for input driver buttons handling testing --- v4 message --- Thank you for comments. ECI input driver changes: - Use driver name in debugfs - reject accessory stating more than maximum 31 buttons - clean out couple comments ECI controller driver changes: - name change, from ecibus to eci_xxx_ctrl - location change, from drivers/ecibus to drivers/input/misc - major rework from platform driver to i2c-driver - removed mrst platform dependency Checkpatch warn about Kconfig, but I think it is false positive, and I will report it to the maintainer. Platform code changes: - medfield_ prefix changed to mrst_ - added ECI controller needed power control - remover ECI controller platform driver stuff - added i2c stuff for ECI controller driver Created against linux-2.6.37-rc8 at: a25cecc Merge branch 'next' of git://git.monstr.eu/linux-2.6-microblaze --- v3 message --- Thank you for comments. ECI input driver changes: - Makefile and Kconfig sorted alphabetically - using sparse keymap library, much cleaner solution - correct string aligment with 80 column - debugfs initialization fix - sysfs cable entry using just 0/1 - using cheaper __set_bit - insane comments fixed - remove unneeded misc device Created against linux-2.6.37-rc8 at: 989d873 Merge master.kernel.org:/home/rmk/linux-2.6-arm --- v2 message --- Thank you for comments. In series two there Kconfig dependency problem fixed, preventing compilation if X86_MRST (Intel Moorestown) or INPUT_MISC is not selected. Also created against linux-2.6.37-rc8 --- v1 message --- Hi all This patch set introduce Multimedia Headset Accessory support for Nokia phones. Technically those are known as ECI (Enhancement Control Interface) If headset has many buttons, like play, vol+, vol- etc. then it is propably ECI accessory. Among several buttons ECI accessories contains memory for storing several parameters. This ECI input driver provides the following features: - reading ECI configuration memory - ECI buttons as input events Drive is constructed as follows: - ECI accessory input driver deals with headset accessory - ECI bus control driver deals the HW transfering data to/from headset - platform data match used HW In the future accessory detection logic will be added using ALSA jack reporting. Created against linux-2.6.37-rc6 Please review. Tapio Vihuri (2): ECI: input: introduce ECI accessory input driver ECI: adding platform data for ECI driver arch/x86/platform/mrst/mrst.c | 32 ++ drivers/input/misc/Kconfig | 47 ++ drivers/input/misc/Makefile | 3 +- drivers/input/misc/eci.c | 851 +++++++++++++++++++++++++++++++++++++ drivers/input/misc/eci_at20-i2c.c | 713 +++++++++++++++++++++++++++++++ include/linux/input/eci.h | 189 ++++++++ 6 files changed, 1834 insertions(+), 1 deletions(-) create mode 100644 drivers/input/misc/eci.c create mode 100644 drivers/input/misc/eci_at20-i2c.c create mode 100644 include/linux/input/eci.h

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[alsa-devel] USB audio devices & kernel panics
by Moustafa Chamli 21 Feb '11

21 Feb '11

Hello, I was told to ask at the alsa-dev list as to if there's a known bug and fix with USB audio devices causing kernel panics. A bug was reported on Launchpad stating that it is fully reproducible: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/715318 Cheers

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[PATCH] ALSA: HDA: Fix mic initialization in VIA auto parser
by David Henningsson 21 Feb '11

21 Feb '11

This typo caused some microphone inputs not to be correctly initialized on VIA codecs. Reported-By: Mark Goldstein <goldstein.mark(a)gmail.com> Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_via.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index a76c326..63b0054 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -567,7 +567,7 @@ static void via_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = cfg->inputs[i].pin; if (spec->smart51_enabled && is_smart51_pins(spec, nid)) ctl = PIN_OUT; - else if (i == AUTO_PIN_MIC) + else if (cfg->inputs[i].type == AUTO_PIN_MIC) ctl = PIN_VREF50; else ctl = PIN_IN; -- 1.7.1 --------------080401030604030008020500--

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[alsa-devel] [ASoC] Problem with codec driver for tlv320aic32x4 and bias level.
by javier Martin 21 Feb '11

21 Feb '11

Hi, I am currently developing a codec driver for tlv320aic32x4 codec for kernel 2.6.37, based on a TI driver made for 2.6.27. The last version of the code is posted below as plain text, not as a patch because it is only meant for showing purposes. As you can see, my "aic32x4_probe" function sets bias level to SND_SOC_BIAS_OFF. As far as I know, when I enable playback (with aplay command), ASoC should call "aic32x4_set_bias_level" and change level to SND_SOC_BIAS_ON and thus triggering some dapm events. However, this won't happen with the current version of the code. Maybe someone could point out some tips. -- /* * linux/sound/soc/codecs/tlv320aic32x4.c * * Copyright 2011 Vista Silicon S.L. * * Author: Javier Martin <javier.martin(a)vista-silicon.com> * * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. * * 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., 51 Franklin Street, Fifth Floor, Boston, * MA 02110-1301, USA. */ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/cdev.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> #include "tlv320aic32x4.h" #define SOC_DOUBLE_R_AIC32X4(xname, reg_left, reg_right, shift, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .info = snd_soc_info_volsw_2r_aic32x4, \ .get = snd_soc_get_volsw_2r_aic32x4, .put = snd_soc_put_volsw_2r_aic32x4, \ .private_value = (reg_left) | ((shift) << 8) | \ ((mask) << 12) | ((invert) << 20) | ((reg_right) << 24) } static int aic32x4_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level); static int snd_soc_info_volsw_2r_aic32x4(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); static int snd_soc_get_volsw_2r_aic32x4(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); static int snd_soc_put_volsw_2r_aic32x4(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); /* ***************************************************************************** * Structure Initialization ***************************************************************************** */ static const struct snd_kcontrol_new aic32x4_snd_controls[] = { /* Output */ /* sound new kcontrol for PCM Playback volume control */ SOC_DOUBLE_R_AIC32X4("PCM Playback Volume", LDAC_VOL, RDAC_VOL, 0, 0xAf, 0), /* sound new kcontrol for HP driver gain */ SOC_DOUBLE_R_AIC32X4("HP Driver Gain", HPL_GAIN, HPR_GAIN, 0, 0x23, 0), /* sound new kcontrol for LO driver gain */ SOC_DOUBLE_R_AIC32X4("LO Driver Gain", LOL_GAIN, LOR_GAIN, 0, 0x23, 0), /* sound new kcontrol for HP mute */ SOC_DOUBLE_R("HP DAC Playback Switch", HPL_GAIN, HPR_GAIN, 6, 0x01, 1), /* sound new kcontrol for LO mute */ SOC_DOUBLE_R("LO DAC Playback Switch", LOL_GAIN, LOR_GAIN, 6, 0x01, 1), /* Input */ /* sound new kcontrol for PGA capture volume */ SOC_DOUBLE_R("ADC Volume Level", LADC_VOL, RADC_VOL, 0, 0x28, 0), SOC_DOUBLE_R("PGA Gain Level", LMICPGA_VOL_CTRL, RMICPGA_VOL_CTRL, 0, 0x5f, 0), /* DAC Automute control */ SOC_SINGLE("Auto-mute", DAC_MUTE_CTRL_REG, 4, 7, 0), /* AGC controls */ SOC_SINGLE("AGC Left Enable Switch", LEFT_CHN_AGC_1, 7, 1, 0), SOC_SINGLE("AGC Right Enable Switch", RIGHT_CHN_AGC_1, 7, 1, 0), SOC_DOUBLE_R("AGC Target Level", LEFT_CHN_AGC_1, RIGHT_CHN_AGC_1, 4, 0x07, 0), SOC_DOUBLE_R("AGC Gain Hysteresis", LEFT_CHN_AGC_1, RIGHT_CHN_AGC_1, 0, 0x03, 0), SOC_DOUBLE_R("AGC Hysteresis", LEFT_CHN_AGC_2, RIGHT_CHN_AGC_2, 6, 0x03, 0), SOC_DOUBLE_R("AGC Noise Threshold", LEFT_CHN_AGC_2, RIGHT_CHN_AGC_2, 1, 0x1F, 0), SOC_DOUBLE_R("AGC Max PGA", LEFT_CHN_AGC_3, RIGHT_CHN_AGC_3, 0, 0x7F, 0), SOC_DOUBLE_R("AGC Attack Time", LEFT_CHN_AGC_4, RIGHT_CHN_AGC_4, 3, 0x1F, 0), SOC_DOUBLE_R("AGC Decay Time", LEFT_CHN_AGC_5, RIGHT_CHN_AGC_5, 3, 0x1F, 0), SOC_DOUBLE_R("AGC Noise Debounce", LEFT_CHN_AGC_6, RIGHT_CHN_AGC_6, 0, 0x1F, 0), SOC_DOUBLE_R("AGC Signal Debounce", LEFT_CHN_AGC_7, RIGHT_CHN_AGC_7, 0, 0x0F, 0), /* MIC BIAS control */ SOC_SINGLE("MIC BIAS", MICBIAS_CTRL, 6, 1, 0), }; /* the sturcture contains the different values for mclk */ static const struct aic32x4_rate_divs aic32x4_divs[] = { /* * mclk, rate, p_val, pll_j, pll_d, dosr, ndac, mdac, aosr, nadc, madc, blck_N, * codec_speficic_initializations */ /* 8k rate */ {12000000, 8000, 1, 7, 6800, 768, 5, 3, 128, 5, 18, 24, {{60, 1}, {61, 1}}}, {24000000, 8000, 2, 7, 6800, 768, 15, 1, 64, 45, 4, 24, {{60, 1}, {61, 1}}}, {25000000, 8000, 2, 7, 3728, 768, 15, 1, 64, 45, 4, 24, {{60, 1}, {61, 1}}}, /* 11.025k rate */ {12000000, 11025, 1, 7, 5264, 512, 8, 2, 128, 8, 8, 16, {{60, 1}, {61, 1}}}, {24000000, 11025, 2, 7, 5264, 512, 16, 1, 64, 32, 4, 16, {{60, 1}, {61, 1}}}, /* 16k rate */ {12000000, 16000, 1, 7, 6800, 384, 5, 3, 128, 5, 9, 12, {{60, 1}, {61, 1}}}, {24000000, 16000, 2, 7, 6800, 384, 15, 1, 64, 18, 5, 12, {{60, 1}, {61, 1}}}, {25000000, 16000, 2, 7, 3728, 384, 15, 1, 64, 18, 5, 12, {{60, 1}, {61, 1}}}, /* 22.05k rate */ {12000000, 22050, 1, 7, 5264, 256, 4, 4, 128, 4, 8, 8, {{60, 1}, {61, 1}}}, {24000000, 22050, 2, 7, 5264, 256, 16, 1, 64, 16, 4, 8, {{60, 1}, {61, 1}}}, {25000000, 22050, 2, 7, 2253, 256, 16, 1, 64, 16, 4, 8, {{60, 1}, {61, 1}}}, /* 32k rate */ {12000000, 32000, 1, 7, 1680, 192, 2, 7, 64, 2, 21, 6, {{60, 1}, {61, 1}}}, {24000000, 32000, 2, 7, 1680, 192, 7, 2, 64, 7, 6, 6, {{60, 1}, {61, 1}}}, /* 44.1k rate */ {12000000, 44100, 1, 7, 5264, 128, 2, 8, 128, 2, 8, 4, {{60, 1}, {61, 1}}}, {24000000, 44100, 2, 7, 5264, 128, 8, 2, 64, 8, 4, 4, {{60, 1}, {61, 1}}}, {25000000, 44100, 2, 7, 2253, 128, 8, 2, 64, 8, 4, 4, {{60, 1}, {61, 1}}}, /* 48k rate */ {12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4, {{60, 1}, {61, 1}}}, {24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4, {{60, 1}, {61, 1}}}, {25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4, {{60, 1}, {61, 1}}}, /*96k rate */ {12000000, 96000, 1, 8, 1920, 64, 2, 8, 64, 2, 8, 2, {{60, 7}, {61, 7}}}, {24000000, 96000, 2, 8, 1920, 64, 4, 4, 64, 8, 2, 2, {{60, 7}, {61, 7}}}, /*192k */ {12000000, 192000, 1, 8, 1920, 32, 2, 8, 32, 2, 8, 1, {{60, 17}, {61, 13}}}, {24000000, 192000, 2, 8, 1920, 32, 4, 4, 32, 4, 4, 1, {{60, 17}, {61, 13}}}, }; /* *---------------------------------------------------------------------------- * @struct aic32x4_priv | * AIC32X4 priviate data structure to set the system clock, mode and * page number. * @field u32 | sysclk | * system clock * @field s32 | master | * master/slave mode setting for AIC32X4 * @field u8 | page_no | * page number. Here, page 0 and page 1 are used. *---------------------------------------------------------------------------- */ struct aic32x4_priv { u32 sysclk; s32 master; u8 page_no; }; #define aic32x4_reset(c) aic32x4_write(c, AIC32X4_RESET, 0x01); /* * aic32x4 initialization data * This structure initialization contains the initialization required for * AIC32X4. * These registers values (reg_val) are written into the respective AIC32X4 * register offset (reg_offset) to initialize AIC32X4. * These values are used in aic32x4_init() function only. */ static const struct aic32x4_configs aic32x4_reg_init[] = { /* Carry out the software reset */ {AIC32X4_RESET, 0x01}, /* Disable crude LDO */ {POW_CFG, 0x08}, /* Switch on the analog blocks */ {LDO_CTL, 0x01}, /* Do not connect IN1_L and IN1_R to CM (differential mode) */ {INPUT_CFG_REG, 0x00}, /* USE LDOIN range 1.8 - 3.6 V */ {CM_MODE_CTL, 0x03}, /* PLL is CODEC_CLKIN */ {CLK_REG_1, PLLCLK_2_CODEC_CLKIN}, /* DAC_MOD_CLK is BCLK source */ {AIS_REG_3, DAC_MOD_CLK_2_BDIV_CLKIN}, /* Setting up DAC Channel */ {DAC_CHN_REG, LDAC_2_LCHN | RDAC_2_RCHN | SOFT_STEP_2WCLK}, /* Headphone powerup */ {HPHONE_STARTUP_CTRL, 0x35}, /* Left Channel DAC recons filter's positive terminal is routed to HPL */ {HPL_ROUTE_CTL, LDAC_CHNL_2_HPL}, /* Right Channel DAC recons filter's positive terminal is routed to HPR */ {HPR_ROUTE_CTL, RDAC_CHNL_2_HPR}, /* Left Channel DAC recons filter's positive terminal is routed to LOL */ {LOL_ROUTE_CTL, LDAC_CHNL_2_HPL}, /* Right Channel DAC recons filter's positive terminal is routed to LOR */ {LOR_ROUTE_CTL, RDAC_CHNL_2_HPR}, /* HPL unmute and gain 0db */ {HPL_GAIN, 0x0}, /* HPR unmute and gain 0db */ {HPR_GAIN, 0x0}, /* LOL unmute and gain 0db */ {LOL_GAIN, 0x0}, /* LOR unmute and gain 0db */ {LOR_GAIN, 0x0}, /* Unmute DAC Left and Right channels */ {DAC_MUTE_CTRL_REG, 0x00}, /* IN2_L is selected for left P */ {LMICPGA_PIN_CFG, 0x10}, /* IN2_R is selected for left M */ {LMICPGA_NIN_CFG, 0x10}, /* IN1_R is selected for right P */ {RMICPGA_PIN_CFG, 0x40}, /* IN1_L is selected for right M */ {RMICPGA_NIN_CFG, 0x10}, /* Left mic PGA unmuted */ {LMICPGA_VOL_CTRL, 0x00}, /* Right mic PGA unmuted */ {RMICPGA_VOL_CTRL, 0x00}, /* ADC volume control change by 2 gain step per ADC Word Clock */ {ADC_REG_1, 0x02}, /* Unmute ADC left and right channels */ {ADC_FGA, 0x00}, /* MICBIAS = 2.075V(CM=0.75V) generated from LDOIN */ {MICBIAS_CTRL, 0x28}, /* FIXME: these registers should be configured from user space using alsactl */ {146, 0x0a}, {147, 0x0a}, {83, 0x28}, {84, 0x28}, {187, 0x20}, {188, 0x20}, {64, 0x10}, {86, 0x80}, {94, 0x80}, {87, 0x3e}, {95, 0x3e}, {88, 0x32}, {96, 0x32}, {179, 0x68}, {88, 0x14} }; /* Left DAC_L Mixer */ static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { SOC_DAPM_SINGLE("L_DAC switch", HPL_ROUTE_CTL, 3, 1, 0), SOC_DAPM_SINGLE("IN1_L switch", HPL_ROUTE_CTL, 2, 1, 0), // SOC_DAPM_SINGLE("Left_Bypass switch", HPL_ROUTE_CTL, 1, 1, 0), }; /* Right DAC_R Mixer */ static const struct snd_kcontrol_new hpr_output_mixer_controls[] = { SOC_DAPM_SINGLE("R_DAC switch", HPR_ROUTE_CTL, 3, 1, 0), SOC_DAPM_SINGLE("IN1_R switch", HPR_ROUTE_CTL, 2, 1, 0), // SOC_DAPM_SINGLE("Right_Bypass switch", HPR_ROUTE_CTL, 1, 1, 0), }; static const struct snd_kcontrol_new lol_output_mixer_controls[] = { SOC_DAPM_SINGLE("L_DAC switch", LOL_ROUTE_CTL, 3, 1, 0), // SOC_DAPM_SINGLE("Left_Bypass switch", HPL_ROUTE_CTL, 1, 1, 0), }; static const struct snd_kcontrol_new lor_output_mixer_controls[] = { SOC_DAPM_SINGLE("R_DAC switch", LOR_ROUTE_CTL, 3, 1, 0), // SOC_DAPM_SINGLE("Right_Bypass switch", LOR_ROUTE_CTL, 1, 1, 0), }; /* Right DAC_R Mixer */ static const struct snd_kcontrol_new left_input_mixer_controls[] = { SOC_DAPM_SINGLE("IN1_L switch", LMICPGA_PIN_CFG, 6, 1, 0), SOC_DAPM_SINGLE("IN2_L switch", LMICPGA_PIN_CFG, 4, 1, 0), SOC_DAPM_SINGLE("IN3_L switch", LMICPGA_PIN_CFG, 2, 1, 0), }; static const struct snd_kcontrol_new right_input_mixer_controls[] = { SOC_DAPM_SINGLE("IN1_R switch", RMICPGA_PIN_CFG, 6, 1, 0), SOC_DAPM_SINGLE("IN2_R switch", RMICPGA_PIN_CFG, 4, 1, 0), SOC_DAPM_SINGLE("IN3_R switch", RMICPGA_PIN_CFG, 2, 1, 0), }; static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { /* Left DAC to Left Outputs */ /* dapm widget (stream domain) for left DAC */ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", DAC_CHN_REG, 7, 0), /* dapm widget (path domain) for left DAC_L Mixer */ SND_SOC_DAPM_MIXER("HPL Output Mixer", SND_SOC_NOPM, 0, 0, &hpl_output_mixer_controls[0], ARRAY_SIZE(hpl_output_mixer_controls)), SND_SOC_DAPM_PGA("HPL Power", OUT_PWR_CTL, 5, 0, NULL, 0), SND_SOC_DAPM_MIXER("LOL Output Mixer", SND_SOC_NOPM, 0, 0, &lol_output_mixer_controls[0], ARRAY_SIZE(lol_output_mixer_controls)), SND_SOC_DAPM_PGA("LOL Power", OUT_PWR_CTL, 3, 0, NULL, 0), /* Right DAC to Right Outputs */ /* dapm widget (stream domain) for right DAC */ SND_SOC_DAPM_DAC("Right DAC", "Right Playback", DAC_CHN_REG, 6, 0), /* dapm widget (path domain) for right DAC_R mixer */ SND_SOC_DAPM_MIXER("HPR Output Mixer", SND_SOC_NOPM, 0, 0, &hpr_output_mixer_controls[0], ARRAY_SIZE(hpr_output_mixer_controls)), SND_SOC_DAPM_PGA("HPR Power", OUT_PWR_CTL, 4, 0, NULL, 0), SND_SOC_DAPM_MIXER("LOR Output Mixer", SND_SOC_NOPM, 0, 0, &lor_output_mixer_controls[0], ARRAY_SIZE(lor_output_mixer_controls)), SND_SOC_DAPM_PGA("LOR Power", OUT_PWR_CTL, 2, 0, NULL, 0), SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0, &left_input_mixer_controls[0], ARRAY_SIZE(left_input_mixer_controls)), SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0, &right_input_mixer_controls[0], ARRAY_SIZE(right_input_mixer_controls)), /* Left Inputs to Left ADC */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", ADC_REG_1, 7, 0), /* Right Inputs to Right ADC */ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", ADC_REG_1, 6, 0), /* dapm widget (platform domain) name for HPLOUT */ SND_SOC_DAPM_OUTPUT("HPL"), /* dapm widget (platform domain) name for HPROUT */ SND_SOC_DAPM_OUTPUT("HPR"), /* dapm widget (platform domain) name for LOLOUT */ SND_SOC_DAPM_OUTPUT("LOL"), /* dapm widget (platform domain) name for LOROUT */ SND_SOC_DAPM_OUTPUT("LOR"), /* dapm widget (platform domain) name for LINE1L */ SND_SOC_DAPM_INPUT("IN1_L"), /* dapm widget (platform domain) name for LINE1R */ SND_SOC_DAPM_INPUT("IN1_R"), /* dapm widget (platform domain) name for LINE2L */ SND_SOC_DAPM_INPUT("IN2_L"), /* dapm widget (platform domain) name for LINE2R */ SND_SOC_DAPM_INPUT("IN2_R"), /* dapm widget (platform domain) name for LINE3L */ SND_SOC_DAPM_INPUT("IN3_L"), /* dapm widget (platform domain) name for LINE3R */ SND_SOC_DAPM_INPUT("IN3_R"), }; static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { /* ******** Left Output ******** */ {"HPL Output Mixer", "L_DAC switch", "Left DAC"}, {"HPL Output Mixer", "IN1_L switch", "IN1_L"}, //{"HPL Output Mixer", "Left_Bypass switch", "Left_Bypass"}, {"HPL Power", NULL, "HPL Output Mixer"}, {"HPL", NULL, "HPL Power"}, {"LOL Output Mixer", "L_DAC switch", "Left DAC"}, // {"LOL Output Mixer", "Left_Bypass switch", "Left_Bypass"}, {"LOL Power", NULL, "LOL Output Mixer"}, {"LOL", NULL, "LOL Power"}, /* ******** Right Output ******** */ {"HPR Output Mixer", "R_DAC switch", "Right DAC"}, {"HPR Output Mixer", "IN1_R switch", "IN1_R"}, //{"HPR Output Mixer", "Right_Bypass switch", "Right_Bypass"}, {"HPR Power", NULL, "HPR Output Mixer"}, {"HPR", NULL, "HPR Power"}, {"LOR Output Mixer", "R_DAC switch", "Right DAC"}, // {"LOR Output Mixer", "Right_Bypass switch", "Right_Bypass"}, {"LOR Power", NULL, "LOR Output Mixer"}, {"LOR", NULL, "LOR Power"}, /* ******** Left input ******** */ {"Left Input Mixer", "IN1_L switch", "IN1_L"}, {"Left Input Mixer", "IN2_L switch", "IN2_L"}, {"Left Input Mixer", "IN3_L switch", "IN3_L"}, //{"Left_Bypass", NULL, "Left Input Mixer"}, {"Left ADC", NULL, "Left Input Mixer"}, /* ******** Right Input ******** */ {"Right Input Mixer", "IN1_R switch", "IN1_R"}, {"Right Input Mixer", "IN2_R switch", "IN2_R"}, {"Right Input Mixer", "IN3_R switch", "IN3_R"}, // {"Right_Bypass", NULL, "Right Input Mixer"}, {"Right ADC", NULL, "Right Input Mixer"}, }; /* ***************************************************************************** * Function Definitions ***************************************************************************** */ static void aic32x4_change_page(struct snd_soc_codec *codec, u8 new_page) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); snd_soc_write(codec, AIC32X4_PSEL, new_page); aic32x4->page_no = new_page; } static void aic32x4_write(struct snd_soc_codec *codec, u16 reg, u8 val) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u8 page = reg / 128; u8 fixed_reg = reg % 128; printk("%s: reg=%d, val=0x%02x\n", __func__, reg, val); /* A write to AIC32X4_PSEL is really a non-explicit page change */ if (reg == AIC32X4_PSEL) { aic32x4_change_page(codec, val); return; } if (aic32x4->page_no != page) { aic32x4_change_page(codec, page); } snd_soc_write(codec, fixed_reg, val); } static u8 aic32x4_read(struct snd_soc_codec *codec, u16 reg) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u8 page = reg / 128; u8 fixed_reg = reg % 128; if (aic32x4->page_no != page) { aic32x4_change_page(codec, page); } return snd_soc_read(codec, fixed_reg); } static int snd_soc_info_volsw_2r_aic32x4(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { int mask = (kcontrol->private_value >> 12) & 0xff; uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; uinfo->value.integer.min = 0; uinfo->value.integer.max = mask; return 0; } /* *---------------------------------------------------------------------------- * Function : snd_soc_get_volsw_2r_aic32x4 * Purpose : Callback to get the value of a double mixer control that spans * two registers. * *---------------------------------------------------------------------------- */ int snd_soc_get_volsw_2r_aic32x4(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); int reg = kcontrol->private_value & AIC32X4_8BITS_MASK; int reg2 = (kcontrol->private_value >> 24) & AIC32X4_8BITS_MASK; int mask; int shift; unsigned short val, val2; if (!strcmp(kcontrol->id.name, "PCM Playback Volume")) { mask = AIC32X4_8BITS_MASK; shift = 0; } else if ((!strcmp(kcontrol->id.name, "HP Driver Gain")) || (!strcmp(kcontrol->id.name, "LO Driver Gain"))) { mask = 0x3F; shift = 0; } else if (!strcmp(kcontrol->id.name, "PGA Capture Volume")) { mask = 0x7F; shift = 0; } else { printk("Invalid kcontrol name\n"); return -1; } val = (snd_soc_read(codec, reg) >> shift) & mask; val2 = (snd_soc_read(codec, reg2) >> shift) & mask; if (!strcmp(kcontrol->id.name, "PCM Playback Volume")) { ucontrol->value.integer.value[0] = (val <= 48) ? (val + 127) : (val - 129); ucontrol->value.integer.value[1] = (val2 <= 48) ? (val2 + 127) : (val2 - 129); } else if ((!strcmp(kcontrol->id.name, "HP Driver Gain")) || (!strcmp(kcontrol->id.name, "LO Driver Gain"))) { ucontrol->value.integer.value[0] = (val <= 29) ? (val + 6) : (val - 58); ucontrol->value.integer.value[1] = (val2 <= 29) ? (val2 + 6) : (val2 - 58); } else if (!strcmp(kcontrol->id.name, "PGA Capture Volume")) { ucontrol->value.integer.value[0] = (val <= 38) ? (val + 25) : (val - 103); ucontrol->value.integer.value[1] = (val2 <= 38) ? (val2 + 25) : (val2 - 103); } return 0; } /* *---------------------------------------------------------------------------- * Function : snd_soc_put_volsw_2r_aic32x4 * Purpose : Callback to set the value of a double mixer control that spans * two registers. * *---------------------------------------------------------------------------- */ int snd_soc_put_volsw_2r_aic32x4(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); int reg = kcontrol->private_value & AIC32X4_8BITS_MASK; int reg2 = (kcontrol->private_value >> 24) & AIC32X4_8BITS_MASK; int err; unsigned short val, val2, val_mask; val = ucontrol->value.integer.value[0]; val2 = ucontrol->value.integer.value[1]; if (!strcmp(kcontrol->id.name, "PCM Playback Volume")) { val = (val >= 127) ? (val - 127) : (val + 129); val2 = (val2 >= 127) ? (val2 - 127) : (val2 + 129); val_mask = AIC32X4_8BITS_MASK; /* 8 bits */ } else if ((!strcmp(kcontrol->id.name, "HP Driver Gain")) || (!strcmp(kcontrol->id.name, "LO Driver Gain"))) { val = (val >= 6) ? (val - 6) : (val + 58); val2 = (val2 >= 6) ? (val2 - 6) : (val2 + 58); val_mask = 0x3F; /* 6 bits */ } else if (!strcmp(kcontrol->id.name, "PGA Capture Volume")) { val = (val >= 25) ? (val - 25) : (val + 103); val2 = (val2 >= 25) ? (val2 - 25) : (val2 + 103); val_mask = 0x7F; /* 7 bits */ } else { printk("Invalid control name\n"); return -1; } if ((err = snd_soc_update_bits(codec, reg, val_mask, val)) < 0) { printk("Error while updating bits\n"); return err; } err = snd_soc_update_bits(codec, reg2, val_mask, val2); return err; } /* *---------------------------------------------------------------------------- * Function : aic32x4_get_divs * Purpose : This function is to get required divisor from the "aic32x4_divs" * table. * *---------------------------------------------------------------------------- */ static inline int aic32x4_get_divs(int mclk, int rate) { int i; for (i = 0; i < ARRAY_SIZE(aic32x4_divs); i++) { if ((aic32x4_divs[i].rate == rate) && (aic32x4_divs[i].mclk == mclk)) { return i; } } printk("Master clock and sample rate is not supported\n"); return -EINVAL; } static int aic32x4_add_widgets(struct snd_soc_codec *codec) { snd_soc_dapm_new_controls(codec, aic32x4_dapm_widgets, ARRAY_SIZE(aic32x4_dapm_widgets)); snd_soc_dapm_add_routes(codec, aic32x4_dapm_routes, ARRAY_SIZE(aic32x4_dapm_routes)); snd_soc_dapm_new_widgets(codec); return 0; } static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); printk("%s\n", __func__); switch (freq) { case AIC32X4_FREQ_12000000: case AIC32X4_FREQ_24000000: case AIC32X4_FREQ_25000000: aic32x4->sysclk = freq; return 0; } printk("Invalid frequency to set DAI system clock\n"); return -EINVAL; } static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u8 iface_reg; printk("%s\n", __func__); iface_reg = aic32x4_read(codec, INTERFACE_SET_REG_1); iface_reg = iface_reg & ~(3 << 6 | 3 << 2); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: aic32x4->master = 1; iface_reg |= BIT_CLK_MASTER | WORD_CLK_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: aic32x4->master = 0; break; default: printk("Invalid DAI master/slave interface\n"); return -EINVAL; } /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: break; case SND_SOC_DAIFMT_DSP_A: iface_reg |= (AIC32X4_DSP_MODE << CLK_REG_3_SHIFT); break; case SND_SOC_DAIFMT_RIGHT_J: iface_reg |= (AIC32X4_RIGHT_JUSTIFIED_MODE << CLK_REG_3_SHIFT); break; case SND_SOC_DAIFMT_LEFT_J: iface_reg |= (AIC32X4_LEFT_JUSTIFIED_MODE << CLK_REG_3_SHIFT); break; default: printk("Invalid DAI interface format\n"); return -EINVAL; } aic32x4_write(codec, INTERFACE_SET_REG_1, iface_reg); return 0; } static int aic32x4_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); int i, j; u8 data; printk("%s\n", __func__); i = aic32x4_get_divs(aic32x4->sysclk, params_rate(params)); if (i < 0) { printk("sampling rate not supported\n"); return i; } /* We will fix R value to 1 and will make P & J=K.D as varialble */ /* Setting P & R values */ aic32x4_write(codec, CLK_REG_2, ((aic32x4_divs[i].p_val << 4) | 0x01)); /* J value */ aic32x4_write(codec, CLK_REG_3, aic32x4_divs[i].pll_j); /* MSB & LSB for D value */ aic32x4_write(codec, CLK_REG_4, (aic32x4_divs[i].pll_d >> 8)); aic32x4_write(codec, CLK_REG_5, (aic32x4_divs[i].pll_d & AIC32X4_8BITS_MASK)); /* NDAC divider value */ aic32x4_write(codec, NDAC_CLK_REG_6, aic32x4_divs[i].ndac); /* MDAC divider value */ aic32x4_write(codec, MDAC_CLK_REG_7, aic32x4_divs[i].mdac); /* DOSR MSB & LSB values */ aic32x4_write(codec, DAC_OSR_MSB, aic32x4_divs[i].dosr >> 8); aic32x4_write(codec, DAC_OSR_LSB, (aic32x4_divs[i].dosr & AIC32X4_8BITS_MASK)); /* NADC divider value */ aic32x4_write(codec, NADC_CLK_REG_8, aic32x4_divs[i].nadc); /* MADC divider value */ aic32x4_write(codec, MADC_CLK_REG_9, aic32x4_divs[i].madc); /* AOSR value */ aic32x4_write(codec, ADC_OSR_REG, aic32x4_divs[i].aosr); /* BCLK N divider */ aic32x4_write(codec, CLK_REG_11, aic32x4_divs[i].blck_N); data = aic32x4_read(codec, INTERFACE_SET_REG_1); data = data & ~(3 << 4); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; case SNDRV_PCM_FORMAT_S20_3LE: data |= (AIC32X4_WORD_LEN_20BITS << DAC_OSR_MSB_SHIFT); break; case SNDRV_PCM_FORMAT_S24_LE: data |= (AIC32X4_WORD_LEN_24BITS << DAC_OSR_MSB_SHIFT); break; case SNDRV_PCM_FORMAT_S32_LE: data |= (AIC32X4_WORD_LEN_32BITS << DAC_OSR_MSB_SHIFT); break; } aic32x4_write(codec, INTERFACE_SET_REG_1, data); for (j = 0; j < NO_FEATURE_REGS; j++) { aic32x4_write(codec, aic32x4_divs[i].codec_specific_regs[j].reg_offset, aic32x4_divs[i].codec_specific_regs[j].reg_val); } return 0; } static int aic32x4_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u8 dac_reg; printk("%s\n", __func__); dac_reg = aic32x4_read(codec, DAC_MUTE_CTRL_REG) & ~MUTE_ON; if (mute) { aic32x4_write(codec, DAC_MUTE_CTRL_REG, dac_reg | MUTE_ON); } else { aic32x4_write(codec, DAC_MUTE_CTRL_REG, dac_reg); } return 0; } static int aic32x4_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u8 value; char *dbgmap[] = {"SND_SOC_BIAS_OFF", "SND_SOC_BIAS_STANDBY", "SND_SOC_BIAS_PREPARE", "SND_SOC_BIAS_ON"}; printk("%s: %s\n", __func__, dbgmap[level]); switch (level) { /* full On */ case SND_SOC_BIAS_ON: /* all power is driven by DAPM system */ if (aic32x4->master) { /* Switch on PLL */ value = aic32x4_read(codec, CLK_REG_2); aic32x4_write(codec, CLK_REG_2, (value | ENABLE_PLL)); /* Switch on NDAC Divider */ value = aic32x4_read(codec, NDAC_CLK_REG_6); aic32x4_write(codec, NDAC_CLK_REG_6, value | ENABLE_NDAC); /* Switch on MDAC Divider */ value = aic32x4_read(codec, MDAC_CLK_REG_7); aic32x4_write(codec, MDAC_CLK_REG_7, value | ENABLE_MDAC); /* Switch on NADC Divider */ value = aic32x4_read(codec, NADC_CLK_REG_8); aic32x4_write(codec, NADC_CLK_REG_8, value | ENABLE_MDAC); /* Switch on MADC Divider */ value = aic32x4_read(codec, MADC_CLK_REG_9); aic32x4_write(codec, MADC_CLK_REG_9, value | ENABLE_MDAC); /* Switch on BCLK_N Divider */ value = aic32x4_read(codec, CLK_REG_11); aic32x4_write(codec, CLK_REG_11, value | ENABLE_BCLK); } break; /* partial On */ case SND_SOC_BIAS_PREPARE: break; /* Off, with power */ case SND_SOC_BIAS_STANDBY: /* * all power is driven by DAPM system, * so output power is safe if bypass was set */; if (aic32x4->master) { /* Switch off PLL */ value = aic32x4_read(codec, CLK_REG_2); aic32x4_write(codec, CLK_REG_2, (value & ~ENABLE_PLL)); /* Switch off NDAC Divider */ value = aic32x4_read(codec, NDAC_CLK_REG_6); aic32x4_write(codec, NDAC_CLK_REG_6, value & ~ENABLE_NDAC); /* Switch off MDAC Divider */ value = aic32x4_read(codec, MDAC_CLK_REG_7); aic32x4_write(codec, MDAC_CLK_REG_7, value & ~ENABLE_MDAC); /* Switch off NADC Divider */ value = aic32x4_read(codec, NADC_CLK_REG_8); aic32x4_write(codec, NADC_CLK_REG_8, value & ~ENABLE_NDAC); /* Switch off MADC Divider */ value = aic32x4_read(codec, MADC_CLK_REG_9); aic32x4_write(codec, MADC_CLK_REG_9, value & ~ENABLE_MDAC); value = aic32x4_read(codec, CLK_REG_11); /* Switch off BCLK_N Divider */ aic32x4_write(codec, CLK_REG_11, value & ~ENABLE_BCLK); } break; /* Off, without power */ case SND_SOC_BIAS_OFF: /* TODO: implement a real shutdown and use a cache for registers */ break; } codec->bias_level = level; return 0; } #define AIC32X4_RATES SNDRV_PCM_RATE_8000_192000 #define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops aic32x4_ops = { .hw_params = aic32x4_hw_params, .digital_mute = aic32x4_mute, .set_fmt = aic32x4_set_dai_fmt, .set_sysclk = aic32x4_set_dai_sysclk, }; static struct snd_soc_dai_driver aic32x4_dai = { .name = "tlv320aic32x4-hifi", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = AIC32X4_RATES, .formats = AIC32X4_FORMATS,}, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = AIC32X4_RATES, .formats = AIC32X4_FORMATS,}, .ops = &aic32x4_ops, .symmetric_rates = 1, }; static int aic32x4_suspend(struct snd_soc_codec *codec, pm_message_t state) { printk("%s\n", __func__); aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static int aic32x4_resume(struct snd_soc_codec *codec) { printk("%s\n", __func__); /* * TODO: Since SND_SOC_BIAS_OFF is not effective there's no need to * sync with a cache here. */ aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } static int aic32x4_probe(struct snd_soc_codec *codec) { int ret, i; printk("%s\n", __func__); ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } // aic32x4_reset(codec); /* TODO: initial configuration of registers -> remove since this is machine specific */ for (i=0; i<ARRAY_SIZE(aic32x4_reg_init); i++) aic32x4_write(codec, aic32x4_reg_init[i].reg_offset, aic32x4_reg_init[i].reg_val); aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); snd_soc_add_controls(codec, aic32x4_snd_controls, ARRAY_SIZE(aic32x4_snd_controls)); aic32x4_add_widgets(codec); return 0; } static int aic32x4_remove(struct snd_soc_codec *codec) { aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { .probe = aic32x4_probe, .remove = aic32x4_remove, .suspend = aic32x4_suspend, .resume = aic32x4_resume, .set_bias_level = aic32x4_set_bias_level, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int aic32x4_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct aic32x4_priv *aic32x4; int ret; aic32x4 = kzalloc(sizeof(struct aic32x4_priv), GFP_KERNEL); if (aic32x4 == NULL) return -ENOMEM; i2c_set_clientdata(i2c, aic32x4); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic32x4, &aic32x4_dai, 1); if (ret < 0) kfree(aic32x4); return ret; } static __devexit int aic32x4_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); kfree(i2c_get_clientdata(client)); return 0; } /* TODO: two possible models: tlv320aic3254/tlv320aic3204 (see other tlv drivers) */ static const struct i2c_device_id aic32x4_i2c_id[] = { { "tlv320aic32x4", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); static struct i2c_driver aic32x4_i2c_driver = { .driver = { .name = "tlv320aic32x4-codec", .owner = THIS_MODULE, }, .probe = aic32x4_i2c_probe, .remove = __devexit_p(aic32x4_i2c_remove), .id_table = aic32x4_i2c_id, }; #endif static int __init aic32x4_modinit(void) { int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&aic32x4_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register aic32x4 I2C driver: %d\n", ret); } #endif return ret; } module_init(aic32x4_modinit); static void __exit aic32x4_exit(void) { #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&aic32x4_i2c_driver); #endif } module_exit(aic32x4_exit); MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver"); MODULE_AUTHOR("Javier Martin <javier.martin(a)vista-silicon.com>"); MODULE_LICENSE("GPL"); -- Javier Martin Vista Silicon S.L. CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

1 0

[alsa-devel] [PATCH] fix one memory leak in sound jack
by Lu Guanqun 21 Feb '11

21 Feb '11

Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- sound/core/jack.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/core/jack.c b/sound/core/jack.c index 4902ae5..53b53e9 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -141,6 +141,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, fail_input: input_free_device(jack->input_dev); + kfree(jack->id); kfree(jack); return err; } -- 1.7.2.3

3 2

Re: [alsa-devel] pci/hda/patch_via.c - broken between 2.6.36 and 2.6.37 (bisected)
by Raymond Yau 21 Feb '11

21 Feb '11

2011/2/20 Wolfgang Scheicher <wolfgang.scheicher(a)gmail.com> > On Sunday 20 February 2011 14:27:52 you wrote: > > 2011/2/20 Wolfgang Scheicher <wolfgang.scheicher(a)gmail.com> > > > >> After the mentioned Kernel upgrade quite a view things went wrong on my > >> hardware ( http://www.asrock.com/mb/overview.asp?Model=890GM%20Pro3 ): > >> > >> * Microphone doesn't work anymore. neither the front mic nor the rear > >> one. Tried all thinkable settings. > >> > >> * Increasing Microphone levels leads to some strange internal feedback. > >> That > >> means, at some point when the feedback starts, it just starts, but i > have > >> no > >> clue what audio source did actually trigger it. > >> > >> * Touching the mixer channel "Smart 5.1" in alsamixer makes alsamicer > >> crash. > > > > Refer to the user manual , the motherboard has 5 audio jacks in rear > panel, > > > > 7.1 need retasking of Line in (blue jacks) as side > > > > Audio Output Channels Front Speaker Rear Speaker Central / Bass Line In > > (No. 9) (No. 6) (No. 5) (No. 8) > > 2 V -- -- -- > > 4 V V -- -- > > 6 V V V -- > > 8 V V V V > > Forgive me, but i don't understand. > I hope this doesn't mean i have to solder on my Mainboard if i intend to > use a > newer kernel. > > If there is something i can do, please explain in more detail. > > -- > Worf > A better way is to disable the creation of smart5.1 switch when (cfg->line-outs != 1 ) and use "surround71 switch" to retask "line in" to side

1 0

Re: [alsa-devel] pci/hda/patch_via.c - broken between 2.6.36 and 2.6.37 (bisected)
by Raymond Yau 21 Feb '11

21 Feb '11

2011/2/20 Wolfgang Scheicher <wolfgang.scheicher(a)gmail.com> > On Sunday 20 February 2011 14:27:52 you wrote: > > 2011/2/20 Wolfgang Scheicher <wolfgang.scheicher(a)gmail.com> > > > >> After the mentioned Kernel upgrade quite a view things went wrong on my > >> hardware ( http://www.asrock.com/mb/overview.asp?Model=890GM%20Pro3 ): > >> > >> * Microphone doesn't work anymore. neither the front mic nor the rear > >> one. Tried all thinkable settings. > >> > >> * Increasing Microphone levels leads to some strange internal feedback. > >> That > >> means, at some point when the feedback starts, it just starts, but i > have > >> no > >> clue what audio source did actually trigger it. > >> > >> * Touching the mixer channel "Smart 5.1" in alsamixer makes alsamicer > >> crash. > > > > Refer to the user manual , the motherboard has 5 audio jacks in rear > panel, > > > > 7.1 need retasking of Line in (blue jacks) as side > > > > Audio Output Channels Front Speaker Rear Speaker Central / Bass Line In > > (No. 9) (No. 6) (No. 5) (No. 8) > > 2 V -- -- -- > > 4 V V -- -- > > 6 V V V -- > > 8 V V V V > > Forgive me, but i don't understand. > I hope this doesn't mean i have to solder on my Mainboard if i intend to > use a > newer kernel. > > If there is something i can do, please explain in more detail. > > -- > Worf > http://git.alsa-project.org/?p=alsa-kernel.git;a=commit;h=1564b2878f5cf160f… Smart 5.1 is for 3-jacks model, to reuse input pins as outputs. While off, they act as "line out" / "line in" / "mic in". While on, they acts as "line out" / "back left/right" / "center/lfe". If the driver use "Smart5.1 switch" for those 5-jacks model, you will need to modify the function is_smart51_pins() static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin) { const struct auto_pin_cfg *cfg = &spec->autocfg; int i; if (cfg->line_outs == 4) return 0; for (i = 0; i < cfg->num_inputs; i++) { if (pin == cfg->inputs[i].pin) { switch(cfg->inputs[i].type){ case AUTO_PIN_LINE_IN: return (cfg->line_outs == 3 || cfg->line_outs == 1); case AUTO_PIN_MIC: if ( strncmp(hda_get_autocfg_input_label(spec->codec,cfg,i),"Rear Mic",8) == 0 ) return (cfg->line_outs == 1); } } } return 0; }

1 0

Re: [alsa-devel] [PATCH] - au88x0 - add Playback Volume to 10 bands Equalizer Controls
by Takashi Iwai 20 Feb '11

20 Feb '11

At Sun, 20 Feb 2011 16:08:26 +0800, Raymond Yau wrote: > > - add " Playback Volume" to 10 bands Equalizer Controls of au88x0 so that > alsa-lib won't regard them as "Capture Volume" Thanks, applied now. Takashi

1 0

[alsa-devel] [PATCH] sound azt3328: hook up new emulated AC97 on AC97 patch side
by Andreas Mohr 19 Feb '11

19 Feb '11

ChangeLog: Make newly created AC97 emulation of azt3328 known to the AC97 layer side. - relocate common functions to the top (due to definition after use) - rename control names - adjust 3D settings to the card's custom layout of this register Signed-off-by: Andreas Mohr <andi(a)lisas.de> --- Patch created on -rc3, tested and checkpatch.pl'd. GIT history seems clean, should thus apply easily. Thanks! Index: linux-2.6/sound/pci/ac97/ac97_codec.c =================================================================== --- linux-2.6.orig/sound/pci/ac97/ac97_codec.c 2011-02-16 23:18:54.000000000 +0100 +++ linux-2.6/sound/pci/ac97/ac97_codec.c 2011-02-16 23:24:12.000000000 +0100 @@ -71,6 +71,12 @@ { 0x414b4d00, 0xffffff00, "Asahi Kasei", NULL, NULL }, { 0x414c4300, 0xffffff00, "Realtek", NULL, NULL }, { 0x414c4700, 0xffffff00, "Realtek", NULL, NULL }, +/* + * This is an _inofficial_ Aztech Labs entry + * (value might differ from unknown official Aztech ID), + * currently used by the AC97 emulation of the almost-AC97 PCI168 card. + */ +{ 0x415a5400, 0xffffff00, "Aztech Labs (emulated)", NULL, NULL }, { 0x434d4900, 0xffffff00, "C-Media Electronics", NULL, NULL }, { 0x43525900, 0xffffff00, "Cirrus Logic", NULL, NULL }, { 0x43585400, 0xffffff00, "Conexant", NULL, NULL }, @@ -127,6 +133,7 @@ { 0x414c4781, 0xffffffff, "ALC658D", NULL, NULL }, /* already patched */ { 0x414c4780, 0xfffffff0, "ALC658", patch_alc655, NULL }, { 0x414c4790, 0xfffffff0, "ALC850", patch_alc850, NULL }, +{ 0x415a5401, 0xffffffff, "AZF3328", patch_aztech_azf3328, NULL }, { 0x434d4941, 0xffffffff, "CMI9738", patch_cm9738, NULL }, { 0x434d4961, 0xffffffff, "CMI9739", patch_cm9739, NULL }, { 0x434d4969, 0xffffffff, "CMI9780", patch_cm9780, NULL }, Index: linux-2.6/sound/pci/ac97/ac97_patch.c =================================================================== --- linux-2.6.orig/sound/pci/ac97/ac97_patch.c 2011-02-16 23:18:54.000000000 +0100 +++ linux-2.6/sound/pci/ac97/ac97_patch.c 2011-02-16 23:19:03.000000000 +0100 @@ -222,6 +222,47 @@ return is_surround_on(ac97); } +/* find a mixer control element with the given name */ +static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97, + const char *name) +{ + struct snd_ctl_elem_id id; + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(id.name, name); + return snd_ctl_find_id(ac97->bus->card, &id); +} + +/* create a virtual master control and add slaves */ +static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, + const unsigned int *tlv, const char **slaves) +{ + struct snd_kcontrol *kctl; + const char **s; + int err; + + kctl = snd_ctl_make_virtual_master(name, tlv); + if (!kctl) + return -ENOMEM; + err = snd_ctl_add(ac97->bus->card, kctl); + if (err < 0) + return err; + + for (s = slaves; *s; s++) { + struct snd_kcontrol *sctl; + + sctl = snd_ac97_find_mixer_ctl(ac97, *s); + if (!sctl) { + snd_printdd("Cannot find slave %s, skipped\n", *s); + continue; + } + err = snd_ctl_add_slave(kctl, sctl); + if (err < 0) + return err; + } + return 0; +} + /* The following snd_ac97_ymf753_... items added by David Shust (dshust(a)shustring.com) */ /* Modified for YMF743 by Keita Maehara <maehara(a)debian.org> */ @@ -2940,6 +2981,49 @@ return 0; } +static int patch_aztech_azf3328_specific(struct snd_ac97 *ac97) +{ + struct snd_kcontrol *kctl_3d_center = + snd_ac97_find_mixer_ctl(ac97, "3D Control - Center"); + struct snd_kcontrol *kctl_3d_depth = + snd_ac97_find_mixer_ctl(ac97, "3D Control - Depth"); + + /* + * 3D register is different from AC97 standard layout + * (also do some renaming, to resemble Windows driver naming) + */ + if (kctl_3d_center) { + kctl_3d_center->private_value = + AC97_SINGLE_VALUE(AC97_3D_CONTROL, 1, 0x07, 0); + snd_ac97_rename_vol_ctl(ac97, + "3D Control - Center", "3D Control - Width" + ); + } + if (kctl_3d_depth) + kctl_3d_depth->private_value = + AC97_SINGLE_VALUE(AC97_3D_CONTROL, 8, 0x03, 0); + + /* Aztech Windows driver calls the + equivalent control "Modem Playback", thus rename it: */ + snd_ac97_rename_vol_ctl(ac97, + "Master Mono Playback", "Modem Playback" + ); + snd_ac97_rename_vol_ctl(ac97, + "Headphone Playback", "FM Synth Playback" + ); + + return 0; +} + +static const struct snd_ac97_build_ops patch_aztech_azf3328_ops = { + .build_specific = patch_aztech_azf3328_specific +}; + +static int patch_aztech_azf3328(struct snd_ac97 *ac97) +{ + ac97->build_ops = &patch_aztech_azf3328_ops; + return 0; +} /* * C-Media CM97xx codecs @@ -3371,47 +3455,6 @@ NULL }; -/* find a mixer control element with the given name */ -static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97, - const char *name) -{ - struct snd_ctl_elem_id id; - memset(&id, 0, sizeof(id)); - id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(id.name, name); - return snd_ctl_find_id(ac97->bus->card, &id); -} - -/* create a virtual master control and add slaves */ -static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, - const unsigned int *tlv, const char **slaves) -{ - struct snd_kcontrol *kctl; - const char **s; - int err; - - kctl = snd_ctl_make_virtual_master(name, tlv); - if (!kctl) - return -ENOMEM; - err = snd_ctl_add(ac97->bus->card, kctl); - if (err < 0) - return err; - - for (s = slaves; *s; s++) { - struct snd_kcontrol *sctl; - - sctl = snd_ac97_find_mixer_ctl(ac97, *s); - if (!sctl) { - snd_printdd("Cannot find slave %s, skipped\n", *s); - continue; - } - err = snd_ctl_add_slave(kctl, sctl); - if (err < 0) - return err; - } - return 0; -} - static int patch_vt1616_specific(struct snd_ac97 * ac97) { struct snd_kcontrol *kctl;

2 5

[alsa-devel] [PATCH] sound azt3328: add custom AC97 semi-emulation, to be able to use standard ALSA AC97 layer
by Andreas Mohr 19 Feb '11

19 Feb '11

ChangeLog: Make use of the very flexible ALSA ac97 layer (hooks for custom I/O!) on this weird AC97 copycat hardware, via semi-extended I/O translation/emulation. Some 5kB binary/loaded size saved (well... additional huge AC97 module penalty not factored in, of course ;-P). Given that the driver previously had 20kB that's not bad, but the much more important thing is to have AC97 layer stress-tested with a thoroughly weird AC97 copycat (or, simply put, if it were not for this AC97 test aspect, this effort would merely have been a nut job ;). The semi-emulated AC97 constellation appears to be working fine. Planning to remove old custom mixer parts if AC97-based operation holds water. Signed-off-by: Andreas Mohr <andi(a)lisas.de> --- Patch created on -rc3, tested and checkpatch.pl'd. GIT history seems clean, should thus apply easily. Thanks! Index: linux-2.6/sound/pci/azt3328.c =================================================================== --- linux-2.6.orig/sound/pci/azt3328.c 2011-02-16 23:18:59.000000000 +0100 +++ linux-2.6/sound/pci/azt3328.c 2011-02-16 23:19:02.000000000 +0100 @@ -1,6 +1,5 @@ -/* - * azt3328.c - driver for Aztech AZF3328 based soundcards (e.g. PCI168). - * Copyright (C) 2002, 2005 - 2010 by Andreas Mohr <andi AT lisas.de> +/* azt3328.c - driver for Aztech AZF3328 based soundcards (e.g. PCI168). + * Copyright (C) 2002, 2005 - 2011 by Andreas Mohr <andi AT lisas.de> * * Framework borrowed from Bart Hartgers's als4000.c. * Driver developed on PCI168 AP(W) version (PCI rev. 10, subsystem ID 1801), @@ -66,6 +65,13 @@ * addresses illegally. So far unfortunately it looks like the very flexible * ALSA AC97 support is still not enough to easily compensate for such a * grave layout violation despite all tweaks and quirks mechanisms it offers. + * Well, not quite: now ac97 layer is much improved (bus-specific ops!), + * thus I was able to implement support - it's actually working quite well. + * An interesting item might be Aztech AMR 2800-W, since it's an AC97 + * modem card which might reveal the Aztech-specific codec ID which + * we might want to pretend, too. Dito PCI168's brother, PCI368, + * where the advertising datasheet says it's AC97-based and has a + * Digital Enhanced Game Port. * - builtin genuine OPL3 - verified to work fine, 20080506 * - full duplex 16bit playback/record at independent sampling rate * - MPU401 (+ legacy address support, claimed by one official spec sheet) @@ -189,6 +195,19 @@ #include <sound/mpu401.h> #include <sound/opl3.h> #include <sound/initval.h> +/* + * For now, use AC97 layer in case CONFIG_SND_AC97_CODEC is available. + * If our AC97 emulation turns out to work nicely, + * then make it unconditional and remove old custom AC97 crap + * (and Kconfig-select SND_AC97_CODEC). + */ +#if defined(CONFIG_SND_AC97_CODEC) || \ + (defined(MODULE) && defined(CONFIG_SND_AC97_CODEC_MODULE)) + #define AZF_USE_AC97_LAYER 1 +#endif +#ifdef AZF_USE_AC97_LAYER +#include <sound/ac97_codec.h> +#endif #include "azt3328.h" MODULE_AUTHOR("Andreas Mohr <andi AT lisas.de>"); @@ -328,6 +347,10 @@ /* playback, recording and I2S out codecs */ struct snd_azf3328_codec_data codecs[3]; +#ifdef AZF_USE_AC97_LAYER + struct snd_ac97 *ac97; +#endif + struct snd_card *card; struct snd_rawmidi *rmidi; @@ -506,7 +529,7 @@ #define AZF_MUTE_BIT 0x80 static bool -snd_azf3328_mixer_set_mute(const struct snd_azf3328 *chip, +snd_azf3328_mixer_mute_control(const struct snd_azf3328 *chip, unsigned reg, bool do_mute ) { @@ -521,6 +544,323 @@ return (do_mute) ? !updated : updated; } +static inline bool +snd_azf3328_mixer_mute_control_master(const struct snd_azf3328 *chip, + bool do_mute +) +{ + return snd_azf3328_mixer_mute_control( + chip, + IDX_MIXER_PLAY_MASTER, + do_mute + ); +} + +static inline bool +snd_azf3328_mixer_mute_control_pcm(const struct snd_azf3328 *chip, + bool do_mute +) +{ + return snd_azf3328_mixer_mute_control( + chip, + IDX_MIXER_WAVEOUT, + do_mute + ); +} + +static inline void +snd_azf3328_mixer_reset(const struct snd_azf3328 *chip) +{ + /* reset (close) mixer: + * first mute master volume, then reset + */ + snd_azf3328_mixer_mute_control_master(chip, 1); + snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000); +} + +#ifdef AZF_USE_AC97_LAYER + +static inline void +snd_azf3328_mixer_ac97_map_unsupported(unsigned short reg, const char *mode) +{ + /* need to add some more or less clever emulation? */ + printk(KERN_WARNING + "azt3328: missing %s emulation for AC97 register 0x%02x!\n", + mode, reg); +} + +/* + * Need to have _special_ AC97 mixer hardware register index mapper, + * to compensate for the issue of a rather AC97-incompatible hardware layout. + */ +#define AZF_REG_MASK 0x3f +#define AZF_AC97_REG_UNSUPPORTED 0x8000 +#define AZF_AC97_REG_REAL_IO_READ 0x4000 +#define AZF_AC97_REG_REAL_IO_WRITE 0x2000 +#define AZF_AC97_REG_REAL_IO_RW \ + (AZF_AC97_REG_REAL_IO_READ | AZF_AC97_REG_REAL_IO_WRITE) +#define AZF_AC97_REG_EMU_IO_READ 0x0400 +#define AZF_AC97_REG_EMU_IO_WRITE 0x0200 +#define AZF_AC97_REG_EMU_IO_RW \ + (AZF_AC97_REG_EMU_IO_READ | AZF_AC97_REG_EMU_IO_WRITE) +static unsigned short +snd_azf3328_mixer_ac97_map_reg_idx(unsigned short reg) +{ + static const struct { + unsigned short azf_reg; + } azf_reg_mapper[] = { + /* Especially when taking into consideration + * mono/stereo-based sequence of azf vs. AC97 control series, + * it's quite obvious that azf simply got rid + * of the AC97_HEADPHONE control at its intended offset, + * thus shifted _all_ controls by one, + * and _then_ simply added it as an FMSYNTH control at the end, + * to make up for the offset. + * This means we'll have to translate indices here as + * needed and then do some tiny AC97 patch action + * (snd_ac97_rename_vol_ctl() etc.) - that's it. + */ + { /* AC97_RESET */ IDX_MIXER_RESET + | AZF_AC97_REG_REAL_IO_WRITE + | AZF_AC97_REG_EMU_IO_READ }, + { /* AC97_MASTER */ IDX_MIXER_PLAY_MASTER }, + /* note large shift: AC97_HEADPHONE to IDX_MIXER_FMSYNTH! */ + { /* AC97_HEADPHONE */ IDX_MIXER_FMSYNTH }, + { /* AC97_MASTER_MONO */ IDX_MIXER_MODEMOUT }, + { /* AC97_MASTER_TONE */ IDX_MIXER_BASSTREBLE }, + { /* AC97_PC_BEEP */ IDX_MIXER_PCBEEP }, + { /* AC97_PHONE */ IDX_MIXER_MODEMIN }, + { /* AC97_MIC */ IDX_MIXER_MIC }, + { /* AC97_LINE */ IDX_MIXER_LINEIN }, + { /* AC97_CD */ IDX_MIXER_CDAUDIO }, + { /* AC97_VIDEO */ IDX_MIXER_VIDEO }, + { /* AC97_AUX */ IDX_MIXER_AUX }, + { /* AC97_PCM */ IDX_MIXER_WAVEOUT }, + { /* AC97_REC_SEL */ IDX_MIXER_REC_SELECT }, + { /* AC97_REC_GAIN */ IDX_MIXER_REC_VOLUME }, + { /* AC97_REC_GAIN_MIC */ AZF_AC97_REG_EMU_IO_RW }, + { /* AC97_GENERAL_PURPOSE */ IDX_MIXER_ADVCTL2 }, + { /* AC97_3D_CONTROL */ IDX_MIXER_ADVCTL1 }, + }; + + unsigned short reg_azf = AZF_AC97_REG_UNSUPPORTED; + + /* azf3328 supports the low-numbered and low-spec:ed range + of AC97 regs only */ + if (reg <= AC97_3D_CONTROL) { + unsigned short reg_idx = reg / 2; + reg_azf = azf_reg_mapper[reg_idx].azf_reg; + /* a translation-only entry means it's real read/write: */ + if (!(reg_azf & ~AZF_REG_MASK)) + reg_azf |= AZF_AC97_REG_REAL_IO_RW; + } else { + switch (reg) { + case AC97_POWERDOWN: + reg_azf = AZF_AC97_REG_EMU_IO_RW; + break; + case AC97_EXTENDED_ID: + reg_azf = AZF_AC97_REG_EMU_IO_READ; + break; + case AC97_EXTENDED_STATUS: + /* I don't know what the h*ll AC97 layer + * would consult this _extended_ register for + * given a base-AC97-advertised card, + * but let's just emulate it anyway :-P + */ + reg_azf = AZF_AC97_REG_EMU_IO_RW; + break; + case AC97_VENDOR_ID1: + case AC97_VENDOR_ID2: + reg_azf = AZF_AC97_REG_EMU_IO_READ; + break; + } + } + return reg_azf; +} + +static const unsigned short +azf_emulated_ac97_caps = + AC97_BC_DEDICATED_MIC | + AC97_BC_BASS_TREBLE | + /* Headphone is an FM Synth control here */ + AC97_BC_HEADPHONE | + /* no AC97_BC_LOUDNESS! */ + /* mask 0x7c00 is + vendor-specific 3D enhancement + vendor indicator. + Since there actually _is_ an + entry for Aztech Labs + (13), make damn sure + to indicate it. */ + (13 << 10); + +static const unsigned short +azf_emulated_ac97_powerdown = + /* pretend everything to be active */ + AC97_PD_ADC_STATUS | + AC97_PD_DAC_STATUS | + AC97_PD_MIXER_STATUS | + AC97_PD_VREF_STATUS; + +/* + * Emulated, _inofficial_ vendor ID + * (there might be some devices such as the MR 2800-W + * which could reveal the real Aztech AC97 ID). + * We choose to use "AZT" prefix, and then use 1 to indicate PCI168 + * (better don't use 0x68 since there's a PCI368 as well). + */ +static const unsigned int +azf_emulated_ac97_vendor_id = 0x415a5401; + +static unsigned short +snd_azf3328_mixer_ac97_read(struct snd_ac97 *ac97, unsigned short reg_ac97) +{ + const struct snd_azf3328 *chip = ac97->private_data; + unsigned short reg_azf = snd_azf3328_mixer_ac97_map_reg_idx(reg_ac97); + unsigned short reg_val = 0; + bool unsupported = 0; + + snd_azf3328_dbgmixer( + "snd_azf3328_mixer_ac97_read reg_ac97 %u\n", + reg_ac97 + ); + if (reg_azf & AZF_AC97_REG_UNSUPPORTED) + unsupported = 1; + else { + if (reg_azf & AZF_AC97_REG_REAL_IO_READ) + reg_val = snd_azf3328_mixer_inw(chip, + reg_azf & AZF_REG_MASK); + else { + /* + * Proceed with dummy I/O read, + * to ensure compatible timing where this may matter. + * (ALSA AC97 layer usually doesn't call I/O functions + * due to intelligent I/O caching anyway) + * Choose a mixer register that's thoroughly unrelated + * to common audio (try to minimize distortion). + */ + snd_azf3328_mixer_inw(chip, IDX_MIXER_SOMETHING30H); + } + + if (reg_azf & AZF_AC97_REG_EMU_IO_READ) { + switch (reg_ac97) { + case AC97_RESET: + reg_val |= azf_emulated_ac97_caps; + break; + case AC97_POWERDOWN: + reg_val |= azf_emulated_ac97_powerdown; + break; + case AC97_EXTENDED_ID: + case AC97_EXTENDED_STATUS: + /* AFAICS we simply can't support anything: */ + reg_val |= 0; + break; + case AC97_VENDOR_ID1: + reg_val = azf_emulated_ac97_vendor_id >> 16; + break; + case AC97_VENDOR_ID2: + reg_val = azf_emulated_ac97_vendor_id & 0xffff; + break; + default: + unsupported = 1; + break; + } + } + } + if (unsupported) + snd_azf3328_mixer_ac97_map_unsupported(reg_ac97, "read"); + + return reg_val; +} + +static void +snd_azf3328_mixer_ac97_write(struct snd_ac97 *ac97, + unsigned short reg_ac97, unsigned short val) +{ + const struct snd_azf3328 *chip = ac97->private_data; + unsigned short reg_azf = snd_azf3328_mixer_ac97_map_reg_idx(reg_ac97); + bool unsupported = 0; + + snd_azf3328_dbgmixer( + "snd_azf3328_mixer_ac97_write reg_ac97 %u val %u\n", + reg_ac97, val + ); + if (reg_azf & AZF_AC97_REG_UNSUPPORTED) + unsupported = 1; + else { + if (reg_azf & AZF_AC97_REG_REAL_IO_WRITE) + snd_azf3328_mixer_outw( + chip, + reg_azf & AZF_REG_MASK, + val + ); + else + if (reg_azf & AZF_AC97_REG_EMU_IO_WRITE) { + switch (reg_ac97) { + case AC97_REC_GAIN_MIC: + case AC97_POWERDOWN: + case AC97_EXTENDED_STATUS: + /* + * Silently swallow these writes. + * Since for most registers our card doesn't + * actually support a comparable feature, + * this is exactly what we should do here. + * The AC97 layer's I/O caching probably + * automatically takes care of all the rest... + * (remembers written values etc.) + */ + break; + default: + unsupported = 1; + break; + } + } + } + if (unsupported) + snd_azf3328_mixer_ac97_map_unsupported(reg_ac97, "write"); +} + +static int __devinit +snd_azf3328_mixer_new(struct snd_azf3328 *chip) +{ + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + static struct snd_ac97_bus_ops ops = { + .write = snd_azf3328_mixer_ac97_write, + .read = snd_azf3328_mixer_ac97_read, + }; + int rc; + + memset(&ac97, 0, sizeof(ac97)); + ac97.scaps = AC97_SCAP_SKIP_MODEM + | AC97_SCAP_AUDIO /* we support audio! */ + | AC97_SCAP_NO_SPDIF; + ac97.private_data = chip; + ac97.pci = chip->pci; + + /* + * ALSA's AC97 layer has terrible init crackling issues, + * unfortunately, and since it makes use of AC97_RESET, + * there's no use trying to mute Master Playback proactively. + */ + + rc = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus); + if (!rc) + rc = snd_ac97_mixer(bus, &ac97, &chip->ac97); + /* + * Make sure to complain loudly in case of AC97 init failure, + * since failure may happen quite often, + * due to this card being a very quirky AC97 "lookalike". + */ + if (rc) + printk(KERN_ERR "azt3328: AC97 init failed, err %d!\n", rc); + + /* If we return an error here, then snd_card_free() should + * free up any ac97 codecs that got created, as well as the bus. + */ + return rc; +} +#else /* AZF_USE_AC97_LAYER */ static void snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, unsigned reg, @@ -945,6 +1285,7 @@ snd_azf3328_dbgcallleave(); return 0; } +#endif /* AZF_USE_AC97_LAYER */ static int snd_azf3328_hw_params(struct snd_pcm_substream *substream, @@ -1233,8 +1574,8 @@ if (is_main_mixer_playback_codec) { /* mute WaveOut (avoid clicking during setup) */ previously_muted = - snd_azf3328_mixer_set_mute( - chip, IDX_MIXER_WAVEOUT, 1 + snd_azf3328_mixer_mute_control_pcm( + chip, 1 ); } @@ -1290,8 +1631,8 @@ if (is_main_mixer_playback_codec) { /* now unmute WaveOut */ if (!previously_muted) - snd_azf3328_mixer_set_mute( - chip, IDX_MIXER_WAVEOUT, 0 + snd_azf3328_mixer_mute_control_pcm( + chip, 0 ); } @@ -1315,8 +1656,8 @@ if (is_main_mixer_playback_codec) { /* mute WaveOut (avoid clicking during setup) */ previously_muted = - snd_azf3328_mixer_set_mute( - chip, IDX_MIXER_WAVEOUT, 1 + snd_azf3328_mixer_mute_control_pcm( + chip, 1 ); } @@ -1341,8 +1682,8 @@ if (is_main_mixer_playback_codec) { /* now unmute WaveOut */ if (!previously_muted) - snd_azf3328_mixer_set_mute( - chip, IDX_MIXER_WAVEOUT, 0 + snd_azf3328_mixer_mute_control_pcm( + chip, 0 ); } @@ -2050,11 +2391,7 @@ if (chip->irq < 0) goto __end_hw; - /* reset (close) mixer: - * first mute master volume, then reset - */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); - snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000); + snd_azf3328_mixer_reset(chip); snd_azf3328_timer_stop(chip->timer); snd_azf3328_gameport_free(chip); @@ -2407,6 +2744,55 @@ } } +static inline void +snd_azf3328_resume_regs(const u32 *saved_regs, + unsigned long io_addr, + unsigned count +) +{ + unsigned reg; + + for (reg = 0; reg < count; ++reg) { + outl(*saved_regs, io_addr); + snd_azf3328_dbgpm("resume: io 0x%04lx: 0x%08x --> 0x%08x\n", + io_addr, *saved_regs, inl(io_addr)); + ++saved_regs; + io_addr += sizeof(*saved_regs); + } +} + +static inline void +snd_azf3328_suspend_ac97(struct snd_azf3328 *chip) +{ +#ifdef AZF_USE_AC97_LAYER + snd_ac97_suspend(chip->ac97); +#else + snd_azf3328_suspend_regs(chip->mixer_io, + ARRAY_SIZE(chip->saved_regs_mixer), chip->saved_regs_mixer); + + /* make sure to disable master volume etc. to prevent looping sound */ + snd_azf3328_mixer_mute_control_master(chip, 1); + snd_azf3328_mixer_mute_control_pcm(chip, 1); +#endif /* AZF_USE_AC97_LAYER */ +} + +static inline void +snd_azf3328_resume_ac97(const struct snd_azf3328 *chip) +{ +#ifdef AZF_USE_AC97_LAYER + snd_ac97_resume(chip->ac97); +#else + snd_azf3328_resume_regs(chip->saved_regs_mixer, chip->mixer_io, + ARRAY_SIZE(chip->saved_regs_mixer)); + + /* unfortunately with 32bit transfers, IDX_MIXER_PLAY_MASTER (0x02) + and IDX_MIXER_RESET (offset 0x00) get touched at the same time, + resulting in a mixer reset condition persisting until _after_ + master vol was restored. Thus master vol needs an extra restore. */ + outw(((u16 *)chip->saved_regs_mixer)[1], chip->mixer_io + 2); +#endif /* AZF_USE_AC97_LAYER */ +} + static int snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) { @@ -2420,12 +2806,7 @@ snd_pcm_suspend_all(chip->pcm[AZF_CODEC_PLAYBACK]); snd_pcm_suspend_all(chip->pcm[AZF_CODEC_I2S_OUT]); - snd_azf3328_suspend_regs(chip->mixer_io, - ARRAY_SIZE(chip->saved_regs_mixer), chip->saved_regs_mixer); - - /* make sure to disable master volume etc. to prevent looping sound */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); + snd_azf3328_suspend_ac97(chip); snd_azf3328_suspend_regs(chip->ctrl_io, ARRAY_SIZE(chip->saved_regs_ctrl), chip->saved_regs_ctrl); @@ -2447,23 +2828,6 @@ return 0; } -static inline void -snd_azf3328_resume_regs(const u32 *saved_regs, - unsigned long io_addr, - unsigned count -) -{ - unsigned reg; - - for (reg = 0; reg < count; ++reg) { - outl(*saved_regs, io_addr); - snd_azf3328_dbgpm("resume: io 0x%04lx: 0x%08x --> 0x%08x\n", - io_addr, *saved_regs, inl(io_addr)); - ++saved_regs; - io_addr += sizeof(*saved_regs); - } -} - static int snd_azf3328_resume(struct pci_dev *pci) { @@ -2487,14 +2851,7 @@ snd_azf3328_resume_regs(chip->saved_regs_opl3, chip->opl3_io, ARRAY_SIZE(chip->saved_regs_opl3)); - snd_azf3328_resume_regs(chip->saved_regs_mixer, chip->mixer_io, - ARRAY_SIZE(chip->saved_regs_mixer)); - - /* unfortunately with 32bit transfers, IDX_MIXER_PLAY_MASTER (0x02) - and IDX_MIXER_RESET (offset 0x00) get touched at the same time, - resulting in a mixer reset condition persisting until _after_ - master vol was restored. Thus master vol needs an extra restore. */ - outw(((u16 *)chip->saved_regs_mixer)[1], chip->mixer_io + 2); + snd_azf3328_resume_ac97(chip); snd_azf3328_resume_regs(chip->saved_regs_ctrl, chip->ctrl_io, ARRAY_SIZE(chip->saved_regs_ctrl));

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[alsa-devel] ASoC updates for 2.6.39
by Mark Brown 19 Feb '11

19 Feb '11

The following changes since commit 46fdaa3bec1bcaa36759b1f5e5e27245ae465b8f: ASoC: soc-cache: dereferencing before checking (2011-02-08 11:27:45 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Dimitris Papastamos (2): ASoC: WM8994: Improve robustness in some use cases ASoC: WM8994: Improve playback robustness Janusz Krzysztofik (1): ASoC: CX20442: fix wrong reg_cache_default content Jarkko Nikula (1): ASoC: Allow use sleeping gpio in soc-jack Mark Brown (21): ASoC: Remove redundant -codec from WM8903 driver name ASoC: Display WM8903 chip revision alphabetically ASoC: Add support for AIF channel muxing on WM8903 ASoC: Convert WM8903 to use PGA_S for output stage enables ASoC: Dynamically manage CLK_SYS in WM8903 ASoC: Fix WM8903 DAC mute default ASoC: Sync initial widget state with hardware Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Actively manage WM8903 DC servo configuration ASoC: Convert WM8903 bias management to use snd_soc_update_bits() ASoC: Don't use write sequencer to power up WM8903 ASoC: Use explicit sequence for WM8903 bias off Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Ensure supplies are maintained for force enabled widgets ASoC: Warn if WM8903 platform data is used to enable microphone IRQ Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Fix missing space in WM8994 ASoC: Remove export of snd_soc_dapm_stream_event() ASoC: Log wm_hubs DC servo operation code when reporting a timeout ASoC: Allow GPIO jack detection to be configured as a wake source ASoC: Add kerneldoc for jack_status_check callback Stephen Warren (8): ASoC: Tegra: Harmony: Add headphone jack detection ASoC: Tegra: Harmony: Add switch control for speaker ASoC: WM8903: Fix mic detection register definitions ASoC: WM8903: Fix mic detection enable logic ASoC: Tegra: Harmony: Implement mic detection ASoC: Harmony: Call snd_soc_dapm_nc_pin ASoC: Tegra: Harmony: Explicitly set mic enables ASoC: Tegra: Add MODULE_ALIAS Vaibhav Bedia (3): ASoC: Davinci: Use resource_size() helper function ASoC: Davinci: Call clk_disable() and clk_put() in case of error ASoC: Davinci: Replace usage of IO_ADDRESS with ioremap() Vinod Koul (10): ASoC: sn95031: Add jack support in the codec ASoC: mfld_machine: Add support for jack detection ASoC: sn95031: Add support for reading mic bias ASoC: add support for multiple jack types ASoC: mid-x86: Use the soc-jack apis for jack type detection ASoC: sn95031: make playback rails depend on actual pins they control ASoC: sn95031: fix the DMIC path routing ASoC: sn95031: fix the amic tlv scale ASoC: mfld_machine: make use of soc_register_card API ASoC: sst_platform: fix the pulseaudio error include/sound/soc.h | 29 ++ include/sound/wm8903.h | 10 +- sound/soc/codecs/cx20442.c | 2 +- sound/soc/codecs/sn95031.c | 239 +++++++++++++++--- sound/soc/codecs/sn95031.h | 33 +++ sound/soc/codecs/wm8903.c | 514 ++++++++++++++++++++++++------------- sound/soc/codecs/wm8903.h | 8 + sound/soc/codecs/wm8994.c | 199 +++++++++++++- sound/soc/codecs/wm_hubs.c | 3 +- sound/soc/davinci/davinci-i2s.c | 28 ++- sound/soc/davinci/davinci-mcasp.c | 29 ++- sound/soc/mid-x86/mfld_machine.c | 209 +++++++++++++-- sound/soc/mid-x86/sst_platform.c | 9 + sound/soc/soc-dapm.c | 24 ++- sound/soc/soc-jack.c | 56 ++++- sound/soc/tegra/harmony.c | 103 +++++++- sound/soc/tegra/tegra_das.c | 1 + sound/soc/tegra/tegra_i2s.c | 1 + sound/soc/tegra/tegra_pcm.c | 5 +- 19 files changed, 1207 insertions(+), 295 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: HDA: Do not announce false surround in Conexant auto
by David Henningsson 19 Feb '11

19 Feb '11

Without this patch, one line-out and one speaker and Conexant's auto parser would announce (non-working) surround capabilities. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] [PATCH] ALSA: HDA: Conexant auto: Handle multiple connections to ADC node
by David Henningsson 19 Feb '11

19 Feb '11

Conexant 20641 has several inputs to its ADC node, with one selector and individual amps for all inputs. This patch adds support in the Conexant auto parser to handle that case. It also means that the pin node's volume is being renamed to "Boost" to avoid name clash with the new volume controls on the ADC node. I was a little unsure whether to rename it to "Boost" or "Boost Capture" - 20641 does not have analog monitoring possibility, so that would be "Boost Capture", but OTOH, if the volume is on the pin, there is no possibility it cannot affect playback, if there is a link. So "Boost" feels more future-safe, whereas "Boost Capture" feels more correct given current situation. What do you think? I'm attaching a codec proc with the codec in question (before applied patch). I would want this in 2.6.38, but if you feel the change is too large too late, I'll merge it downstream (in Ubuntu) for the time being. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] [PATCH v9 00/12] Media controller (core and V4L2)
by Laurent Pinchart 19 Feb '11

19 Feb '11

Hi everybody, Here is the ninth version of the media controller core and V4L2 patches. Quick reminder for those who missed the previous version. let me quote the documentation (Documentation/DocBook/v4l/media-controller.xml). "Discovering a [media] device internal topology, and configuring it at runtime, is one of the goals of the media controller API. To achieve this, hardware devices are modelled as an oriented graph of building blocks called entities connected through pads." The code has been extensively reviewed by the V4L community, and this version is the first one to incorporate comments from the ALSA community (big thanks to Mark Brown and Clemens Ladisch). Two issues are not fully addressed yet, namely power management (I need to discuss this some more with the ALSA developers to really understand their requirements) and entities type names. I'm still posting this for review, as other developers have showed interest in commenting on the code. I want to emphasize once again that the media controller API does not replace the V4L, DVB or ALSA APIs. It complements them. The first user of the media controller API is the OMAP3 ISP driver. You can find it (as well as these patches and other V4L-specific patches) in a git tree at http://git.linuxtv.org/pinchartl/media.git (media-0005-omap3isp branch). The OMAP3 ISP driver patches are regularly posted for review on the linux-media list. Laurent Pinchart (11): media: Media device node support media: Media device media: Entities, pads and links media: Entity use count media: Media device information query media: Entities, pads and links enumeration media: Links setup media: Pipelines and media streams v4l: Add a media_device pointer to the v4l2_device structure v4l: Make video_device inherit from media_entity v4l: Make v4l2_subdev inherit from media_entity Sakari Ailus (1): media: Entity graph traversal Documentation/ABI/testing/sysfs-bus-media | 6 + Documentation/DocBook/media-entities.tmpl | 24 + Documentation/DocBook/media.tmpl | 3 + Documentation/DocBook/v4l/media-controller.xml | 89 ++++ Documentation/DocBook/v4l/media-func-close.xml | 59 +++ Documentation/DocBook/v4l/media-func-ioctl.xml | 116 +++++ Documentation/DocBook/v4l/media-func-open.xml | 94 ++++ .../DocBook/v4l/media-ioc-device-info.xml | 133 +++++ .../DocBook/v4l/media-ioc-enum-entities.xml | 308 +++++++++++ Documentation/DocBook/v4l/media-ioc-enum-links.xml | 207 ++++++++ Documentation/DocBook/v4l/media-ioc-setup-link.xml | 93 ++++ Documentation/media-framework.txt | 353 +++++++++++++ Documentation/video4linux/v4l2-framework.txt | 72 +++- drivers/media/Kconfig | 13 + drivers/media/Makefile | 6 + drivers/media/media-device.c | 382 ++++++++++++++ drivers/media/media-devnode.c | 321 ++++++++++++ drivers/media/media-entity.c | 536 ++++++++++++++++++++ drivers/media/video/v4l2-dev.c | 49 ++- drivers/media/video/v4l2-device.c | 49 ++- drivers/media/video/v4l2-subdev.c | 28 +- include/linux/Kbuild | 1 + include/linux/media.h | 132 +++++ include/media/media-device.h | 95 ++++ include/media/media-devnode.h | 97 ++++ include/media/media-entity.h | 151 ++++++ include/media/v4l2-dev.h | 7 + include/media/v4l2-device.h | 4 + include/media/v4l2-subdev.h | 6 + 29 files changed, 3413 insertions(+), 21 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-media create mode 100644 Documentation/DocBook/v4l/media-controller.xml create mode 100644 Documentation/DocBook/v4l/media-func-close.xml create mode 100644 Documentation/DocBook/v4l/media-func-ioctl.xml create mode 100644 Documentation/DocBook/v4l/media-func-open.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-device-info.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-enum-entities.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-enum-links.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-setup-link.xml create mode 100644 Documentation/media-framework.txt create mode 100644 drivers/media/media-device.c create mode 100644 drivers/media/media-devnode.c create mode 100644 drivers/media/media-entity.c create mode 100644 include/linux/media.h create mode 100644 include/media/media-device.h create mode 100644 include/media/media-devnode.h create mode 100644 include/media/media-entity.h -- Regards, Laurent Pinchart

3 15

[alsa-devel] [PATCH 1/2] ASoC: Allow GPIO jack detection to be configured as a wake source
by Mark Brown 18 Feb '11

18 Feb '11

Some systems wish to use jacks as wake sources. Provide a wake flag in the GPIO configuration which causes the driver to enable the IRQ as a wake source. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 3 +++ sound/soc/soc-jack.c | 8 ++++++++ 2 files changed, 11 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4ccf1e4..fb57c33 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -436,6 +436,7 @@ struct snd_soc_jack_zone { * @report: value to report when jack detected * @invert: report presence in low state * @debouce_time: debouce time in ms + * @wake: enable as wake source */ #ifdef CONFIG_GPIOLIB struct snd_soc_jack_gpio { @@ -444,6 +445,8 @@ struct snd_soc_jack_gpio { int report; int invert; int debounce_time; + bool wake; + struct snd_soc_jack *jack; struct delayed_work work; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 4579ee0..1382251 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -330,6 +330,14 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, if (ret) goto err; + if (gpios[i].wake) { + ret = set_irq_wake(gpio_to_irq(gpios[i].gpio), 1); + if (ret != 0) + printk(KERN_ERR + "Failed to mark GPIO %d as wake source: %d\n", + gpios[i].gpio, ret); + } + #ifdef CONFIG_GPIO_SYSFS /* Expose GPIO value over sysfs for diagnostic purposes */ gpio_export(gpios[i].gpio, false); -- 1.7.2.3

2 2

[alsa-devel] [PATCH] ASoC: Log wm_hubs DC servo operation code when reporting a timeout
by Mark Brown 18 Feb '11

18 Feb '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 613df5d..a5c1556 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -82,7 +82,8 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) } while (reg & op && count < 400); if (reg & op) - dev_err(codec->dev, "Timed out waiting for DC Servo\n"); + dev_err(codec->dev, "Timed out waiting for DC Servo %x\n", + op); } /* -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Remove export of snd_soc_dapm_stream_event()
by Mark Brown 18 Feb '11

18 Feb '11

The only thing that should ever be calling this is soc-core and that is built as part of the same module so doesn't need the export. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1bd24d5..3bee4b2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2263,7 +2263,6 @@ int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, mutex_unlock(&codec->mutex); return 0; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); /** * snd_soc_dapm_enable_pin - enable pin. -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Fix missing space in WM8994
by Mark Brown 18 Feb '11

18 Feb '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d78abec..c0adbf8 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -526,7 +526,7 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct wm8994_priv *wm8994 =snd_soc_codec_get_drvdata(codec); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int block = wm8994_get_retune_mobile_block(kcontrol->id.name); ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block]; -- 1.7.2.3

2 1

[alsa-devel] [PATCH v2 00/13] OMAP: McBSP: hwmod adaptation and runtime conversion
by Kishon Vijay Abraham I 18 Feb '11

18 Feb '11

Modify OMAP McBSP driver to use omap hwmod framework and pm runtime APIs. Created on top of linux OMAP master (linux-omap-2.6 :master) Did digital loopback testing on OMAP4430, OMAP3430 and OMAP2430 SDP boards. Verified that this patch series does not break the OMAP1 build. Patch series modifies audio layer and hence would appreciate the help of some audio guy to test this series. Patch series requires the following patch to be present http://permalink.gmane.org/gmane.linux.ports.arm.omap/51132 http://permalink.gmane.org/gmane.linux.ports.arm.omap/51133 http://permalink.gmane.org/gmane.linux.ports.arm.omap/51134 V2: * Added omap_hwmod_lookup() in the callback to omap_hwmod_for_each_by_class() to obtain hwmod data for sidetone. Previously this nesting of hwmod APIs was prevented by the use of mutex. * Added a revision member in hwmod database inorder to facilitate the driver to differentiate between different OMAP. * Created APIs to pass DMA params from McBSP driver to client drivers * Cleaned up sound soc by removing the use of macros to obtain base address and DMA channel number and instead use APIs exposed by the driver. * Removed macros defined in mcbsp driver for data that is obtained from hwmod database V1: * McBSP is designed to use multiple hwmods for a single device when the McBSP device has sidetone feature. * To avoid funcionality break of OMAP1 McBSP in between the series and to keep the patches readable, implementation was done in two steps: - First modify mcbsp driver to use platform_get* APIs - then convert it to use hwmod framework for OMAP2+. * API's like omap_device_noidle() and omap_device_default_idle() is used to change the SYCONFIG register bits. This change is done to align with the discussion on [2] * Use '.rev' of omap_hwmod class to identify OMAP3 specific settings * Use *ST_* macros for idlest_idle bit * Incorporate other general review comments provided for hwmod adpatation of other OMAP driver's (eg., do pdata free after a omap_device_build()) * Retain fclk even after pm_runtime adaptation to facilitate switching of functional clock from one source to another * Add member 'name' to omap_hwmod_addr_space struct so that the driver need not rely on the order to get the proper resource [3]. Discussions related to the first RFC patch can be found at [1] [1]: http://www.mail-archive.com/linux-omap@vger.kernel.org/msg36743.html [2]: http://www.mail-archive.com/linux-omap@vger.kernel.org/msg39615.html [3]: https://patchwork.kernel.org/patch/233211/ Benoit Cousson (1): OMAP4: hwmod data: Add McBSP Charulatha V (3): OMAP2420: hwmod data: Add McBSP OMAP2430: hwmod data: Add McBSP OMAP3: hwmod data: Add McBSP Kishon Vijay Abraham I (9): OMAP: hwmod: Add member 'name' to omap_hwmod_addr_space struct OMAP: McBSP: Convert McBSP to platform device model OMAP3: hwmod: add dev_attr for McBSP sidetone OMAP2+: McBSP: hwmod adaptation for McBSP OMAP: McBSP: use omap_device APIs to modify SYSCONFIG OMAP: McBSP: Add pm runtime support OMAP: McBSP: APIs to pass DMA params from McBSP driver to client drivers ASoC: McBSP: get hw params from McBSP driver OMAP: hwmod: Removal of macros for data that is obtained from hwmod database arch/arm/mach-omap1/mcbsp.c | 383 +++++++++++++++---- arch/arm/mach-omap2/mcbsp.c | 228 +++--------- arch/arm/mach-omap2/omap_hwmod.c | 1 + arch/arm/mach-omap2/omap_hwmod_2420_data.c | 167 ++++++++ arch/arm/mach-omap2/omap_hwmod_2430_data.c | 417 ++++++++++++++++++++ arch/arm/mach-omap2/omap_hwmod_3xxx_data.c | 544 ++++++++++++++++++++++++++ arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 321 +++++++++++++++ arch/arm/mach-omap2/prcm-common.h | 4 + arch/arm/plat-omap/devices.c | 10 +- arch/arm/plat-omap/include/plat/mcbsp.h | 69 +--- arch/arm/plat-omap/include/plat/omap_hwmod.h | 4 +- arch/arm/plat-omap/mcbsp.c | 207 +++++++--- sound/soc/omap/omap-mcbsp.c | 126 +------ 13 files changed, 2006 insertions(+), 475 deletions(-)

9 38

[alsa-devel] Channel swapping problem on Atmel SSC audio
by Ryan Mallon 18 Feb '11

18 Feb '11

Hi all, I'm working on a custom board using the AT91SAM9G45 processor and a TLV320AIC26 SPI attached audio codec using I2S format audio. When doing audio playback on the board the left and right channels are occasionally flipped. I have verified this by generating a custom wav file which has constant data on one channel and an incrementing counter on the other channel. The digitial out from the CPU shows that no data is being lost, but around 1/5 times the audio channels will be flipped. The problem appears to be that the DMA is starting on the wrong edge of the LRCLK. I have using both SND_SOC_DAIFMT_CBM_CFM (Codec provides clocks) and SND_SOC_DAIFMT_CBS_CFS (CPU provides clocks) but the problem occurs in both situations. I have also tried changing TCMR:CKO to SSC_CKO_TRANSFER and TCMR:START to SSC_START_LOW_RF, but the problem still occurs. I have looked through the PCM/SSC DMA code and cannot find any obvious bugs. I have checked the errata for the At91SAM9G45 processor (our chip marking is CU-1004-A), but can't find anything which relates directly to the problem we are seeing. Has anybody else had this problem, or does anybody have any AT91/Atmel hardware that they could test to see if it occurs on other hardware also? Thanks, ~Ryan -- Bluewater Systems Ltd - ARM Technology Solution Centre Ryan Mallon 5 Amuri Park, 404 Barbadoes St ryan(a)bluewatersys.com PO Box 13 889, Christchurch 8013 http://www.bluewatersys.com New Zealand Phone: +64 3 3779127 Freecall: Australia 1800 148 751 Fax: +64 3 3779135 USA 1800 261 2934

6 6

[alsa-devel] [PATCH] ASoC: Support direct microphone detection IRQ
by Mark Brown 18 Feb '11

18 Feb '11

Allow direct routing of the WM8958 microphone detection signal to a GPIO to be used, saving the need to demux the interrupt. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/mfd/wm8994/pdata.h | 5 +++ sound/soc/codecs/wm8994.c | 57 +++++++++++++++++++++++-------------- 2 files changed, 40 insertions(+), 22 deletions(-) diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index 9eab263..0686946 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -103,6 +103,11 @@ struct wm8994_pdata { unsigned int lineout1fb:1; unsigned int lineout2fb:1; + /* IRQ for microphone detection if brought out directly as a + * signal. + */ + int micdet_irq; + /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ unsigned int micbias1_lvl:1; unsigned int micbias2_lvl:1; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c0adbf8..dce2aa7 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -104,6 +104,7 @@ struct wm8994_priv { void *jack_cb_data; bool jack_is_mic; bool jack_is_video; + int micdet_irq; int revision; struct wm8994_pdata *pdata; @@ -3103,6 +3104,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->pdata = dev_get_platdata(codec->dev->parent); wm8994->codec = codec; + if (wm8994->pdata && wm8994->pdata->micdet_irq) + wm8994->micdet_irq = wm8994->pdata->micdet_irq; + else if (wm8994->pdata && wm8994->pdata->irq_base) + wm8994->micdet_irq = wm8994->pdata->irq_base + + WM8994_IRQ_MIC1_DET; + pm_runtime_enable(codec->dev); pm_runtime_resume(codec->dev); @@ -3151,14 +3158,17 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (control->type) { case WM8994: - ret = wm8994_request_irq(codec->control_data, - WM8994_IRQ_MIC1_DET, - wm8994_mic_irq, "Mic 1 detect", - wm8994); - if (ret != 0) - dev_warn(codec->dev, - "Failed to request Mic1 detect IRQ: %d\n", - ret); + if (wm8994->micdet_irq) { + ret = request_threaded_irq(wm8994->micdet_irq, NULL, + wm8994_mic_irq, + IRQF_TRIGGER_RISING, + "Mic1 detect", + wm8994); + if (ret != 0) + dev_warn(codec->dev, + "Failed to request Mic1 detect IRQ: %d\n", + ret); + } ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, @@ -3189,15 +3199,17 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM8958: - ret = wm8994_request_irq(codec->control_data, - WM8994_IRQ_MIC1_DET, - wm8958_mic_irq, "Mic detect", - wm8994); - if (ret != 0) - dev_warn(codec->dev, - "Failed to request Mic detect IRQ: %d\n", - ret); - break; + if (wm8994->micdet_irq) { + ret = request_threaded_irq(wm8994->micdet_irq, NULL + wm8958_mic_irq, + IRQF_TRIGGER_RISING, + "Mic detect", + wm8994); + if (ret != 0) + dev_warn(codec->dev, + "Failed to request Mic detect IRQ: %d\n", + ret); + } } /* Remember if AIFnLRCLK is configured as a GPIO. This should be @@ -3329,7 +3341,8 @@ err_irq: wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994); - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, wm8994); + if (wm8994->micdet_irq) + free_irq(wm8994->micdet_irq, wm8994); err: kfree(wm8994); return ret; @@ -3346,8 +3359,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) switch (control->type) { case WM8994: - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT, - wm8994); + if (wm8994->micdet_irq) + free_irq(wm8994->micdet_irq); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, @@ -3357,8 +3370,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) break; case WM8958: - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, - wm8994); + if (wm8994->micdet_irq) + free_irq(wm8994->micdet_irq); break; } kfree(wm8994->retune_mobile_texts); -- 1.7.2.3

2 1

[alsa-devel] [PATCH 13/29] twl4030: mfd_cell is now implicitly available to drivers (v2)
by Andres Salomon 18 Feb '11

18 Feb '11

The cell's platform_data is now accessed with a helper function; change clients to use that, and remove the now-unused data_size. v2: add some missing mfd/core.h includes. Signed-off-by: Andres Salomon <dilinger(a)queued.net> --- drivers/input/misc/twl4030-vibra.c | 3 ++- drivers/mfd/twl4030-codec.c | 2 -- sound/soc/codecs/twl4030.c | 6 ++++-- 3 files changed, 6 insertions(+), 5 deletions(-) diff --git a/drivers/input/misc/twl4030-vibra.c b/drivers/input/misc/twl4030-vibra.c index 014dd4a..6a11694 100644 --- a/drivers/input/misc/twl4030-vibra.c +++ b/drivers/input/misc/twl4030-vibra.c @@ -29,6 +29,7 @@ #include <linux/workqueue.h> #include <linux/i2c/twl.h> #include <linux/mfd/twl4030-codec.h> +#include <linux/mfd/core.h> #include <linux/input.h> #include <linux/slab.h> @@ -196,7 +197,7 @@ static SIMPLE_DEV_PM_OPS(twl4030_vibra_pm_ops, static int __devinit twl4030_vibra_probe(struct platform_device *pdev) { - struct twl4030_codec_vibra_data *pdata = pdev->dev.platform_data; + struct twl4030_codec_vibra_data *pdata = mfd_get_data(pdev); struct vibra_info *info; int ret; diff --git a/drivers/mfd/twl4030-codec.c b/drivers/mfd/twl4030-codec.c index 9a4b196..0f57426 100644 --- a/drivers/mfd/twl4030-codec.c +++ b/drivers/mfd/twl4030-codec.c @@ -209,14 +209,12 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) cell = &codec->cells[childs]; cell->name = "twl4030-codec"; cell->platform_data = pdata->audio; - cell->data_size = sizeof(*pdata->audio); childs++; } if (pdata->vibra) { cell = &codec->cells[childs]; cell->name = "twl4030-vibra"; cell->platform_data = pdata->vibra; - cell->data_size = sizeof(*pdata->vibra); childs++; } diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e4d464b..8512800 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -26,6 +26,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/mfd/core.h> #include <linux/i2c/twl.h> #include <linux/slab.h> #include <sound/core.h> @@ -732,7 +733,8 @@ static int aif_event(struct snd_soc_dapm_widget *w, static void headset_ramp(struct snd_soc_codec *codec, int ramp) { - struct twl4030_codec_audio_data *pdata = codec->dev->platform_data; + struct twl4030_codec_audio_data *pdata = + mfd_get_data(to_platform_device(codec->dev)); unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); /* Base values for ramp delay calculation: 2^19 - 2^26 */ @@ -2297,7 +2299,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { static int __devinit twl4030_codec_probe(struct platform_device *pdev) { - struct twl4030_codec_audio_data *pdata = pdev->dev.platform_data; + struct twl4030_codec_audio_data *pdata = mfd_get_data(pdev); if (!pdata) { dev_err(&pdev->dev, "platform_data is missing\n"); -- 1.7.2.3

2 1

[alsa-devel] [PATCH 20/29] wm8400-codec: use mfd_data instead of driver_data (v2)
by Andres Salomon 18 Feb '11

18 Feb '11

Use mfd_data for passing information from mfd drivers to soc clients. The mfd_cell's driver_data field is being phased out. Clients that were using driver_data now access .mfd_data via mfd_get_data(). v2: add some missing mfd/core.h includes. Signed-off-by: Andres Salomon <dilinger(a)queued.net> --- drivers/mfd/wm8400-core.c | 2 +- sound/soc/codecs/wm8400.c | 3 ++- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/drivers/mfd/wm8400-core.c b/drivers/mfd/wm8400-core.c index 1bfef48..3a6e78c 100644 --- a/drivers/mfd/wm8400-core.c +++ b/drivers/mfd/wm8400-core.c @@ -245,7 +245,7 @@ static int wm8400_register_codec(struct wm8400 *wm8400) { struct mfd_cell cell = { .name = "wm8400-codec", - .driver_data = wm8400, + .mfd_data = wm8400, }; return mfd_add_devices(wm8400->dev, -1, &cell, 1, NULL, 0); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 3c3bc07..736b785 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -22,6 +22,7 @@ #include <linux/regulator/consumer.h> #include <linux/mfd/wm8400-audio.h> #include <linux/mfd/wm8400-private.h> +#include <linux/mfd/core.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -1377,7 +1378,7 @@ static void wm8400_probe_deferred(struct work_struct *work) static int wm8400_codec_probe(struct snd_soc_codec *codec) { - struct wm8400 *wm8400 = dev_get_platdata(codec->dev); + struct wm8400 *wm8400 = mfd_get_data(to_platform_device(codec->dev)); struct wm8400_priv *priv; int ret; u16 reg; -- 1.7.2.3

2 1

[alsa-devel] [PATCH 19/29] davinci_voicecodec: use mfd_data instead of driver_data
by Andres Salomon 18 Feb '11

18 Feb '11

Use mfd_data for passing information from mfd drivers to soc clients. The mfd_cell's driver_data field is being phased out. Clients that were using driver_data now access .mfd_data via mfd_get_data(). Signed-off-by: Andres Salomon <dilinger(a)queued.net> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- drivers/mfd/davinci_voicecodec.c | 4 ++-- sound/soc/codecs/cq93vc.c | 3 ++- sound/soc/davinci/davinci-vcif.c | 2 +- 3 files changed, 5 insertions(+), 4 deletions(-) diff --git a/drivers/mfd/davinci_voicecodec.c b/drivers/mfd/davinci_voicecodec.c index 33c923d..f946c0b 100644 --- a/drivers/mfd/davinci_voicecodec.c +++ b/drivers/mfd/davinci_voicecodec.c @@ -119,12 +119,12 @@ static int __init davinci_vc_probe(struct platform_device *pdev) /* Voice codec interface client */ cell = &davinci_vc->cells[DAVINCI_VC_VCIF_CELL]; cell->name = "davinci_vcif"; - cell->driver_data = davinci_vc; + cell->mfd_data = davinci_vc; /* Voice codec CQ93VC client */ cell = &davinci_vc->cells[DAVINCI_VC_CQ93VC_CELL]; cell->name = "cq93vc"; - cell->driver_data = davinci_vc; + cell->mfd_data = davinci_vc; ret = mfd_add_devices(&pdev->dev, pdev->id, davinci_vc->cells, DAVINCI_VC_CELLS, NULL, 0); diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 46dbfd0..b8066ef 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -153,7 +153,8 @@ static int cq93vc_resume(struct snd_soc_codec *codec) static int cq93vc_probe(struct snd_soc_codec *codec) { - struct davinci_vc *davinci_vc = codec->dev->platform_data; + struct davinci_vc *davinci_vc = + mfd_get_data(to_platform_device(codec->dev)); davinci_vc->cq93vc.codec = codec; codec->control_data = davinci_vc; diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9d2afcc..13e05a3 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -205,7 +205,7 @@ static struct snd_soc_dai_driver davinci_vcif_dai = { static int davinci_vcif_probe(struct platform_device *pdev) { - struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct davinci_vc *davinci_vc = mfd_get_data(pdev); struct davinci_vcif_dev *davinci_vcif_dev; int ret; -- 1.7.2.3

1 0

[alsa-devel] [PATCH 10/29] wl1273: mfd_cell is now implicitly available to drivers
by Andres Salomon 18 Feb '11

18 Feb '11

The cell's platform_data is now accessed with a helper function; change clients to use that, and remove the now-unused data_size. Signed-off-by: Andres Salomon <dilinger(a)queued.net> --- drivers/media/radio/radio-wl1273.c | 2 +- drivers/mfd/wl1273-core.c | 2 -- sound/soc/codecs/wl1273.c | 3 ++- 3 files changed, 3 insertions(+), 4 deletions(-) diff --git a/drivers/media/radio/radio-wl1273.c b/drivers/media/radio/radio-wl1273.c index 7ecc8e6..4698eb0 100644 --- a/drivers/media/radio/radio-wl1273.c +++ b/drivers/media/radio/radio-wl1273.c @@ -2138,7 +2138,7 @@ static int wl1273_fm_radio_remove(struct platform_device *pdev) static int __devinit wl1273_fm_radio_probe(struct platform_device *pdev) { - struct wl1273_core **core = pdev->dev.platform_data; + struct wl1273_core **core = mfd_get_data(pdev); struct wl1273_device *radio; struct v4l2_ctrl *ctrl; int r = 0; diff --git a/drivers/mfd/wl1273-core.c b/drivers/mfd/wl1273-core.c index d2ecc24..703085e 100644 --- a/drivers/mfd/wl1273-core.c +++ b/drivers/mfd/wl1273-core.c @@ -80,7 +80,6 @@ static int __devinit wl1273_core_probe(struct i2c_client *client, cell = &core->cells[children]; cell->name = "wl1273_fm_radio"; cell->platform_data = &core; - cell->data_size = sizeof(core); children++; if (pdata->children & WL1273_CODEC_CHILD) { @@ -89,7 +88,6 @@ static int __devinit wl1273_core_probe(struct i2c_client *client, dev_dbg(&client->dev, "%s: Have codec.\n", __func__); cell->name = "wl1273-codec"; cell->platform_data = &core; - cell->data_size = sizeof(core); children++; } diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 861b28f..1ad0d5a 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -436,7 +436,8 @@ EXPORT_SYMBOL_GPL(wl1273_get_format); static int wl1273_probe(struct snd_soc_codec *codec) { - struct wl1273_core **core = codec->dev->platform_data; + struct wl1273_core **core = + mfd_get_data(to_platform_device(codec->dev)); struct wl1273_priv *wl1273; int r; -- 1.7.2.3

1 0

[alsa-devel] [PATCH] sound ac97: replace open-coded, error-prone stuff with AC97 bit defines
by Andreas Mohr 17 Feb '11

17 Feb '11

ChangeLog: Use AC97 macros (sometimes already existing, or newly added) instead of error-prone repetition of open-coded values. Signed-off-by: Andreas Mohr <andi(a)lisas.de> --- Patch created on -rc3, tested and checkpatch.pl'd. GIT history seems clean, should thus apply easily. Optional, suggested patch. Thanks! Index: linux-2.6/include/sound/ac97_codec.h =================================================================== --- linux-2.6.orig/include/sound/ac97_codec.h 2011-02-16 23:18:39.000000000 +0100 +++ linux-2.6/include/sound/ac97_codec.h 2011-02-16 23:19:04.000000000 +0100 @@ -96,6 +96,10 @@ #define AC97_FUNC_INFO 0x68 /* Function Information */ #define AC97_SENSE_INFO 0x6a /* Sense Details */ +/* volume controls */ +#define AC97_MUTE_MASK_MONO 0x8000 +#define AC97_MUTE_MASK_STEREO 0x8080 + /* slot allocation */ #define AC97_SLOT_TAG 0 #define AC97_SLOT_CMD_ADDR 1 @@ -138,6 +142,7 @@ #define AC97_BC_18BIT_ADC 0x0100 /* 18-bit ADC resolution */ #define AC97_BC_20BIT_ADC 0x0200 /* 20-bit ADC resolution */ #define AC97_BC_ADC_MASK 0x0300 +#define AC97_BC_3D_TECH_ID_MASK 0x7c00 /* Per-vendor ID of 3D enhancement */ /* general purpose */ #define AC97_GP_DRSS_MASK 0x0c00 /* double rate slot select */ Index: linux-2.6/sound/pci/ac97/ac97_codec.c =================================================================== --- linux-2.6.orig/sound/pci/ac97/ac97_codec.c 2011-02-16 23:19:03.000000000 +0100 +++ linux-2.6/sound/pci/ac97/ac97_codec.c 2011-02-16 23:19:04.000000000 +0100 @@ -597,9 +597,9 @@ snd_ac97_page_restore(ac97, page_save); #ifdef CONFIG_SND_AC97_POWER_SAVE /* check analog mixer power-down */ - if ((val_mask & 0x8000) && + if ((val_mask & AC97_PD_EAPD) && (kcontrol->private_value & (1<<30))) { - if (val & 0x8000) + if (val & AC97_PD_EAPD) ac97->power_up &= ~(1 << (reg>>1)); else ac97->power_up |= 1 << (reg>>1); @@ -1042,20 +1042,20 @@ static int snd_ac97_try_volume_mix(struct snd_ac97 * ac97, int reg) { - unsigned short val, mask = 0x8000; + unsigned short val, mask = AC97_MUTE_MASK_MONO; if (! snd_ac97_valid_reg(ac97, reg)) return 0; switch (reg) { case AC97_MASTER_TONE: - return ac97->caps & 0x04 ? 1 : 0; + return ac97->caps & AC97_BC_BASS_TREBLE ? 1 : 0; case AC97_HEADPHONE: - return ac97->caps & 0x10 ? 1 : 0; + return ac97->caps & AC97_BC_HEADPHONE ? 1 : 0; case AC97_REC_GAIN_MIC: - return ac97->caps & 0x01 ? 1 : 0; + return ac97->caps & AC97_BC_DEDICATED_MIC ? 1 : 0; case AC97_3D_CONTROL: - if (ac97->caps & 0x7c00) { + if (ac97->caps & AC97_BC_3D_TECH_ID_MASK) { val = snd_ac97_read(ac97, reg); /* if nonzero - fixed and we can't set it */ return val == 0; @@ -1111,7 +1111,10 @@ *lo_max = *hi_max = 0; for (i = 0 ; i < ARRAY_SIZE(cbit); i++) { unsigned short val; - snd_ac97_write(ac97, reg, 0x8080 | cbit[i] | (cbit[i] << 8)); + snd_ac97_write( + ac97, reg, + AC97_MUTE_MASK_STEREO | cbit[i] | (cbit[i] << 8) + ); /* Do the read twice due to buffers on some ac97 codecs. * e.g. The STAC9704 returns exactly what you wrote to the register * if you read it immediately. This causes the detect routine to fail. @@ -1146,14 +1149,14 @@ unsigned short val, val1; *max = 63; - val = 0x8080 | (0x20 << shift); + val = AC97_MUTE_MASK_STEREO | (0x20 << shift); snd_ac97_write(ac97, reg, val); val1 = snd_ac97_read(ac97, reg); if (val != val1) { *max = 31; } /* reset volume to zero */ - snd_ac97_write_cache(ac97, reg, 0x8080); + snd_ac97_write_cache(ac97, reg, AC97_MUTE_MASK_STEREO); } static inline int printable(unsigned int x) @@ -1190,16 +1193,16 @@ if (! snd_ac97_valid_reg(ac97, reg)) return 0; - mute_mask = 0x8000; + mute_mask = AC97_MUTE_MASK_MONO; val = snd_ac97_read(ac97, reg); if (check_stereo || (ac97->flags & AC97_STEREO_MUTES)) { /* check whether both mute bits work */ - val1 = val | 0x8080; + val1 = val | AC97_MUTE_MASK_STEREO; snd_ac97_write(ac97, reg, val1); if (val1 == snd_ac97_read(ac97, reg)) - mute_mask = 0x8080; + mute_mask = AC97_MUTE_MASK_STEREO; } - if (mute_mask == 0x8080) { + if (mute_mask == AC97_MUTE_MASK_STEREO) { struct snd_kcontrol_new tmp = AC97_DOUBLE(name, reg, 15, 7, 1, 1); if (check_amix) tmp.private_value |= (1 << 30); @@ -1275,9 +1278,11 @@ err = snd_ctl_add(card, kctl); if (err < 0) return err; - snd_ac97_write_cache(ac97, reg, - (snd_ac97_read(ac97, reg) & 0x8080) | - lo_max | (hi_max << 8)); + snd_ac97_write_cache( + ac97, reg, + (snd_ac97_read(ac97, reg) & AC97_MUTE_MASK_STEREO) + | lo_max | (hi_max << 8) + ); return 0; } @@ -1339,7 +1344,7 @@ return err; } - ac97->regs[AC97_CENTER_LFE_MASTER] = 0x8080; + ac97->regs[AC97_CENTER_LFE_MASTER] = AC97_MUTE_MASK_STEREO; /* build center controls */ if ((snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER)) @@ -1417,8 +1422,12 @@ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_pc_beep[idx], ac97))) < 0) return err; set_tlv_db_scale(kctl, db_scale_4bit); - snd_ac97_write_cache(ac97, AC97_PC_BEEP, - snd_ac97_read(ac97, AC97_PC_BEEP) | 0x801e); + snd_ac97_write_cache( + ac97, + AC97_PC_BEEP, + (snd_ac97_read(ac97, AC97_PC_BEEP) + | AC97_MUTE_MASK_MONO | 0x001e) + ); } /* build Phone controls */ @@ -1552,7 +1561,7 @@ } /* build Simulated Stereo Enhancement control */ - if (ac97->caps & 0x0008) { + if (ac97->caps & AC97_BC_SIM_STEREO) { if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_general[AC97_GENERAL_STEREO_ENHANCEMENT], ac97))) < 0) return err; } @@ -1564,7 +1573,7 @@ } /* build Loudness control */ - if (ac97->caps & 0x0020) { + if (ac97->caps & AC97_BC_LOUDNESS) { if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_general[AC97_GENERAL_LOUDNESS], ac97))) < 0) return err; } @@ -2549,8 +2558,8 @@ schedule_timeout_uninterruptible(1); } while (time_after_eq(end_time, jiffies)); /* FIXME: extra delay */ - ac97->bus->ops->write(ac97, AC97_MASTER, 0x8000); - if (snd_ac97_read(ac97, AC97_MASTER) != 0x8000) + ac97->bus->ops->write(ac97, AC97_MASTER, AC97_MUTE_MASK_MONO); + if (snd_ac97_read(ac97, AC97_MASTER) != AC97_MUTE_MASK_MONO) msleep(250); } else { end_time = jiffies + msecs_to_jiffies(100); @@ -2754,12 +2763,12 @@ int rshift = (kcontrol->private_value >> 12) & 0x0f; unsigned short mask; if (shift != rshift) - mask = 0x8080; + mask = AC97_MUTE_MASK_STEREO; else - mask = 0x8000; - snd_ac97_update_bits(ac97, AC97_POWERDOWN, 0x8000, + mask = AC97_MUTE_MASK_MONO; + snd_ac97_update_bits(ac97, AC97_POWERDOWN, AC97_PD_EAPD, (ac97->regs[AC97_MASTER] & mask) == mask ? - 0x8000 : 0); + AC97_PD_EAPD : 0); } return err; } @@ -2772,7 +2781,10 @@ return -ENOENT; msw->put = master_mute_sw_put; snd_ac97_remove_ctl(ac97, "External Amplifier", NULL); - snd_ac97_update_bits(ac97, AC97_POWERDOWN, 0x8000, 0x8000); /* mute LED on */ + snd_ac97_update_bits( + ac97, AC97_POWERDOWN, + AC97_PD_EAPD, AC97_PD_EAPD /* mute LED on */ + ); ac97->scaps |= AC97_SCAP_EAPD_LED; return 0; } @@ -2787,12 +2799,12 @@ int rshift = (kcontrol->private_value >> 12) & 0x0f; unsigned short mask; if (shift != rshift) - mask = 0x8080; + mask = AC97_MUTE_MASK_STEREO; else - mask = 0x8000; - snd_ac97_update_bits(ac97, AC97_POWERDOWN, 0x8000, + mask = AC97_MUTE_MASK_MONO; + snd_ac97_update_bits(ac97, AC97_POWERDOWN, AC97_PD_EAPD, (ac97->regs[AC97_MASTER] & mask) == mask ? - 0x8000 : 0); + AC97_PD_EAPD : 0); } return err; } @@ -2808,7 +2820,10 @@ snd_ac97_remove_ctl(ac97, "External Amplifier", NULL); snd_ac97_remove_ctl(ac97, "Headphone Playback", "Switch"); snd_ac97_remove_ctl(ac97, "Headphone Playback", "Volume"); - snd_ac97_update_bits(ac97, AC97_POWERDOWN, 0x8000, 0x8000); /* mute LED on */ + snd_ac97_update_bits( + ac97, AC97_POWERDOWN, + AC97_PD_EAPD, AC97_PD_EAPD /* mute LED on */ + ); return 0; }

2 1

[alsa-devel] ASOC: AD193X, AT91SAM9263 and Channel Slot Shifting
by Jason Fox 17 Feb '11

17 Feb '11

I am currently working with an AD1939 being driven by an AT91SAM9263 and currently configured for 4 channels. The AT91 is successfully sending data to to the CODEC but the DMA transfer is not correctly synchronized to the LRCLK frame sync. So, the data for channel 0 might show up in slot 1, 2, 3 - or sometimes slot 0 where it belongs! Can anyone give me a suggestion as to where I might look in the drivers to debug this frame sync problem? My hardware engineer assures me that the AT91 SSC and AD1939 are configured correctly which regard to LRCLK and BCLK. Thanks! Jason Fox ******************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This footnote also confirms that this email message has been swept for the presence of computer viruses. www.Hubbell.com - Hubbell Incorporated**

1 0

[alsa-devel] [PATCH] ASoC: Remove export of snd_soc_dapm_stream_event()
by Mark Brown 17 Feb '11

17 Feb '11

The only thing that should ever be calling this is soc-core and that is built as part of the same module so doesn't need the export. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1bd24d5..3bee4b2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2263,7 +2263,6 @@ int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, mutex_unlock(&codec->mutex); return 0; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); /** * snd_soc_dapm_enable_pin - enable pin. -- 1.7.2.3

1 0

[alsa-devel] [PATCH v3] ASoC: Add Freescale SGTL5000 codec support
by zhaoming.zeng＠freescale.com 17 Feb '11

17 Feb '11

From: Zeng Zhaoming <zhaoming.zeng(a)freescale.com> Add Freescale SGTL5000 codec support Signed-off-by: Zeng Zhaoming <zhaoming.zeng(a)freescale.com> --- changes since v2: 1. clean up register default values 2. rewrite codec power up code, add sgtl5000_set_power_regs() 3. rewrite codec clock configure code sgtl5000_set_clock() 4. reimplement PM hooks, restore register by particular order. 5. clean up dapm code, remove dac and adc event hooks. 6. clean up codec private structure, remove unnecessary fields. 7. add comments for uncommon code. Thanks for Mark's review. --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/sgtl5000.c | 1229 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sgtl5000.h | 403 ++++++++++++++ 4 files changed, 1637 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c48b23c..57b909a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 + select SND_SOC_SGTL5000 if I2C select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS @@ -176,6 +177,9 @@ config SND_SOC_MAX98088 config SND_SOC_PCM3008 tristate +config SND_SOC_SGTL5000 + tristate + config SND_SOC_SPDIF tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 579af9c..78e36cc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -18,6 +18,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c new file mode 100644 index 0000000..2646ecb --- /dev/null +++ b/sound/soc/codecs/sgtl5000.c @@ -0,0 +1,1229 @@ +/* + * sgtl5000.c -- SGTL5000 ALSA SoC Audio driver + * + * Copyright 2010-2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <sound/core.h> +#include <sound/tlv.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <mach/hardware.h> + +#include "sgtl5000.h" + +#define SGTL5000_DAP_REG_OFFSET 0x0100 +#define SGTL5000_MAX_REG_OFFSET 0x013A + +/* default value of sgtl5000 registers except DAP */ +static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] = { + 0xa011, /* 0x0000, CHIP_ID. 11 stand for revison 17 */ + 0x0000, /* 0x0002, CHIP_DIG_POWER. */ + 0x0008, /* 0x0004, CHIP_CKL_CTRL */ + 0x0010, /* 0x0006, CHIP_I2S_CTRL */ + 0x0000, /* 0x0008, reserved */ + 0x0008, /* 0x000A, CHIP_SSS_CTRL */ + 0x0000, /* 0x000C, reserved */ + 0x020c, /* 0x000E, CHIP_ADCDAC_CTRL */ + 0x3c3c, /* 0x0010, CHIP_DAC_VOL */ + 0x0000, /* 0x0012, reserved */ + 0x015f, /* 0x0014, CHIP_PAD_STRENGTH */ + 0x0000, /* 0x0016, reserved */ + 0x0000, /* 0x0018, reserved */ + 0x0000, /* 0x001A, reserved */ + 0x0000, /* 0x001E, reserved */ + 0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */ + 0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */ + 0x0111, /* 0x0024, CHIP_ANN_CTRL */ + 0x0000, /* 0x0026, CHIP_LINREG_CTRL */ + 0x0000, /* 0x0028, CHIP_REF_CTRL */ + 0x0000, /* 0x002A, CHIP_MIC_CTRL */ + 0x0000, /* 0x002C, CHIP_LINE_OUT_CTRL */ + 0x0404, /* 0x002E, CHIP_LINE_OUT_VOL */ + 0x7060, /* 0x0030, CHIP_ANA_POWER */ + 0x5000, /* 0x0032, CHIP_PLL_CTRL */ + 0x0000, /* 0x0034, CHIP_CLK_TOP_CTRL */ + 0x0000, /* 0x0036, CHIP_ANA_STATUS */ + 0x0000, /* 0x0038, reserved */ + 0x0000, /* 0x003A, CHIP_ANA_TEST2 */ + 0x0000, /* 0x003C, CHIP_SHORT_CTRL */ + 0x0000, /* reserved */ +}; + +/* default value of dap registers */ +static const u16 sgtl5000_dap_regs[] = { + 0x0000, /* 0x0100, DAP_CONTROL */ + 0x0000, /* 0x0102, DAP_PEQ */ + 0x0040, /* 0x0104, DAP_BASS_ENHANCE */ + 0x051f, /* 0x0106, DAP_BASS_ENHANCE_CTRL */ + 0x0000, /* 0x0108, DAP_AUDIO_EQ */ + 0x0040, /* 0x010A, DAP_SGTL_SURROUND */ + 0x0000, /* 0x010C, DAP_FILTER_COEF_ACCESS */ + 0x0000, /* 0x010E, DAP_COEF_WR_B0_MSB */ + 0x0000, /* 0x0110, DAP_COEF_WR_B0_LSB */ + 0x0000, /* 0x0112, reserved */ + 0x0000, /* 0x0114, reserved */ + 0x002f, /* 0x0116, DAP_AUDIO_EQ_BASS_BAND0 */ + 0x002f, /* 0x0118, DAP_AUDIO_EQ_BAND0 */ + 0x002f, /* 0x011A, DAP_AUDIO_EQ_BAND2 */ + 0x002f, /* 0x011C, DAP_AUDIO_EQ_BAND3 */ + 0x002f, /* 0x011E, DAP_AUDIO_EQ_TREBLE_BAND4 */ + 0x8000, /* 0x0120, DAP_MAIN_CHAN */ + 0x0000, /* 0x0122, DAP_MIX_CHAN */ + 0x0510, /* 0x0124, DAP_AVC_CTRL */ + 0x1473, /* 0x0126, DAP_AVC_THRESHOLD */ + 0x0028, /* 0x0128, DAP_AVC_ATTACK */ + 0x0050, /* 0x012A, DAP_AVC_DECAY */ + 0x0000, /* 0x012C, DAP_COEF_WR_B1_MSB */ + 0x0000, /* 0x012E, DAP_COEF_WR_B1_LSB */ + 0x0000, /* 0x0130, DAP_COEF_WR_B2_MSB */ + 0x0000, /* 0x0132, DAP_COEF_WR_B2_LSB */ + 0x0000, /* 0x0134, DAP_COEF_WR_A1_MSB */ + 0x0000, /* 0x0136, DAP_COEF_WR_A1_LSB */ + 0x0000, /* 0x0138, DAP_COEF_WR_A2_MSB */ + 0x0000, /* 0x013A, DAP_COEF_WR_A2_LSB */ +}; + +/* regulator supplies for sgtl5000, VDDD is an option external supply */ +enum sgtl5000_regulator_supplies { + VDDA, + VDDIO, + VDDD, + SGTL5000_SUPPLY_NUM +}; + +/* vddd is optinal supply */ +static const char *supply_names[SGTL5000_SUPPLY_NUM] = { + "VDDA", + "VDDIO", + "VDDD" +}; + +/* sgtl5000 private structure in codec */ +struct sgtl5000_priv { + int sysclk; /* sysclk rate */ + int master; /* i2s master or not */ + int fmt; /* i2s data format */ + struct regulator *supplies[SGTL5000_SUPPLY_NUM]; /* all regulators. */ +}; + +/* + * mic_bias power on/off share the same register bits with + * output impedance of mic bias, when power on mic bias, we + * need reclaim it to impedance value. + * 0x0 = Powered off + * 0x1 = 2Kohm + * 0x2 = 4Kohm + * 0x3 = 8Kohm + */ +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* change mic bias resistor to 4Kohm */ + snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_4k, SGTL5000_BIAS_R_4k); + + return 0; +} + +/* + * using codec assist to small pop, hp_powerup or lineout_powerup + * should stay setting until vag_powerup is fully ramped down, + * vag fully ramped down require 400ms. + */ +static int small_pop_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, 0); + msleep(400); + break; + default: + break; + } + + return 0; +} + +/* input sources for ADC */ +static const char *adc_mux_text[] = { + "MIC_IN", "LINE_IN" +}; + +static const struct soc_enum adc_enum = +SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 2, 2, adc_mux_text); + +static const struct snd_kcontrol_new adc_mux = +SOC_DAPM_ENUM("Capture Mux", adc_enum); + +/* input sources for DAC */ +static const char *dac_mux_text[] = { + "DAC", "LINE_IN" +}; + +static const struct soc_enum dac_enum = +SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 6, 2, dac_mux_text); + +static const struct snd_kcontrol_new dac_mux = +SOC_DAPM_ENUM("Headphone Mux", dac_enum); + +static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINE_IN"), + SND_SOC_DAPM_INPUT("MIC_IN"), + + SND_SOC_DAPM_OUTPUT("HP_OUT"), + SND_SOC_DAPM_OUTPUT("LINE_OUT"), + + SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, + mic_bias_event, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0, + small_pop_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0, + small_pop_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), + SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), + + /* aif for i2s input */ + SND_SOC_DAPM_AIF_IN("AIFIN", "Playback", + 0, SGTL5000_CHIP_DIG_POWER, + 0, 0), + + /* aif for i2s output */ + SND_SOC_DAPM_AIF_OUT("AIFOUT", "Capture", + 0, SGTL5000_CHIP_DIG_POWER, + 1, 0), + + SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), + + SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), +}; + +/* routes for sgtl5000 */ +static const struct snd_soc_dapm_route audio_map[] = { + {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ + {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ + + {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ + {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ + + {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ + {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ + {"LO", NULL, "DAC"}, /* dac --> line_out */ + + {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ + {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ + + {"LINE_OUT", NULL, "LO"}, + {"HP_OUT", NULL, "HP"}, +}; + +/* custom function to fetch info of PCM playback volume */ +static int dac_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xfc - 0x3c; + return 0; +} + +/* + * custom function to get of PCM playback volume + * PCM volume with 0.5017 dB steps from 0 to -90 dB + * 0x3B and less = Reserved + * 0x3C = 0 dB + * 0x3D = -0.5 dB + * 0xF0 = -90 dB + * 0xFC and greater = Muted + */ +static int dac_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg, l, r; + + reg = snd_soc_read(codec, SGTL5000_CHIP_DAC_VOL); + l = (reg & SGTL5000_DAC_VOL_LEFT_MASK) >> SGTL5000_DAC_VOL_LEFT_SHIFT; + r = (reg & SGTL5000_DAC_VOL_RIGHT_MASK) >> SGTL5000_DAC_VOL_RIGHT_SHIFT; + l = l < 0x3c ? 0x3c : l; + l = l > 0xfc ? 0xfc : l; + r = r < 0x3c ? 0x3c : r; + r = r > 0xfc ? 0xfc : r; + + /* revert it */ + l = 0xfc - l; + r = 0xfc - r; + + ucontrol->value.integer.value[0] = l; + ucontrol->value.integer.value[1] = r; + + return 0; +} + +/* + * custom put function for PCM playback volume + * PCM volume with 0.5017 dB steps from 0 to -90 dB + * 0x3B and less = Reserved + * 0x3C = 0 dB + * 0x3D = -0.5 dB + * 0xF0 = -90 dB + * 0xFC and greater = Muted + */ +static int dac_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg, l, r; + + l = ucontrol->value.integer.value[0]; + r = ucontrol->value.integer.value[1]; + + l = l < 0 ? 0 : l; + l = l > 0xfc - 0x3c ? 0xfc - 0x3c : l; + r = r < 0 ? 0 : r; + r = r > 0xfc - 0x3c ? 0xfc - 0x3c : r; + l = 0xfc - l; + r = 0xfc - r; + + reg = l << SGTL5000_DAC_VOL_LEFT_SHIFT | + r << SGTL5000_DAC_VOL_RIGHT_SHIFT; + + snd_soc_update_bits_locked(codec, SGTL5000_CHIP_DAC_VOL, reg, reg); + + return 0; +} + +/* need SOC_DOUBLE_S8_TLV with invert */ +#define SGTL5000_PCM_PLAYBACK_CONTROL \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "PCM Playback Volume", \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = dac_info_volsw, \ + .get = dac_get_volsw, \ + .put = dac_put_volsw, \ + } + +/* tlv for mic gain, 0db 20db 30db 40db */ +static const unsigned int mic_gain_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0), +}; + +/* tlv for hp volume, -51.5db to 12.0db, step .5db */ +static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); + +static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { + SGTL5000_PCM_PLAYBACK_CONTROL, + + SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0), + SOC_SINGLE("Capture Attenuate Switch (-6db)", + SGTL5000_CHIP_ANA_ADC_CTRL, + 8, 1, 0), + SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0), + + SOC_DOUBLE_TLV("Headphone Playback Volume", + SGTL5000_CHIP_ANA_HP_CTRL, + 0, 8, + 0x7f, 1, + headphone_volume), + SOC_SINGLE("Headphone Playback ZC Switch", SGTL5000_CHIP_ANA_CTRL, + 5, 1, 0), + + SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, + 0, 4, 0, mic_gain_tlv), +}; + +/* mute the codec used by alsa core */ +static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + u16 adcdac_ctrl = SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT; + + if (mute) + snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, + adcdac_ctrl, adcdac_ctrl); + else + snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, + adcdac_ctrl, 0); + + return 0; +} + +/* set codec format */ +static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + u16 i2sctl = 0; + + sgtl5000->master = 0; + /* + * i2s clock and frame master setting. + * ONLY support: + * - clock and frame slave, + * - clock and frame master + */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBM_CFM: + i2sctl |= SGTL5000_I2S_MASTER; + sgtl5000->master = 1; + break; + default: + return -EINVAL; + } + + /* setting i2s data format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + i2sctl |= SGTL5000_I2S_MODE_PCM; + break; + case SND_SOC_DAIFMT_DSP_B: + i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_LRALIGN; + break; + case SND_SOC_DAIFMT_I2S: + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + break; + case SND_SOC_DAIFMT_RIGHT_J: + i2sctl |= SGTL5000_I2S_MODE_RJ; + i2sctl |= SGTL5000_I2S_LRPOL; + break; + case SND_SOC_DAIFMT_LEFT_J: + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_LRALIGN; + break; + default: + return -EINVAL; + } + + sgtl5000->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + /* Clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + i2sctl |= SGTL5000_I2S_SCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, SGTL5000_CHIP_I2S_CTRL, i2sctl); + + return 0; +} + +/* set codec sysclk */ +static int sgtl5000_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case SGTL5000_SYSCLK: + sgtl5000->sysclk = freq; + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * set clock according to i2s frame clock, + * sgtl5000 provide 2 clock sources. + * 1. sys_mclk. sample freq can only configure to + * 1/256, 1/384, 1/512 of sys_mclk. + * 2. pll. can derive any audio clocks. + * + * clock setting rules: + * 1. in slave mode, only sys_mclk can use. + * 2. as constraint by sys_mclk, sample freq should + * set to 32k, 44.1k and above. + * 3. using sys_mclk prefer to pll to save power. + */ +static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int clk_ctl = 0; + int sys_fs; /* sample freq */ + + /* + * sample freq should be divided by frame clock, + * if frame clock lower than 44.1khz, sample feq should set to + * 32khz or 44.1khz. + */ + switch (frame_rate) { + case 8000: + case 16000: + sys_fs = 32000; + break; + case 11025: + case 22050: + sys_fs = 44100; + break; + default: + sys_fs = frame_rate; + break; + } + + /* set devided factor of frame clock */ + switch (sys_fs / frame_rate) { + case 4: + clk_ctl |= SGTL5000_RATE_MODE_DIV_4 << SGTL5000_RATE_MODE_SHIFT; + break; + case 2: + clk_ctl |= SGTL5000_RATE_MODE_DIV_2 << SGTL5000_RATE_MODE_SHIFT; + break; + /* + * this implict case 1: + * because default: sys_fs = lrclk; + */ + default: + clk_ctl |= SGTL5000_RATE_MODE_DIV_1 << SGTL5000_RATE_MODE_SHIFT; + break; + } + + /* set the sys_fs accroding to frame rate */ + switch (sys_fs) { + case 32000: + clk_ctl |= SGTL5000_SYS_FS_32k << SGTL5000_SYS_FS_SHIFT; + break; + case 44100: + clk_ctl |= SGTL5000_SYS_FS_44_1k << SGTL5000_SYS_FS_SHIFT; + break; + case 48000: + clk_ctl |= SGTL5000_SYS_FS_48k << SGTL5000_SYS_FS_SHIFT; + break; + case 96000: + clk_ctl |= SGTL5000_SYS_FS_96k << SGTL5000_SYS_FS_SHIFT; + break; + default: + dev_err(codec->dev, "frame rate %d not supported\n", + frame_rate); + return -EINVAL; + } + + /* + * calculate the divider of mclk/sample_freq, + * factor of freq =96k can only be 256, since mclk in range (12m,27m) + */ + if (sys_fs * 256 == sgtl5000->sysclk) + clk_ctl |= SGTL5000_MCLK_FREQ_256FS << + SGTL5000_MCLK_FREQ_SHIFT; + else if (sys_fs * 384 == sgtl5000->sysclk) + clk_ctl |= SGTL5000_MCLK_FREQ_384FS << + SGTL5000_MCLK_FREQ_SHIFT; + else if (sys_fs * 512 == sgtl5000->sysclk) + clk_ctl |= SGTL5000_MCLK_FREQ_512FS << + SGTL5000_MCLK_FREQ_SHIFT; + else { + /* if mclk not satisify the divider, using pll */ + if (sgtl5000->master) + clk_ctl |= SGTL5000_MCLK_FREQ_PLL << + SGTL5000_MCLK_FREQ_SHIFT; + else { + pr_err("%s: PLL not supported in slave mode\n", + __func__); + return -EINVAL; + } + } + + /* if using pll, please check manual 6.4.2 for detail */ + if ((clk_ctl & SGTL5000_MCLK_FREQ_MASK) == SGTL5000_MCLK_FREQ_PLL) { + u64 out, t; + int div2; + int pll_ctl; + unsigned int in, int_div, frac_div; + + if (sgtl5000->sysclk > 17000000) { + div2 = 1; + in = sgtl5000->sysclk / 2; + } else { + div2 = 0; + in = sgtl5000->sysclk; + } + if (sys_fs == 44100) + out = 180633600; + else + out = 196608000; + t = do_div(out, in); + int_div = out; + t *= 2048; + do_div(t, in); + frac_div = t; + pll_ctl = int_div << SGTL5000_PLL_INT_DIV_SHIFT | + frac_div << SGTL5000_PLL_FRAC_DIV_SHIFT; + + snd_soc_write(codec, SGTL5000_CHIP_PLL_CTRL, pll_ctl); + if (div2) + snd_soc_update_bits(codec, + SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INPUT_FREQ_DIV2, + SGTL5000_INPUT_FREQ_DIV2); + else + snd_soc_update_bits(codec, + SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INPUT_FREQ_DIV2, + 0); + + /* power up pll */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP); + } else + /* power down pll */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, + 0); + + /* if using pll, clk_ctrl must be set after pll power up */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); + + return 0; +} + +/* + * Set PCM DAI bit size and sample rate. + * input: params_rate, params_fmt + */ +static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int channels = params_channels(params); + int i2s_ctl = 0; + int stereo; + int ret; + + /* sysclk should already set */ + if (!sgtl5000->sysclk) { + dev_err(codec->dev, "%s: set sysclk first!\n", __func__); + return -EFAULT; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + stereo = SGTL5000_DAC_STEREO; + else + stereo = SGTL5000_ADC_STEREO; + + /* set mono to save power */ + if (channels == 1) + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, stereo, 0); + else + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + stereo, stereo); + + /* set codec clock base on lrclk */ + ret = sgtl5000_set_clock(codec, params_rate(params)); + if (ret) + return ret; + + /* set i2s data format */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) + return -EINVAL; + i2s_ctl |= SGTL5000_I2S_DLEN_16 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_32FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + i2s_ctl |= SGTL5000_I2S_DLEN_20 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + i2s_ctl |= SGTL5000_I2S_DLEN_24 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S32_LE: + if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) + return -EINVAL; + i2s_ctl |= SGTL5000_I2S_DLEN_32 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, i2s_ctl, i2s_ctl); + + return 0; +} + +/* + * set dac bias + * common state changes: + * startup: + * off --> standby --> prepare --> on + * standby --> prepare --> on + * + * stop: + * on --> prepare --> standby + */ +static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int i; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + for (i = 0; i < SGTL5000_SUPPLY_NUM; i++) { + if (!sgtl5000->supplies[i]) + continue; + regulator_enable(sgtl5000->supplies[i]); + } + } + break; + case SND_SOC_BIAS_OFF: + for (i = 0; i < SGTL5000_SUPPLY_NUM; i++) { + if (!sgtl5000->supplies[i]) + continue; + regulator_disable(sgtl5000->supplies[i]); + } + break; + } + + codec->bias_level = level; + return 0; +} + +#define SGTL5000_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai_ops sgtl5000_ops = { + .hw_params = sgtl5000_pcm_hw_params, + .digital_mute = sgtl5000_digital_mute, + .set_fmt = sgtl5000_set_dai_fmt, + .set_sysclk = sgtl5000_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver sgtl5000_dai = { + .name = "sgtl5000", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + /* + * only support 8~48K + 96K, + * TODO modify hw_param to support more + */ + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000, + .formats = SGTL5000_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000, + .formats = SGTL5000_FORMATS, + }, + .ops = &sgtl5000_ops, + .symmetric_rates = 1, +}; + +static int sgtl5000_volatile_register(unsigned int reg) +{ + switch (reg) { + case SGTL5000_CHIP_ID: + case SGTL5000_CHIP_ADCDAC_CTRL: + case SGTL5000_CHIP_ANA_STATUS: + return 1; + } + + return 0; +} + +static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +/* + * restore all sgtl5000 registers, + * since a big hole between dap and regular registers, + * we will restore them respectively. + */ +static int sgtl5000_restore_regs(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i; + int regular_regs = SGTL5000_CHIP_SHORT_CTRL >> 1; + + /* restore regular registers */ + for (i = 0; i < regular_regs; i++) { + int reg = i << 1; + + /* this regs depends on the others */ + if (reg == SGTL5000_CHIP_ANA_POWER || + reg == SGTL5000_CHIP_CLK_CTRL || + reg == SGTL5000_CHIP_LINREG_CTRL || + reg == SGTL5000_CHIP_LINE_OUT_CTRL || + reg == SGTL5000_CHIP_CLK_CTRL) + continue; + + snd_soc_write(codec, reg, cache[i]); + } + + /* restore dap registers */ + for (i = SGTL5000_DAP_REG_OFFSET >> 1; + i < SGTL5000_MAX_REG_OFFSET >> 1; i++) { + int reg = i << 1; + + snd_soc_write(codec, reg, cache[i]); + } + + /* + * restore power and other regs accroding + * to set_power() and set_clock() + */ + snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, + cache[SGTL5000_CHIP_LINREG_CTRL >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, + cache[SGTL5000_CHIP_ANA_POWER >> 1]); + mdelay(1); + + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, + cache[SGTL5000_CHIP_CLK_CTRL >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, + cache[SGTL5000_CHIP_REF_CTRL >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, + cache[SGTL5000_CHIP_LINE_OUT_CTRL >> 1]); + return 0; +} + +static int sgtl5000_resume(struct snd_soc_codec *codec) +{ + /* Bring the codec back up to standby to enable regulators */ + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Restore registers by cached in memory */ + sgtl5000_restore_regs(codec); + return 0; +} + +/* + * sgtl5000 has 3 internal power supplies: + * 1. VAG, normally set to vdda/2 + * 2. chargepump, set to different value + * according to voltage of vdda and vddio + * 3. line out VAG, normally set to vddio/2 + * + * and should be set accroding to: + * 1. vddd provided by external or not + * 2. vdda and vddio voltage value. > 3.1v or not + * 3. chip revision >=0x11 or not. If >=0x11, not use external vddd. + */ +static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) +{ + int vddd, vdda, vddio; + u16 ana_pwr, lreg_ctrl; + int vag; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + vdda = regulator_get_voltage(sgtl5000->supplies[VDDA]) / 1000; + vddio = regulator_get_voltage(sgtl5000->supplies[VDDIO]) / 1000; + + if (sgtl5000->supplies[VDDD]) + vddd = regulator_get_voltage(sgtl5000->supplies[VDDD]) / 1000; + else + vddd = 0; + + /* reset value */ + ana_pwr = SGTL5000_DAC_STEREO | + SGTL5000_ADC_STEREO | + SGTL5000_REFTOP_POWERUP; + lreg_ctrl = 0; + + /* if no external vddd, use internal vddd */ + if (!vddd) { + /* set VDDD to 1.2v */ + lreg_ctrl |= 0x8 << SGTL5000_LINREG_VDDD_SHIFT; + /* power up internal linear regulator */ + ana_pwr |= SGTL5000_LINEREG_D_POWERUP | + SGTL5000_LINREG_SIMPLE_POWERUP | + SGTL5000_STARTUP_POWERUP; + } + + if (vddio < 3100 && vdda < 3100) { + /* enable internal oscillator used for charge pump */ + snd_soc_update_bits(codec, SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INT_OSC_EN, + SGTL5000_INT_OSC_EN); + /* Enable VDDC charge pump */ + ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP; + } else if (vddio >= 3100 && vdda >= 3100) { + /* + * if vddio and vddd > 3.1v, + * charge pump should be clean before set ana_pwr + */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VDDC_CHRGPMP_POWERUP, 0); + + /* VDDC use VDDIO rail */ + lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD; + lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO << + SGTL5000_VDDC_MAN_ASSN_SHIFT; + } + + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr); + mdelay(1); + + snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl); + + /* + * if vddd linear reg has been enabled, + * simple digital supply should be clear to get + * proper VDDD voltage. + */ + if (ana_pwr & SGTL5000_LINEREG_D_POWERUP) { + ana_pwr &= ~SGTL5000_LINREG_SIMPLE_POWERUP; + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINREG_SIMPLE_POWERUP | + SGTL5000_STARTUP_POWERUP, + 0); + mdelay(1); + } + + /* + * set ADC/DAC VAG to vdda / 2, + * should stay in range (0.8v, 1.575v) + */ + vag = vdda / 2; + if (vag <= SGTL5000_ANA_GND_BASE) + vag = 0; + else if (vag >= SGTL5000_ANA_GND_BASE + SGTL5000_ANA_GND_STP * + (SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT)) + vag = SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT; + else + vag = (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP; + + snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, + vag << SGTL5000_ANA_GND_SHIFT, + vag << SGTL5000_ANA_GND_SHIFT); + + /* set line out VAG to vddio / 2, in range (0.8v, 1.675v) */ + vag = vddio / 2; + if (vag <= SGTL5000_LINE_OUT_GND_BASE) + vag = 0; + else if (vag >= SGTL5000_LINE_OUT_GND_BASE + + SGTL5000_LINE_OUT_GND_STP * SGTL5000_LINE_OUT_GND_MAX) + vag = SGTL5000_LINE_OUT_GND_MAX; + else + vag = (vag - SGTL5000_LINE_OUT_GND_BASE) / + SGTL5000_LINE_OUT_GND_STP; + + snd_soc_update_bits(codec, SGTL5000_CHIP_LINE_OUT_CTRL, + vag << SGTL5000_LINE_OUT_GND_SHIFT | + SGTL5000_LINE_OUT_CURRENT_360u << + SGTL5000_LINE_OUT_CURRENT_SHIFT, + vag << SGTL5000_LINE_OUT_GND_SHIFT | + SGTL5000_LINE_OUT_CURRENT_360u << + SGTL5000_LINE_OUT_CURRENT_SHIFT); + + return 0; +} + +static int sgtl5000_probe(struct snd_soc_codec *codec) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + u16 reg; + int ret; + int rev; + int i; + + /* setup i2c data ops */ + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* read chip information */ + reg = snd_soc_read(codec, SGTL5000_CHIP_ID); + if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != + SGTL5000_PARTID_PART_ID) { + dev_err(codec->dev, + "Device with ID register %x is not a sgtl5000\n", reg); + ret = -ENODEV; + goto err_out; + } + + rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + dev_info(codec->dev, "sgtl5000 revision %d\n", rev); + + /* get and enable all regulators */ + for (i = 0; i < SGTL5000_SUPPLY_NUM; i++) { + struct regulator *reg; + + /* workaround for revision 0x11, using internal LDO */ + if (i == VDDD && rev == 0x11) + continue; + + reg = regulator_get(codec->dev, supply_names[i]); + + if (IS_ERR(reg)) + continue; + + regulator_enable(reg); + sgtl5000->supplies[i] = reg; + } + + /* + * vdda and vddio regulator must configured, + * vddd is an optional regulator, if vddd not supply externally, + * internal vddd will be enabled in sgtl5000_set_power_regs() + */ + if (!sgtl5000->supplies[VDDA] || !sgtl5000->supplies[VDDIO]) { + dev_err(codec->dev, + "Not set vdda or vddio regulator correctly\n"); + + /* platform regulator not configured correctly */ + ret = -ENODEV; + goto err_out; + } + + /* power up sgtl5000 */ + ret = sgtl5000_set_power_regs(codec); + if (ret) + goto err_out; + + /* enable small pop, introduce 400ms delay in turning off */ + snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, + SGTL5000_SMALL_POP, + SGTL5000_SMALL_POP); + + /* enable short cut detect */ + snd_soc_write(codec, SGTL5000_CHIP_SHORT_CTRL, 0); + + /* + * set i2s as default input of sound switch + * TODO: add sound switch to control and dapm widge. + */ + snd_soc_write(codec, SGTL5000_CHIP_SSS_CTRL, + SGTL5000_DAC_SEL_I2S_IN << SGTL5000_DAC_SEL_SHIFT); + snd_soc_write(codec, SGTL5000_CHIP_DIG_POWER, + SGTL5000_ADC_EN | SGTL5000_DAC_EN); + + /* enable dac vol ramp by default */ + snd_soc_write(codec, SGTL5000_CHIP_ADCDAC_CTRL, + SGTL5000_DAC_VOL_RAMP_EN | + SGTL5000_DAC_MUTE_RIGHT | + SGTL5000_DAC_MUTE_LEFT); + + snd_soc_write(codec, SGTL5000_CHIP_PAD_STRENGTH, 0x015f); + + /* set default volume of adc */ + reg = snd_soc_read(codec, SGTL5000_CHIP_ANA_ADC_CTRL); + reg &= ~SGTL5000_ADC_VOL_M6DB; + reg &= ~(SGTL5000_ADC_VOL_LEFT_MASK | SGTL5000_ADC_VOL_RIGHT_MASK); + reg |= (0xf << SGTL5000_ADC_VOL_LEFT_SHIFT) + | (0xf << SGTL5000_ADC_VOL_RIGHT_SHIFT); + snd_soc_write(codec, SGTL5000_CHIP_ANA_ADC_CTRL, reg); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_CTRL, + SGTL5000_HP_ZCD_EN | + SGTL5000_ADC_ZCD_EN); + + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + + /* + * disable DAP + * TODO: + * Enable DAP in control and dapm. + */ + snd_soc_write(codec, SGTL5000_DAP_CTRL, 0); + + /* leading to standby state */ + ret = sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto err_out; + + snd_soc_add_controls(codec, sgtl5000_snd_controls, + ARRAY_SIZE(sgtl5000_snd_controls)); + + snd_soc_dapm_new_controls(codec, sgtl5000_dapm_widgets, + ARRAY_SIZE(sgtl5000_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + + return 0; + +err_out: + for (i = 0; i < SGTL5000_SUPPLY_NUM; i++) { + if (!sgtl5000->supplies[i]) + continue; + + regulator_disable(sgtl5000->supplies[i]); + regulator_put(sgtl5000->supplies[i]); + } + + return ret; +} + +static int sgtl5000_remove(struct snd_soc_codec *codec) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int i; + + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_dapm_free(codec); + + for (i = 0; i < SGTL5000_SUPPLY_NUM; i++) { + if (!sgtl5000->supplies[i]) + continue; + + regulator_disable(sgtl5000->supplies[i]); + regulator_put(sgtl5000->supplies[i]); + } + + return 0; +} + +struct snd_soc_codec_driver sgtl5000_driver = { + .probe = sgtl5000_probe, + .remove = sgtl5000_remove, + .suspend = sgtl5000_suspend, + .resume = sgtl5000_resume, + .set_bias_level = sgtl5000_set_bias_level, + .reg_cache_size = ARRAY_SIZE(sgtl5000_regs), + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, + .reg_cache_default = sgtl5000_regs, + .volatile_register = sgtl5000_volatile_register, +}; + +static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct sgtl5000_priv *sgtl5000; + int ret; + + sgtl5000 = kzalloc(sizeof(struct sgtl5000_priv), GFP_KERNEL); + if (!sgtl5000) + return -ENOMEM; + + /* + * copy DAP default values to default value array. + * sgtl5000 register space has a big hole, merge it + * at init phase makes life easy. + * FIXME: should we drop 'const' of sgtl5000_regs? + */ + memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)), + sgtl5000_dap_regs, + 0x3a); + + i2c_set_clientdata(client, sgtl5000); + + ret = snd_soc_register_codec(&client->dev, + &sgtl5000_driver, &sgtl5000_dai, 1); + if (ret) { + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + kfree(sgtl5000); + return ret; + } + + return 0; +} + +static __devexit int sgtl5000_i2c_remove(struct i2c_client *client) +{ + struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + kfree(sgtl5000); + return 0; +} + +static const struct i2c_device_id sgtl5000_id[] = { + {"sgtl5000", 0}, + {}, +}; + +MODULE_DEVICE_TABLE(i2c, sgtl5000_id); + +static struct i2c_driver sgtl5000_i2c_driver = { + .driver = { + .name = "sgtl5000", + .owner = THIS_MODULE, + }, + .probe = sgtl5000_i2c_probe, + .remove = __devexit_p(sgtl5000_i2c_remove), + .id_table = sgtl5000_id, +}; + +static int __init sgtl5000_modinit(void) +{ + return i2c_add_driver(&sgtl5000_i2c_driver); +} +module_init(sgtl5000_modinit); + +static void __exit sgtl5000_exit(void) +{ + i2c_del_driver(&sgtl5000_i2c_driver); +} +module_exit(sgtl5000_exit); + +MODULE_DESCRIPTION("ASoC sgtl5000 driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h new file mode 100644 index 0000000..7f24655 --- /dev/null +++ b/sound/soc/codecs/sgtl5000.h @@ -0,0 +1,403 @@ +/* + * sgtl5000.h - SGTL5000 audio codec interface + * + * Copyright 2008-2009 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _SGTL5000_H +#define _SGTL5000_H + +#include <linux/i2c.h> + +/* + * Register values. + */ +#define SGTL5000_CHIP_ID 0x0000 +#define SGTL5000_CHIP_DIG_POWER 0x0002 +#define SGTL5000_CHIP_CLK_CTRL 0x0004 +#define SGTL5000_CHIP_I2S_CTRL 0x0006 +#define SGTL5000_CHIP_SSS_CTRL 0x000a +#define SGTL5000_CHIP_ADCDAC_CTRL 0x000e +#define SGTL5000_CHIP_DAC_VOL 0x0010 +#define SGTL5000_CHIP_PAD_STRENGTH 0x0014 +#define SGTL5000_CHIP_ANA_ADC_CTRL 0x0020 +#define SGTL5000_CHIP_ANA_HP_CTRL 0x0022 +#define SGTL5000_CHIP_ANA_CTRL 0x0024 +#define SGTL5000_CHIP_LINREG_CTRL 0x0026 +#define SGTL5000_CHIP_REF_CTRL 0x0028 +#define SGTL5000_CHIP_MIC_CTRL 0x002a +#define SGTL5000_CHIP_LINE_OUT_CTRL 0x002c +#define SGTL5000_CHIP_LINE_OUT_VOL 0x002e +#define SGTL5000_CHIP_ANA_POWER 0x0030 +#define SGTL5000_CHIP_PLL_CTRL 0x0032 +#define SGTL5000_CHIP_CLK_TOP_CTRL 0x0034 +#define SGTL5000_CHIP_ANA_STATUS 0x0036 +#define SGTL5000_CHIP_SHORT_CTRL 0x003c +#define SGTL5000_CHIP_ANA_TEST2 0x003a +#define SGTL5000_DAP_CTRL 0x0100 +#define SGTL5000_DAP_PEQ 0x0102 +#define SGTL5000_DAP_BASS_ENHANCE 0x0104 +#define SGTL5000_DAP_BASS_ENHANCE_CTRL 0x0106 +#define SGTL5000_DAP_AUDIO_EQ 0x0108 +#define SGTL5000_DAP_SURROUND 0x010a +#define SGTL5000_DAP_FLT_COEF_ACCESS 0x010c +#define SGTL5000_DAP_COEF_WR_B0_MSB 0x010e +#define SGTL5000_DAP_COEF_WR_B0_LSB 0x0110 +#define SGTL5000_DAP_EQ_BASS_BAND0 0x0116 +#define SGTL5000_DAP_EQ_BASS_BAND1 0x0118 +#define SGTL5000_DAP_EQ_BASS_BAND2 0x011a +#define SGTL5000_DAP_EQ_BASS_BAND3 0x011c +#define SGTL5000_DAP_EQ_BASS_BAND4 0x011e +#define SGTL5000_DAP_MAIN_CHAN 0x0120 +#define SGTL5000_DAP_MIX_CHAN 0x0122 +#define SGTL5000_DAP_AVC_CTRL 0x0124 +#define SGTL5000_DAP_AVC_THRESHOLD 0x0126 +#define SGTL5000_DAP_AVC_ATTACK 0x0128 +#define SGTL5000_DAP_AVC_DECAY 0x012a +#define SGTL5000_DAP_COEF_WR_B1_MSB 0x012c +#define SGTL5000_DAP_COEF_WR_B1_LSB 0x012e +#define SGTL5000_DAP_COEF_WR_B2_MSB 0x0130 +#define SGTL5000_DAP_COEF_WR_B2_LSB 0x0132 +#define SGTL5000_DAP_COEF_WR_A1_MSB 0x0134 +#define SGTL5000_DAP_COEF_WR_A1_LSB 0x0136 +#define SGTL5000_DAP_COEF_WR_A2_MSB 0x0138 +#define SGTL5000_DAP_COEF_WR_A2_LSB 0x013a + +/* + * Field Definitions. + */ + +/* + * SGTL5000_CHIP_ID + */ +#define SGTL5000_PARTID_MASK 0xff00 +#define SGTL5000_PARTID_SHIFT 8 +#define SGTL5000_PARTID_WIDTH 8 +#define SGTL5000_PARTID_PART_ID 0xa0 +#define SGTL5000_REVID_MASK 0x00ff +#define SGTL5000_REVID_SHIFT 0 +#define SGTL5000_REVID_WIDTH 8 + +/* + * SGTL5000_CHIP_DIG_POWER + */ +#define SGTL5000_ADC_EN 0x0040 +#define SGTL5000_DAC_EN 0x0020 +#define SGTL5000_DAP_POWERUP 0x0010 +#define SGTL5000_I2S_OUT_POWERUP 0x0002 +#define SGTL5000_I2S_IN_POWERUP 0x0001 + +/* + * SGTL5000_CHIP_CLK_CTRL + */ +#define SGTL5000_RATE_MODE_MASK 0x0030 +#define SGTL5000_RATE_MODE_SHIFT 4 +#define SGTL5000_RATE_MODE_WIDTH 2 +#define SGTL5000_RATE_MODE_DIV_1 0 +#define SGTL5000_RATE_MODE_DIV_2 1 +#define SGTL5000_RATE_MODE_DIV_4 2 +#define SGTL5000_RATE_MODE_DIV_6 3 +#define SGTL5000_SYS_FS_MASK 0x000c +#define SGTL5000_SYS_FS_SHIFT 2 +#define SGTL5000_SYS_FS_WIDTH 2 +#define SGTL5000_SYS_FS_32k 0x0 +#define SGTL5000_SYS_FS_44_1k 0x1 +#define SGTL5000_SYS_FS_48k 0x2 +#define SGTL5000_SYS_FS_96k 0x3 +#define SGTL5000_MCLK_FREQ_MASK 0x0003 +#define SGTL5000_MCLK_FREQ_SHIFT 0 +#define SGTL5000_MCLK_FREQ_WIDTH 2 +#define SGTL5000_MCLK_FREQ_256FS 0x0 +#define SGTL5000_MCLK_FREQ_384FS 0x1 +#define SGTL5000_MCLK_FREQ_512FS 0x2 +#define SGTL5000_MCLK_FREQ_PLL 0x3 + +/* + * SGTL5000_CHIP_I2S_CTRL + */ +#define SGTL5000_I2S_SCLKFREQ_MASK 0x0100 +#define SGTL5000_I2S_SCLKFREQ_SHIFT 8 +#define SGTL5000_I2S_SCLKFREQ_WIDTH 1 +#define SGTL5000_I2S_SCLKFREQ_64FS 0x0 +#define SGTL5000_I2S_SCLKFREQ_32FS 0x1 /* Not for RJ mode */ +#define SGTL5000_I2S_MASTER 0x0080 +#define SGTL5000_I2S_SCLK_INV 0x0040 +#define SGTL5000_I2S_DLEN_MASK 0x0030 +#define SGTL5000_I2S_DLEN_SHIFT 4 +#define SGTL5000_I2S_DLEN_WIDTH 2 +#define SGTL5000_I2S_DLEN_32 0x0 +#define SGTL5000_I2S_DLEN_24 0x1 +#define SGTL5000_I2S_DLEN_20 0x2 +#define SGTL5000_I2S_DLEN_16 0x3 +#define SGTL5000_I2S_MODE_MASK 0x000c +#define SGTL5000_I2S_MODE_SHIFT 2 +#define SGTL5000_I2S_MODE_WIDTH 2 +#define SGTL5000_I2S_MODE_I2S_LJ 0x0 +#define SGTL5000_I2S_MODE_RJ 0x1 +#define SGTL5000_I2S_MODE_PCM 0x2 +#define SGTL5000_I2S_LRALIGN 0x0002 +#define SGTL5000_I2S_LRPOL 0x0001 /* set for which mode */ + +/* + * SGTL5000_CHIP_SSS_CTRL + */ +#define SGTL5000_DAP_MIX_LRSWAP 0x4000 +#define SGTL5000_DAP_LRSWAP 0x2000 +#define SGTL5000_DAC_LRSWAP 0x1000 +#define SGTL5000_I2S_OUT_LRSWAP 0x0400 +#define SGTL5000_DAP_MIX_SEL_MASK 0x0300 +#define SGTL5000_DAP_MIX_SEL_SHIFT 8 +#define SGTL5000_DAP_MIX_SEL_WIDTH 2 +#define SGTL5000_DAP_MIX_SEL_ADC 0x0 +#define SGTL5000_DAP_MIX_SEL_I2S_IN 0x1 +#define SGTL5000_DAP_SEL_MASK 0x00c0 +#define SGTL5000_DAP_SEL_SHIFT 6 +#define SGTL5000_DAP_SEL_WIDTH 2 +#define SGTL5000_DAP_SEL_ADC 0x0 +#define SGTL5000_DAP_SEL_I2S_IN 0x1 +#define SGTL5000_DAC_SEL_MASK 0x0030 +#define SGTL5000_DAC_SEL_SHIFT 4 +#define SGTL5000_DAC_SEL_WIDTH 2 +#define SGTL5000_DAC_SEL_ADC 0x0 +#define SGTL5000_DAC_SEL_I2S_IN 0x1 +#define SGTL5000_DAC_SEL_DAP 0x3 +#define SGTL5000_I2S_OUT_SEL_MASK 0x0003 +#define SGTL5000_I2S_OUT_SEL_SHIFT 0 +#define SGTL5000_I2S_OUT_SEL_WIDTH 2 +#define SGTL5000_I2S_OUT_SEL_ADC 0x0 +#define SGTL5000_I2S_OUT_SEL_I2S_IN 0x1 +#define SGTL5000_I2S_OUT_SEL_DAP 0x3 + +/* + * SGTL5000_CHIP_ADCDAC_CTRL + */ +#define SGTL5000_VOL_BUSY_DAC_RIGHT 0x2000 +#define SGTL5000_VOL_BUSY_DAC_LEFT 0x1000 +#define SGTL5000_DAC_VOL_RAMP_EN 0x0200 +#define SGTL5000_DAC_VOL_RAMP_EXPO 0x0100 +#define SGTL5000_DAC_MUTE_RIGHT 0x0008 +#define SGTL5000_DAC_MUTE_LEFT 0x0004 +#define SGTL5000_ADC_HPF_FREEZE 0x0002 +#define SGTL5000_ADC_HPF_BYPASS 0x0001 + +/* + * SGTL5000_CHIP_DAC_VOL + */ +#define SGTL5000_DAC_VOL_RIGHT_MASK 0xff00 +#define SGTL5000_DAC_VOL_RIGHT_SHIFT 8 +#define SGTL5000_DAC_VOL_RIGHT_WIDTH 8 +#define SGTL5000_DAC_VOL_LEFT_MASK 0x00ff +#define SGTL5000_DAC_VOL_LEFT_SHIFT 0 +#define SGTL5000_DAC_VOL_LEFT_WIDTH 8 + +/* + * SGTL5000_CHIP_PAD_STRENGTH + */ +#define SGTL5000_PAD_I2S_LRCLK_MASK 0x0300 +#define SGTL5000_PAD_I2S_LRCLK_SHIFT 8 +#define SGTL5000_PAD_I2S_LRCLK_WIDTH 2 +#define SGTL5000_PAD_I2S_SCLK_MASK 0x00c0 +#define SGTL5000_PAD_I2S_SCLK_SHIFT 6 +#define SGTL5000_PAD_I2S_SCLK_WIDTH 2 +#define SGTL5000_PAD_I2S_DOUT_MASK 0x0030 +#define SGTL5000_PAD_I2S_DOUT_SHIFT 4 +#define SGTL5000_PAD_I2S_DOUT_WIDTH 2 +#define SGTL5000_PAD_I2C_SDA_MASK 0x000c +#define SGTL5000_PAD_I2C_SDA_SHIFT 2 +#define SGTL5000_PAD_I2C_SDA_WIDTH 2 +#define SGTL5000_PAD_I2C_SCL_MASK 0x0003 +#define SGTL5000_PAD_I2C_SCL_SHIFT 0 +#define SGTL5000_PAD_I2C_SCL_WIDTH 2 + +/* + * SGTL5000_CHIP_ANA_ADC_CTRL + */ +#define SGTL5000_ADC_VOL_M6DB 0x0100 +#define SGTL5000_ADC_VOL_RIGHT_MASK 0x00f0 +#define SGTL5000_ADC_VOL_RIGHT_SHIFT 4 +#define SGTL5000_ADC_VOL_RIGHT_WIDTH 4 +#define SGTL5000_ADC_VOL_LEFT_MASK 0x000f +#define SGTL5000_ADC_VOL_LEFT_SHIFT 0 +#define SGTL5000_ADC_VOL_LEFT_WIDTH 4 + +/* + * SGTL5000_CHIP_ANA_HP_CTRL + */ +#define SGTL5000_HP_VOL_RIGHT_MASK 0x7f00 +#define SGTL5000_HP_VOL_RIGHT_SHIFT 8 +#define SGTL5000_HP_VOL_RIGHT_WIDTH 7 +#define SGTL5000_HP_VOL_LEFT_MASK 0x007f +#define SGTL5000_HP_VOL_LEFT_SHIFT 0 +#define SGTL5000_HP_VOL_LEFT_WIDTH 7 + +/* + * SGTL5000_CHIP_ANA_CTRL + */ +#define SGTL5000_LINE_OUT_MUTE 0x0100 +#define SGTL5000_HP_SEL_MASK 0x0040 +#define SGTL5000_HP_SEL_SHIFT 6 +#define SGTL5000_HP_SEL_WIDTH 1 +#define SGTL5000_HP_SEL_DAC 0x0 +#define SGTL5000_HP_SEL_LINE_IN 0x1 +#define SGTL5000_HP_ZCD_EN 0x0020 +#define SGTL5000_HP_MUTE 0x0010 +#define SGTL5000_ADC_SEL_MASK 0x0004 +#define SGTL5000_ADC_SEL_SHIFT 2 +#define SGTL5000_ADC_SEL_WIDTH 1 +#define SGTL5000_ADC_SEL_MIC 0x0 +#define SGTL5000_ADC_SEL_LINE_IN 0x1 +#define SGTL5000_ADC_ZCD_EN 0x0002 +#define SGTL5000_ADC_MUTE 0x0001 + +/* + * SGTL5000_CHIP_LINREG_CTRL + */ +#define SGTL5000_VDDC_MAN_ASSN_MASK 0x0040 +#define SGTL5000_VDDC_MAN_ASSN_SHIFT 6 +#define SGTL5000_VDDC_MAN_ASSN_WIDTH 1 +#define SGTL5000_VDDC_MAN_ASSN_VDDA 0x0 +#define SGTL5000_VDDC_MAN_ASSN_VDDIO 0x1 +#define SGTL5000_VDDC_ASSN_OVRD 0x0020 +#define SGTL5000_LINREG_VDDD_MASK 0x000f +#define SGTL5000_LINREG_VDDD_SHIFT 0 +#define SGTL5000_LINREG_VDDD_WIDTH 4 + +/* + * SGTL5000_CHIP_REF_CTRL + */ +#define SGTL5000_ANA_GND_MASK 0x01f0 +#define SGTL5000_ANA_GND_SHIFT 4 +#define SGTL5000_ANA_GND_WIDTH 5 +#define SGTL5000_ANA_GND_BASE 800 /* mv */ +#define SGTL5000_ANA_GND_STP 25 /*mv */ +#define SGTL5000_BIAS_CTRL_MASK 0x000e +#define SGTL5000_BIAS_CTRL_SHIFT 1 +#define SGTL5000_BIAS_CTRL_WIDTH 3 +#define SGTL5000_SMALL_POP 0x0001 + +/* + * SGTL5000_CHIP_MIC_CTRL + */ +#define SGTL5000_BIAS_R_MASK 0x0200 +#define SGTL5000_BIAS_R_SHIFT 8 +#define SGTL5000_BIAS_R_WIDTH 2 +#define SGTL5000_BIAS_R_off 0x0 +#define SGTL5000_BIAS_R_2K 0x1 +#define SGTL5000_BIAS_R_4k 0x2 +#define SGTL5000_BIAS_R_8k 0x3 +#define SGTL5000_BIAS_VOLT_MASK 0x0070 +#define SGTL5000_BIAS_VOLT_SHIFT 4 +#define SGTL5000_BIAS_VOLT_WIDTH 3 +#define SGTL5000_MIC_GAIN_MASK 0x0003 +#define SGTL5000_MIC_GAIN_SHIFT 0 +#define SGTL5000_MIC_GAIN_WIDTH 2 + +/* + * SGTL5000_CHIP_LINE_OUT_CTRL + */ +#define SGTL5000_LINE_OUT_CURRENT_MASK 0x0f00 +#define SGTL5000_LINE_OUT_CURRENT_SHIFT 8 +#define SGTL5000_LINE_OUT_CURRENT_WIDTH 4 +#define SGTL5000_LINE_OUT_CURRENT_180u 0x0 +#define SGTL5000_LINE_OUT_CURRENT_270u 0x1 +#define SGTL5000_LINE_OUT_CURRENT_360u 0x3 +#define SGTL5000_LINE_OUT_CURRENT_450u 0x7 +#define SGTL5000_LINE_OUT_CURRENT_540u 0xf +#define SGTL5000_LINE_OUT_GND_MASK 0x003f +#define SGTL5000_LINE_OUT_GND_SHIFT 0 +#define SGTL5000_LINE_OUT_GND_WIDTH 6 +#define SGTL5000_LINE_OUT_GND_BASE 800 /* mv */ +#define SGTL5000_LINE_OUT_GND_STP 25 +#define SGTL5000_LINE_OUT_GND_MAX 0x23 + +/* + * SGTL5000_CHIP_LINE_OUT_VOL + */ +#define SGTL5000_LINE_OUT_VOL_RIGHT_MASK 0x1f00 +#define SGTL5000_LINE_OUT_VOL_RIGHT_SHIFT 8 +#define SGTL5000_LINE_OUT_VOL_RIGHT_WIDTH 5 +#define SGTL5000_LINE_OUT_VOL_LEFT_MASK 0x001f +#define SGTL5000_LINE_OUT_VOL_LEFT_SHIFT 0 +#define SGTL5000_LINE_OUT_VOL_LEFT_WIDTH 5 + +/* + * SGTL5000_CHIP_ANA_POWER + */ +#define SGTL5000_DAC_STEREO 0x4000 +#define SGTL5000_LINREG_SIMPLE_POWERUP 0x2000 +#define SGTL5000_STARTUP_POWERUP 0x1000 +#define SGTL5000_VDDC_CHRGPMP_POWERUP 0x0800 +#define SGTL5000_PLL_POWERUP 0x0400 +#define SGTL5000_LINEREG_D_POWERUP 0x0200 +#define SGTL5000_VCOAMP_POWERUP 0x0100 +#define SGTL5000_VAG_POWERUP 0x0080 +#define SGTL5000_ADC_STEREO 0x0040 +#define SGTL5000_REFTOP_POWERUP 0x0020 +#define SGTL5000_HP_POWERUP 0x0010 +#define SGTL5000_DAC_POWERUP 0x0008 +#define SGTL5000_CAPLESS_HP_POWERUP 0x0004 +#define SGTL5000_ADC_POWERUP 0x0002 +#define SGTL5000_LINE_OUT_POWERUP 0x0001 + +/* + * SGTL5000_CHIP_PLL_CTRL + */ +#define SGTL5000_PLL_INT_DIV_MASK 0xf800 +#define SGTL5000_PLL_INT_DIV_SHIFT 11 +#define SGTL5000_PLL_INT_DIV_WIDTH 5 +#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700 +#define SGTL5000_PLL_FRAC_DIV_SHIFT 0 +#define SGTL5000_PLL_FRAC_DIV_WIDTH 11 + +/* + * SGTL5000_CHIP_CLK_TOP_CTRL + */ +#define SGTL5000_INT_OSC_EN 0x0800 +#define SGTL5000_INPUT_FREQ_DIV2 0x0008 + +/* + * SGTL5000_CHIP_ANA_STATUS + */ +#define SGTL5000_HP_LRSHORT 0x0200 +#define SGTL5000_CAPLESS_SHORT 0x0100 +#define SGTL5000_PLL_LOCKED 0x0010 + +/* + * SGTL5000_CHIP_SHORT_CTRL + */ +#define SGTL5000_LVLADJR_MASK 0x7000 +#define SGTL5000_LVLADJR_SHIFT 12 +#define SGTL5000_LVLADJR_WIDTH 3 +#define SGTL5000_LVLADJL_MASK 0x0700 +#define SGTL5000_LVLADJL_SHIFT 8 +#define SGTL5000_LVLADJL_WIDTH 3 +#define SGTL5000_LVLADJC_MASK 0x0070 +#define SGTL5000_LVLADJC_SHIFT 4 +#define SGTL5000_LVLADJC_WIDTH 3 +#define SGTL5000_LR_SHORT_MOD_MASK 0x000c +#define SGTL5000_LR_SHORT_MOD_SHIFT 2 +#define SGTL5000_LR_SHORT_MOD_WIDTH 2 +#define SGTL5000_CM_SHORT_MOD_MASK 0x0003 +#define SGTL5000_CM_SHORT_MOD_SHIFT 0 +#define SGTL5000_CM_SHORT_MOD_WIDTH 2 + +/* + *SGTL5000_CHIP_ANA_TEST2 + */ +#define SGTL5000_MONO_DAC 0x1000 + +/* + * SGTL5000_DAP_CTRL + */ +#define SGTL5000_DAP_MIX_EN 0x0010 +#define SGTL5000_DAP_EN 0x0001 + +#define SGTL5000_SYSCLK 0x00 +#define SGTL5000_LRCLK 0x01 + +#endif -- 1.7.0.4

5 11

[alsa-devel] [PATCH] mfd: Remove bitrotted genirq comment from wm831x IRQ code
by Mark Brown 17 Feb '11

17 Feb '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm831x-irq.c | 9 --------- 1 files changed, 0 insertions(+), 9 deletions(-) diff --git a/drivers/mfd/wm831x-irq.c b/drivers/mfd/wm831x-irq.c index 03eb61d..2e45f60 100644 --- a/drivers/mfd/wm831x-irq.c +++ b/drivers/mfd/wm831x-irq.c @@ -26,15 +26,6 @@ #include <linux/delay.h> -/* - * Since generic IRQs don't currently support interrupt controllers on - * interrupt driven buses we don't use genirq but instead provide an - * interface that looks very much like the standard ones. This leads - * to some bodges, including storing interrupt handler information in - * the static irq_data table we use to look up the data for individual - * interrupts, but hopefully won't last too long. - */ - struct wm831x_irq_data { int primary; int reg; -- 1.7.2.3

1 1

[alsa-devel] [PATCH 2/9] davinci_voicecodec: use mfd_data instead of driver_data
by Andres Salomon 16 Feb '11

16 Feb '11

Use mfd_data for passing information from mfd drivers to soc clients. The mfd_cell's driver_data field is being phased out. Clients that were using driver_data now access .mfd_data via mfd_get_data(). Signed-off-by: Andres Salomon <dilinger(a)queued.net> --- drivers/mfd/davinci_voicecodec.c | 4 ++-- sound/soc/codecs/cq93vc.c | 3 ++- sound/soc/davinci/davinci-vcif.c | 2 +- 3 files changed, 5 insertions(+), 4 deletions(-) diff --git a/drivers/mfd/davinci_voicecodec.c b/drivers/mfd/davinci_voicecodec.c index 33c923d..f946c0b 100644 --- a/drivers/mfd/davinci_voicecodec.c +++ b/drivers/mfd/davinci_voicecodec.c @@ -119,12 +119,12 @@ static int __init davinci_vc_probe(struct platform_device *pdev) /* Voice codec interface client */ cell = &davinci_vc->cells[DAVINCI_VC_VCIF_CELL]; cell->name = "davinci_vcif"; - cell->driver_data = davinci_vc; + cell->mfd_data = davinci_vc; /* Voice codec CQ93VC client */ cell = &davinci_vc->cells[DAVINCI_VC_CQ93VC_CELL]; cell->name = "cq93vc"; - cell->driver_data = davinci_vc; + cell->mfd_data = davinci_vc; ret = mfd_add_devices(&pdev->dev, pdev->id, davinci_vc->cells, DAVINCI_VC_CELLS, NULL, 0); diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 46dbfd0..b8066ef 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -153,7 +153,8 @@ static int cq93vc_resume(struct snd_soc_codec *codec) static int cq93vc_probe(struct snd_soc_codec *codec) { - struct davinci_vc *davinci_vc = codec->dev->platform_data; + struct davinci_vc *davinci_vc = + mfd_get_data(to_platform_device(codec->dev)); davinci_vc->cq93vc.codec = codec; codec->control_data = davinci_vc; diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9d2afcc..13e05a3 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -205,7 +205,7 @@ static struct snd_soc_dai_driver davinci_vcif_dai = { static int davinci_vcif_probe(struct platform_device *pdev) { - struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct davinci_vc *davinci_vc = mfd_get_data(pdev); struct davinci_vcif_dev *davinci_vcif_dev; int ret; -- 1.7.2.3

2 1

[alsa-devel] [PATCH 3/9] wm8400-codec: use mfd_data instead of driver_data
by Andres Salomon 16 Feb '11

16 Feb '11

Use mfd_data for passing information from mfd drivers to soc clients. The mfd_cell's driver_data field is being phased out. Clients that were using driver_data now access .mfd_data via mfd_get_data(). Signed-off-by: Andres Salomon <dilinger(a)queued.net> --- drivers/mfd/wm8400-core.c | 2 +- sound/soc/codecs/wm8400.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/mfd/wm8400-core.c b/drivers/mfd/wm8400-core.c index 1bfef48..3a6e78c 100644 --- a/drivers/mfd/wm8400-core.c +++ b/drivers/mfd/wm8400-core.c @@ -245,7 +245,7 @@ static int wm8400_register_codec(struct wm8400 *wm8400) { struct mfd_cell cell = { .name = "wm8400-codec", - .driver_data = wm8400, + .mfd_data = wm8400, }; return mfd_add_devices(wm8400->dev, -1, &cell, 1, NULL, 0); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 3c3bc07..ee105c2 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1377,7 +1377,7 @@ static void wm8400_probe_deferred(struct work_struct *work) static int wm8400_codec_probe(struct snd_soc_codec *codec) { - struct wm8400 *wm8400 = dev_get_platdata(codec->dev); + struct wm8400 *wm8400 = mfd_get_data(to_platform_device(codec->dev)); struct wm8400_priv *priv; int ret; u16 reg; -- 1.7.2.3

2 1

[PATCH] ALSA: HDA: Do not announce false surround in Conexant auto
by David Henningsson 16 Feb '11

16 Feb '11

Without this patch, one line-out and one speaker and Conexant's auto parser would announce (non-working) surround capabilities. BugLink: http://bugs.launchpad.net/bugs/721126 Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index fbe97d3..029741f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3410,7 +3410,7 @@ static void cx_auto_parse_output(struct hda_codec *codec) } } spec->multiout.dac_nids = spec->private_dac_nids; - spec->multiout.max_channels = nums * 2; + spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (cfg->hp_outs > 0) spec->auto_mute = 1; -- 1.7.1 --------------060309080607070205030803--

1 0

[alsa-devel] ASoC HDMI codec location
by Stephen Warren 16 Feb '11

16 Feb '11

Mark, I'm looking at enabling audio-over-HDMI on Tegra in the ChromeOS kernel. This will use the SPDIF controller as the CPU DAI. Right now, I have things working with the stub SPDIF codec (codecs/spdif_transciever.c), but to support a non-hard-coded sample rate, need a real codec driver that writes registers in the HDMI controller. I see that a Samsung chip has such a codec in drivers/video/sh_mobile_hdmi.c. The equivalent for Tegra would be drivers/video/tegra/dc/hdmi.c. Note that this location doesn't yet exist upstream. I just wanted to confirm that this location makes sense to you, vs. implementing a codec in sound/soc/tegra/hdmi.c, and having that call functions exposed by the Tegra dc/hdmi.c driver? Note that the SPDIF controller isn't involved in the Tegra Digital Audio Switch muxing/switching module at all. Thanks. -- nvpublic

2 3

[alsa-devel] alsa and terratec aureon 7.1 USB sound card
by Vincent Meserette 16 Feb '11

16 Feb '11

Hi all, I just wanted to know if anyone of you already used the terratec aureon 7.1 USB sound card with linux. Is it well recognized by alsa ? Thanks in advance for your reply. Kind regards Vincent

2 1

[alsa-devel] Codec driver for TLV320AIC3204.
by javier Martin 16 Feb '11

16 Feb '11

Hi, is anyone working on a driver for tlv320aic3204 codec? -- Javier Martin Vista Silicon S.L. CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

1 0

[alsa-devel] [PATCH v19 0/3] TI Wl1273 FM radio driver.
by Matti J. Aaltonen 16 Feb '11

16 Feb '11

Hello. Now I've refactored the code so that the I2C I/O functions are in the MFD core. Also now the codec can be compiled without compiling the V4L2 driver. I haven't implemented the audio routing (yet), but I've added a TODO comment about it in the codec file. Cheers, Matti Matti J. Aaltonen (3): MFD: WL1273 FM Radio: MFD driver for the FM radio. V4L2: WL1273 FM Radio: TI WL1273 FM radio driver ASoC: WL1273 FM radio: Access I2C IO functions through pointers. drivers/media/radio/Kconfig | 16 + drivers/media/radio/Makefile | 1 + drivers/media/radio/radio-wl1273.c | 2183 ++++++++++++++++++++++++++++++++++++ drivers/mfd/Kconfig | 10 + drivers/mfd/Makefile | 1 + drivers/mfd/wl1273-core.c | 295 +++++ include/linux/mfd/wl1273-core.h | 291 +++++ sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wl1273.c | 38 +- sound/soc/codecs/wl1273.h | 71 -- 10 files changed, 2817 insertions(+), 91 deletions(-) create mode 100644 drivers/media/radio/radio-wl1273.c create mode 100644 drivers/mfd/wl1273-core.c create mode 100644 include/linux/mfd/wl1273-core.h

3 5

[alsa-devel] [PATCH 0/5] mfld machine and codec fixes
by Koul, Vinod 16 Feb '11

16 Feb '11

From: Vinod Koul <vinod.koul(a)intel.com> Hi, This 5 patch series introduces some minor fixes in the TI sn95031 codec driver and the mfld_machine driver and sst_platfrom driver. This also changes mfld_machine to use the soc_register_card() API. Vinod Koul (5): ASoC: sn95031: make playback rails depend on actual pins they control ASoC: sn95031: fix the DMIC path routing ASoC: sn95031: fix the amic tlv scale ASoC: sst_platform: fix the pulseaudio error ASoC: mfld_machine: make use of soc_register_card API sound/soc/codecs/sn95031.c | 58 +++++++++++++++++++------------------- sound/soc/mid-x86/mfld_machine.c | 20 ++++--------- sound/soc/mid-x86/sst_platform.c | 9 ++++++ 3 files changed, 44 insertions(+), 43 deletions(-) -- 1.7.2.3

4 9

[alsa-devel] [PATCH 1/3] ASoC: omap: rx51: Use gpio_request_one to configure tvout_sel gpio
by Jarkko Nikula 15 Feb '11

15 Feb '11

Just slight cleanup to be sync with upcoming change. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/omap/rx51.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 2d7d411..9411969 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -394,10 +394,10 @@ static int __init rx51_soc_init(void) if (!machine_is_nokia_rx51()) return -ENODEV; - err = gpio_request(RX51_TVOUT_SEL_GPIO, "tvout_sel"); + err = gpio_request_one(RX51_TVOUT_SEL_GPIO, + GPIOF_DIR_OUT | GPIOF_INIT_LOW, "tvout_sel"); if (err) goto err_gpio_tvout_sel; - gpio_direction_output(RX51_TVOUT_SEL_GPIO, 0); rx51_snd_device = platform_device_alloc("soc-audio", -1); if (!rx51_snd_device) { -- 1.7.0.4

3 4

[PATCH] ALSA: HDA: Conexant auto: Handle multiple connections to ADC node
by David Henningsson 15 Feb '11

15 Feb '11

Conexant 20641 has several inputs to its ADC node, with one selector and individual amps for all inputs. This patch adds support in the Conexant auto parser to handle that case. It also means that the pin node's volume is being renamed to "Boost" to avoid name clash with the new volume controls on the ADC node. BugLink: http://bugs.launchpad.net/bugs/719524 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 61 +++++++++++++++++++++++++++++++-------- 1 files changed, 48 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index fbe97d3..cd29eaf 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3729,9 +3729,9 @@ static int cx_auto_init(struct hda_codec *codec) return 0; } -static int cx_auto_add_volume(struct hda_codec *codec, const char *basename, +static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, const char *dir, int cidx, - hda_nid_t nid, int hda_dir) + hda_nid_t nid, int hda_dir, int amp_idx) { static char name[32]; static struct snd_kcontrol_new knew[] = { @@ -3743,7 +3743,8 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename, for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; - knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, hda_dir); + knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx, + hda_dir); knew[i].subdevice = HDA_SUBDEV_AMP_FLAG; knew[i].index = cidx; snprintf(name, sizeof(name), "%s%s %s", basename, dir, sfx[i]); @@ -3759,6 +3760,9 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename, return 0; } +#define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \ + cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0) + #define cx_auto_add_pb_volume(codec, nid, str, idx) \ cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) @@ -3808,29 +3812,60 @@ static int cx_auto_build_input_controls(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; static const char *prev_label; - int i, err, cidx; + int i, err, cidx, conn_len; + hda_nid_t conn[HDA_MAX_CONNECTIONS]; + + int multi_adc_volume = 0; /* If the ADC nid has several input volumes */ + int adc_nid = spec->adc_nids[0]; + + conn_len = snd_hda_get_connections(codec, adc_nid, conn, + HDA_MAX_CONNECTIONS); + if (conn_len < 0) + return conn_len; + + multi_adc_volume = cfg->num_inputs > 1 && conn_len > 1; + if (!multi_adc_volume) { + err = cx_auto_add_volume(codec, "Capture", "", 0, adc_nid, + HDA_INPUT); + if (err < 0) + return err; + } - err = cx_auto_add_volume(codec, "Capture", "", 0, spec->adc_nids[0], - HDA_INPUT); - if (err < 0) - return err; prev_label = NULL; cidx = 0; for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; const char *label; - if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) + int j; + int pin_amp = get_wcaps(codec, nid) & AC_WCAP_IN_AMP; + if (!pin_amp && !multi_adc_volume) continue; + label = hda_get_autocfg_input_label(codec, cfg, i); if (label == prev_label) cidx++; else cidx = 0; prev_label = label; - err = cx_auto_add_volume(codec, label, " Capture", cidx, - nid, HDA_INPUT); - if (err < 0) - return err; + + if (pin_amp) { + err = cx_auto_add_volume(codec, label, " Boost", cidx, + nid, HDA_INPUT); + if (err < 0) + return err; + } + + if (!multi_adc_volume) + continue; + for (j = 0; j < conn_len; j++) { + if (conn[j] == nid) { + err = cx_auto_add_volume_idx(codec, label, + " Capture", cidx, adc_nid, HDA_INPUT, j); + if (err < 0) + return err; + break; + } + } } return 0; } -- 1.7.1 --------------070007020807020600040301 Content-Type: text/plain; name="conexant_20641.txt" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename="conexant_20641.txt" Codec: Conexant CX20641 Address: 2 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x14f150a1 Subsystem Id: 0x102804f5 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=5, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CONEXANT Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x49 0x49] Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x46 0x46] Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0f, mute=0 Amp-Out vals: [0x00] Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="CONEXANT Analog", type="Audio", device=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x50 0x50] [0x45 0x45] [0x00 0x00] [0x4a 0x4a] Converter: stream=1, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a 0x1b 0x1d* 0x1e Node 0x15 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a* 0x1b 0x1d 0x1e Node 0x16 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x22 Node 0x17 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x00 0x00] Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x18 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00010014: OUT EAPD Detect EAPD 0x0: Pin Default 0x04f001f0: [Jack] Other at Ext Right Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 3 0x10* 0x11 0x17 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x02214040: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=37, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 3 0x10 0x11* 0x17 Node 0x1a [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Front Mic Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: [0x04 0x04] Pincap 0x00001324: IN Detect Vref caps: HIZ 50 80 Pin Default 0x02a19020: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1b [Pin Complex] wcaps 0x40058b: Stereo Amp-In Control: name="Rear Mic Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: [0x04 0x04] Pincap 0x00011334: IN OUT EAPD Detect Vref caps: HIZ 50 80 EAPD 0x0: Pin Default 0x01a1903e: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0xe Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 3 0x10* 0x11 0x17 Node 0x1c [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 3 0x10* 0x11 0x17 Node 0x1d [Pin Complex] wcaps 0x40058b: Stereo Amp-In Control: name="Line Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: [0x04 0x04] Pincap 0x00010034: IN OUT EAPD Detect EAPD 0x0: Pin Default 0x01813030: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 3 0x10* 0x11 0x17 Node 0x1e [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000024: IN Detect Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1f [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00000010: OUT Pin Default 0x9217011f: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0xf Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 3 0x10 0x11 0x17* Node 0x20 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00010010: OUT EAPD EAPD 0x0: Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 3 0x10* 0x11 0x17 Node 0x22 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 3 0x10 0x11 0x17 Node 0x23 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x24 [Volume Knob Widget] wcaps 0x600080: Mono Volume-Knob: delta=1, steps=74, direct=0, val=60 Unsolicited: tag=00, enabled=0 Connection: 3 0x10 0x11 0x17 Node 0x25 [Audio Input] wcaps 0x100711: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x26 [Pin Complex] wcaps 0x400681: Stereo Digital Pincap 0x00000024: IN Detect Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 --------------070007020807020600040301--

1 0

[alsa-devel] plughw, access modes
by Nenad Sljivic 15 Feb '11

15 Feb '11

Greetings guys, I am dealing with the device driver for the chip that has his own hardware buffer and the DMA is not available. Therefore, driver doesn't support MMAP access and the data is copied using the copy file operation. My HW natively supports BE format. However, I would like to be able to play little endian pcm as well. The problem, occurs when I try to use plughw device in order to play LE pcm file " aplay: set_params:1022: Unable to install hw params: " My debugging session proved me that plughw device is trying to access device driver with MMAP_INTERLEAVED mode. Does it mean that this is the only way for plughw to access device driver, and that I am not able to use plugins unless do some heavy modifications of device drivers (implement mmap access). Am I missing something? Any info on this would be most welcome. Thanks, Nenad

2 2

[alsa-devel] [PATCH] New quirk for USB DAC 08bb:2704
by Omari Stephens 15 Feb '11

15 Feb '11

Howdy, all I have one of these: http://www.chordelectronics.co.uk/products_detail.asp?id=63 It's got one of these USB DACs inside ID 08bb:2704 Texas Instruments Japan The DAC has a single mixer control, Master, which allows you to adjust volume and mute/unmute. The fun thing is that the volume adjustment has no effect whatsoever. On machines running pulse, this means that output will be at 0dB pretty much all the time. (Of course, this is true of non-pulse machines also). The attached patch disables the Master control, which causes pulse to control levels entirely in software, as it should for this device. The main downside to this patch is that, while the volume part of the Master control is non-functional, the mute/unmute aspect does work as expected. I couldn't figure out how to change the Master control to just a mute/unmute control, but if someone has suggestions on what to do, that'd be appreciated. Judging by the comments in the code (and contrary to what I wrote in the map comment), I presume the mute/unmute control is the only reason why 08bb:2702 isn't doing this same thing. Cheers, --xsdg

2 3

[alsa-devel] [PATCH] And it's another happy day in the land of quirks
by David Henningsson 15 Feb '11

15 Feb '11

Guess what - it's another quirk! :-) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[PATCH] ALSA: HDA: Add position_fix quirk for an Asus device
by David Henningsson 14 Feb '11

14 Feb '11

The bug reporter claims that position_fix=1 is needed for his microphone to work. The controller PCI vendor-id is [1002:4383] (rev 40). Reported-by: Kjell L. BugLink: http://bugs.launchpad.net/bugs/718402 Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0baffcd..fcedad9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2308,6 +2308,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB), -- 1.7.1 --------------070706030207060809010808--

1 0

[alsa-devel] [PATCH] ALSA : au88x0 - modify pointer callback of au88x0 to give accurate playback position
by Raymond Yau 14 Feb '11

14 Feb '11

Modify pointer callback of au88x0 to give accurate playback position Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com>

2 1

[alsa-devel] Patches for new audio devices from Native Instruments
by Daniel Mack 14 Feb '11

14 Feb '11

Hi, Native Instruments announced a new set of audio hardware recently, and I have some patches ready to support them. See here for the details on their website: http://www.native-instruments.com/en/products/dj/traktor/ As I'm currently travelling and the only elaborated piece of hardware I got left is my mobile phone, I have to send them along as attachments. Many thanks to Mark Hills who rebased the patches for me so they should apply without trouble. In case of any doubts or questions - I'm reading my mails occasionally :-) Best regards, Daniel

2 1

[alsa-devel] [PATCH] ALSA HDA - simplify multistreaming playback model of ad1988
by Raymond Yau 14 Feb '11

14 Feb '11

- remove unnecessary ad198x_alt_playback_pcm_prepare() and ad198x_alt_playback_pcm_cleanup() - remove unnecessary controls from ad1988_6stack_fp_mixers - remove unnecessary init verbs from ad1988_6stack_fp_init_verbs

2 1

[alsa-devel] [PATCH v3 0/4] Adding OMAP DMIC driver to kernel
by David Lambert 14 Feb '11

14 Feb '11

This is a patch series to add the OMAP Digital Microphone driver for OMAP4. It includes the driver, a generic DMIC codec, platform devices, as well as HWMOD entries for OMAP44xx chipsets. Note: The DMIC codec driver was removed from this series as it was accepted for and merged for the 2.6.38 kernel. Benoit Cousson (1): OMAP4: hwmod: add entries for DMIC driver David Lambert (3): ASoC: DMIC: Adding the OMAP DMIC driver OMAP4: DMIC: initializes the DMIC driver OMAP4: DMIC: Add DMIC codec platform devices arch/arm/mach-omap2/board-4430sdp.c | 11 + arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 91 +++++ arch/arm/plat-omap/devices.c | 35 ++ arch/arm/plat-omap/include/plat/dmic.h | 82 +++++ sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap-dmic.c | 534 ++++++++++++++++++++++++++++ 7 files changed, 758 insertions(+), 0 deletions(-) create mode 100644 arch/arm/plat-omap/include/plat/dmic.h create mode 100644 sound/soc/omap/omap-dmic.c

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[alsa-devel] Support for Acer Aspire One 522 HDA-Intel - HDA ATI SB (Conexant CX20584, Vendor Id: 0x14f15068, Subsystem Id: 0x10250543, Revision Id: 0x100302)
by Andreas Reichel 14 Feb '11

14 Feb '11

Dear All, I can not make my sound card working correctly. When I use hda-intel-generic I get sound from external and internal speaker, but only at the same time. External micro works well, internal micro does not. Could not test HDMI yet, but the mixer lacks all controls so far. When I compile in conexant-codec, then I get more controls, but can not make external output working. External micro works, internal micro does not. I have 4 micros as choice, only one is functional. The module for ati-hdmi is never loaded. When I use Ubuntu 10.10 instead with kernel-2.3.35 and alsa-drivers-1.0.23 I get internal and external output correctly. Still internal microphone does not work. The module for ati-hdmi is loaded, conexant-codec is not loaded but hda-intel-generic only. So far this combination works best related to sound, but i can not go back, because the graphic is not supported yet. Graphic support starts with 2.6.38-rc4. I tried already all possible models for the module loading, nothing helped. Attached please find a lot of technical output. Please tell me, in case you need more information. Please may I ask you for any help and support to setup the device correctly. It is such a nice laptop and it's a pity I can not benefit from sound. Thank you very much in advance! Andreas -- Laptop: Acer Aspire One 522 (AMD C50, Ati HD6250) are@netbook ~/Downloads/alsa-driver-1.0.24 $ lsmod Module Size Used by snd_hda_codec_conexant 36956 1 snd_hda_codec_hdmi 20861 1 snd_hda_intel 21268 0 snd_hda_codec 81683 3 snd_hda_codec_conexant,snd_hda_codec_hdmi,snd_hda_intel snd_hwdep 4738 1 snd_hda_codec snd_pcm 69772 3 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_timer 17643 1 snd_pcm snd 47681 7 snd_hda_codec_conexant,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer snd_page_alloc 6704 2 snd_hda_intel,snd_pcm are@netbook ~/Downloads/alsa-driver-1.0.24 $ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.24. Compiled on Feb 11 2011 for kernel 2.6.38-rc4 (SMP). are@netbook ~/Downloads/alsa-driver-1.0.24 $ cat /proc/asound/cards 0 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0x90544000 irq 41 1 [SB ]: HDA-Intel - HDA ATI SB HDA ATI SB at 0x90540000 irq 16 are@netbook ~/Downloads/alsa-driver-1.0.24 $ cat /proc/asound/hwdep 00-00: HDA Codec 0 01-00: HDA Codec 0 are@netbook ~/Downloads/alsa-driver-1.0.24 $ cat /proc/asound/modules 0 snd_hda_intel 1 snd_hda_intel are@netbook ~/Downloads/alsa-driver-1.0.24 $ cat /proc/asound/card0/codec#0 Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100200 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x201: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Device: name="HDMI 0", type="HDMI", device=3 Converter: stream=0, channel=0 Digital: Digital category: 0x0 Node 0x03 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000094: OUT Detect HDMI Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=03, enabled=1 Connection: 1 0x02 are@netbook ~/Downloads/alsa-driver-1.0.24 $ cat /proc/asound/card1/codec#0 Codec: Conexant CX20584 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x14f15068 Subsystem Id: 0x10250543 Revision Id: 0x100302 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Master Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CONEXANT Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x36 0x36] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x3c 0x3c] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0f, mute=0 Amp-Out vals: [0x00] Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Device: name="CONEXANT Analog", type="Audio", device=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x50 0x50] [0x80 0x80] [0x50 0x50] [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x15 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x16 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x17 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x03 0x03] Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Connection: 4 0x1a 0x1b* 0x1d 0x1e Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Connection: 4 0x1a* 0x1b 0x1d 0x1e Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x03211040: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1a [Pin Complex] wcaps 0x400481: Stereo Pincap 0x00001324: IN Detect Vref caps: HIZ 50 80 Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00011334: IN OUT EAPD Detect Vref caps: HIZ 50 80 EAPD 0x2: EAPD Pin Default 0x03a19030: [Jack] Mic at Ext Left Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1c [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1d [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00010034: IN OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1e [Pin Complex] wcaps 0x400481: Stereo Pincap 0x00000024: IN Detect Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Node 0x1f [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00000010: OUT Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x20 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Node 0x22 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Connection: 1 0x21 Node 0x23 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x90a60150: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x5, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Node 0x24 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Power states: D0 D1 D2 D3 D3cold Power: setting=D0, actual=D0 Connection: 2 0x10 0x11 Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono are@netbook ~ $ sudo arecord -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=SB HDA ATI SB, CONEXANT Analog Default Audio Device front:CARD=SB,DEV=0 HDA ATI SB, CONEXANT Analog Front speakers surround40:CARD=SB,DEV=0 HDA ATI SB, CONEXANT Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=SB,DEV=0 HDA ATI SB, CONEXANT Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=SB,DEV=0 HDA ATI SB, CONEXANT Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=SB,DEV=0 HDA ATI SB, CONEXANT Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=SB,DEV=0 HDA ATI SB, CONEXANT Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers are@netbook ~ $ sudo lspci 00:00.0 Host bridge: Advanced Micro Devices [AMD] Device 1510 00:01.0 VGA compatible controller: ATI Technologies Inc Device 9804 00:01.1 Audio device: ATI Technologies Inc Device 1314 00:11.0 SATA controller: ATI Technologies Inc SB700/SB800 SATA Controller [AHCI mode] 00:12.0 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI0 Controller 00:12.2 USB Controller: ATI Technologies Inc SB700/SB800 USB EHCI Controller 00:13.0 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI0 Controller 00:13.2 USB Controller: ATI Technologies Inc SB700/SB800 USB EHCI Controller 00:14.0 SMBus: ATI Technologies Inc SBx00 SMBus Controller (rev 42) 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) (rev 40) 00:14.3 ISA bridge: ATI Technologies Inc SB700/SB800 LPC host controller (rev 40) 00:14.4 PCI bridge: ATI Technologies Inc SBx00 PCI to PCI Bridge (rev 40) 00:15.0 PCI bridge: ATI Technologies Inc Device 43a0 00:15.2 PCI bridge: ATI Technologies Inc Device 43a2 00:15.3 PCI bridge: ATI Technologies Inc Device 43a3 00:18.0 Host bridge: Advanced Micro Devices [AMD] Device 1700 (rev 43) 00:18.1 Host bridge: Advanced Micro Devices [AMD] Device 1701 00:18.2 Host bridge: Advanced Micro Devices [AMD] Device 1702 00:18.3 Host bridge: Advanced Micro Devices [AMD] Device 1703 00:18.4 Host bridge: Advanced Micro Devices [AMD] Device 1704 00:18.5 Host bridge: Advanced Micro Devices [AMD] Device 1718 00:18.6 Host bridge: Advanced Micro Devices [AMD] Device 1716 00:18.7 Host bridge: Advanced Micro Devices [AMD] Device 1719 06:00.0 Ethernet controller: Attansic Technology Corp. Device 2062 (rev c1) 07:00.0 Network controller: Atheros Communications Inc. AR9285 Wireless Network Adapter (PCI-Express) (rev 01) 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) (rev 40) Subsystem: Acer Incorporated [ALI] Device 0543 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin ? routed to IRQ 16 Region 0: Memory at 90540000 (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 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel

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[alsa-devel] Unable to attach mixer to device AndroidOut: No such file or directory
by kuldeep ghan 14 Feb '11

14 Feb '11

I am trying to get audio working with android kernel. I am able to play wav files with alsa_aplay but when I try to open Output or Input stream using libadio Its failing. Can anyone help me with this? what could be wrong?

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Re: [alsa-devel] [PATCH v8 07/12] media: Entities, pads and links enumeration
by Laurent Pinchart 14 Feb '11

14 Feb '11

Hi Sylwester, On Sunday 13 February 2011 22:59:44 Sylwester Nawrocki wrote: > On 2011-01-27 13:30, Laurent Pinchart wrote: > > Create the following two ioctls and implement them at the media device > > level to enumerate entities, pads and links. > > > > - MEDIA_IOC_ENUM_ENTITIES: Enumerate entities and their properties > > - MEDIA_IOC_ENUM_LINKS: Enumerate all pads and links for a given entity > > > > Entity IDs can be non-contiguous. Userspace applications should > > enumerate entities using the MEDIA_ENT_ID_FLAG_NEXT flag. When the flag > > is set in the entity ID, the MEDIA_IOC_ENUM_ENTITIES will return the > > next entity with an ID bigger than the requested one. > > > > Only forward links that originate at one of the entity's source pads are > > returned during the enumeration process. > > > > Signed-off-by: Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> > > Signed-off-by: Sakari Ailus <sakari.ailus(a)maxwell.research.nokia.com> > > --- > > > > Documentation/DocBook/media-entities.tmpl | 8 + > > Documentation/DocBook/v4l/media-controller.xml | 2 + > > .../DocBook/v4l/media-ioc-device-info.xml | 3 +- > > .../DocBook/v4l/media-ioc-enum-entities.xml | 308 > > ++++++++++++++++++++ Documentation/DocBook/v4l/media-ioc-enum-links.xml > > | 202 +++++++++++++ drivers/media/media-device.c > > | 123 ++++++++ include/linux/media.h | > > 85 ++++++ include/media/media-entity.h | 24 +-- > > 8 files changed, 731 insertions(+), 24 deletions(-) > > create mode 100644 Documentation/DocBook/v4l/media-ioc-enum-entities.xml > > create mode 100644 Documentation/DocBook/v4l/media-ioc-enum-links.xml > > ... > > > diff --git a/Documentation/DocBook/v4l/media-ioc-enum-links.xml > > b/Documentation/DocBook/v4l/media-ioc-enum-links.xml new file mode > > 100644 > > index 0000000..daf0360 > > --- /dev/null > > +++ b/Documentation/DocBook/v4l/media-ioc-enum-links.xml > > @@ -0,0 +1,202 @@ > > +<refentry id="media-ioc-enum-links"> > > + <refmeta> > > + <refentrytitle>ioctl MEDIA_IOC_ENUM_LINKS</refentrytitle> > > + &manvol; > > + </refmeta> > > + > > + <refnamediv> > > + <refname>MEDIA_IOC_ENUM_LINKS</refname> > > + <refpurpose>Enumerate all pads and links for a given > > entity</refpurpose> + </refnamediv> > > + > > + <refsynopsisdiv> > > + <funcsynopsis> > > + <funcprototype> > > + <funcdef>int <function>ioctl</function></funcdef> > > + <paramdef>int <parameter>fd</parameter></paramdef> > > + <paramdef>int <parameter>request</parameter></paramdef> > > + <paramdef>struct media_links_enum > > *<parameter>argp</parameter></paramdef> + </funcprototype> > > + </funcsynopsis> > > + </refsynopsisdiv> > > + > > + <refsect1> > > + <title>Arguments</title> > > + > > + <variablelist> > > + <varlistentry> > > + <term><parameter>fd</parameter></term> > > + <listitem> > > + <para>File descriptor returned by > > + <link > > linkend='media-func-open'><function>open()</function></link>.</para> > > + </listitem> > > + </varlistentry> > > + <varlistentry> > > + <term><parameter>request</parameter></term> > > + <listitem> > > + <para>MEDIA_IOC_ENUM_LINKS</para> > > + </listitem> > > + </varlistentry> > > + <varlistentry> > > + <term><parameter>argp</parameter></term> > > + <listitem> > > + <para></para> > > + </listitem> > > + </varlistentry> > > + </variablelist> > > + </refsect1> > > + > > + <refsect1> > > + <title>Description</title> > > + > > + <para>To enumerate pads and/or links for a given entity, > > applications set + the entity field of a &media-links-enum; structure > > and initialize the + &media-pad-desc; and &media-link-desc; structure > > arrays pointed by the + <structfield>pads</structfield> and > > <structfield>links</structfield> fields. + They then call the > > MEDIA_IOC_ENUM_LINKS ioctl with a pointer to this + structure.</para> > > + <para>If the <structfield>pads</structfield> field is not NULL, the > > driver + fills the <structfield>pads</structfield> array with > > information about the + entity's pads. The array must have enough > > room to store all the entity's + pads. The number of pads can be > > retrieved with the &MEDIA-IOC-ENUM-ENTITIES; + ioctl.</para> > > + <para>If the <structfield>links</structfield> field is not NULL, the > > driver + fills the <structfield>links</structfield> array with > > information about the + entity's outbound links. The array must have > > enough room to store all the + entity's outbound links. The number of > > outbound links can be retrieved with + the &MEDIA-IOC-ENUM-ENTITIES; > > ioctl.</para> > > + <para>Only forward links that originate at one of the entity's > > source pads + are returned during the enumeration process.</para> > > + > > + <table pgwide="1" frame="none" id="media-links-enum"> > > + <title>struct <structname>media_links_enum</structname></title> > > + <tgroup cols="3"> > > + &cs-str; > > + <tbody valign="top"> > > + <row> > > + <entry>__u32</entry> > > + <entry><structfield>entity</structfield></entry> > > + <entry>Entity id, set by the application.</entry> > > + </row> > > + <row> > > + <entry>struct &media-pad-desc;</entry> > > + <entry>*<structfield>pads</structfield></entry> > > + <entry>Pointer to a pads array allocated by the application. > > Ignored + if NULL.</entry> > > + </row> > > + <row> > > + <entry>struct &media-link-desc;</entry> > > + <entry>*<structfield>links</structfield></entry> > > + <entry>Pointer to a links array allocated by the application. > > Ignored + if NULL.</entry> > > + </row> > > + </tbody> > > + </tgroup> > > + </table> > > + > > + <table pgwide="1" frame="none" id="media-pad-desc"> > > + <title>struct <structname>media_pad_desc</structname></title> > > + <tgroup cols="3"> > > + &cs-str; > > + <tbody valign="top"> > > + <row> > > + <entry>__u32</entry> > > + <entry><structfield>entity</structfield></entry> > > + <entry>ID of the entity this pad belongs to.</entry> > > + </row> > > + <row> > > + <entry>__u16</entry> > > + <entry><structfield>index</structfield></entry> > > + <entry>0-based pad index.</entry> > > + </row> > > + <row> > > + <entry>__u32</entry> > > + <entry><structfield>flags</structfield></entry> > > + <entry>Pad flags, see <xref linkend="media-pad-flag" /> for more > > details.</entry> + </row> > > + </tbody> > > + </tgroup> > > + </table> > > + > > + <table frame="none" pgwide="1" id="media-pad-flag"> > > + <title>Media pad flags</title> > > + <tgroup cols="2"> > > + <colspec colname="c1"/> > > + <colspec colname="c2"/> > > + <tbody valign="top"> > > + <row> > > + <entry><constant>MEDIA_PAD_FL_INPUT</constant></entry> > > + <entry>Input pad, relative to the entity. Input pads sink data and > > + are targets of links.</entry> > > + </row> > > + <row> > > + <entry><constant>MEDIA_PAD_FL_OUTPUT</constant></entry> > > + <entry>Output pad, relative to the entity. Output pads source data > > + and are origins of links.</entry> > > + </row> > > + </tbody> > > + </tgroup> > > + </table> > > + > > + <table pgwide="1" frame="none" id="media-link-desc"> > > + <title>struct <structname>media_links_enum</structname></title> > > ..the above line should perhaps be: > > + <title>struct <structname>media_links_desc</structname></title> > > as "struct media_links_desc" is described above. Thanks for catching this. I'll fix it. > BTW, I was wondering how one makes use of the pads associated with > a media entity of type other than MEDIA_ENT_T_V4L2_SUBDEV*, > e.g. V4L2 device - MEDIA_ENT_T_DEVNODE_V4L. > > I guess new standard ioctls should be added for per pads operations > at V4L2 device, similarly is it is done with v4l2-subdevs. > Is this right? I don't think we need now ioctls there. As DEVNODE_V4L entities are just an interface between the driver and userspace, applications should use V4L2 ioctls on the video device node. For non-V4L entities other ioctls will obviously be needed, but they will be defined by other subsystems such as ALSA (provided the entities have a device node). -- Regards, Laurent Pinchart

1 0

[alsa-devel] [PATCH - snd-usb-audio 1/1] Support for Power/Status LED on Creative USB X-Fi S51
by emailmandar＠gmail.com 13 Feb '11

13 Feb '11

From: Mandar Joshi <emailmandar(a)gmail.com> This patch adds support for Power/Status LED on Creative USB X-Fi S51. There is just one LED on the device. The LED can either be On or it can be set to Blink. There doesn't seem to be a way to switch it off. The control message to change LED status is similar to that of audigy2nx except that the index is to be set to 0 and value is 1 for Blink and 0 for On. The 'Power LED' control in alsamixer when muted will cause the LED to Blink continuously. When unmuted the LED will stay On. The Creative driver under Windows sets the LED to blink whenever audio is muted. This LED can be treated as the CMSS LED but I figured since there is just one LED, it should be treated as the Power LED. Is that alright? I've also changed the comment "Usb X-Fi" to "Usb X-Fi S51" as there are other external X-Fi devices from Creative like Usb X-Fi Go and Xmod. The volume knob and LED support patch doesn't apply to them. Signed-off-by: Mandar Joshi <emailmandar(a)gmail.com> diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index e7df1e5..aea92d6 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -60,6 +60,7 @@ static const struct rc_config { { USB_ID(0x041e, 0x3000), 0, 1, 2, 1, 18, 0x0013 }, /* Extigy */ { USB_ID(0x041e, 0x3020), 2, 1, 6, 6, 18, 0x0013 }, /* Audigy 2 NX */ { USB_ID(0x041e, 0x3040), 2, 2, 6, 6, 2, 0x6e91 }, /* Live! 24-bit */ + { USB_ID(0x041e, 0x3042), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 */ { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; @@ -182,10 +183,16 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e if (value > 1) return -EINVAL; changed = value != mixer->audigy2nx_leds[index]; - err = snd_usb_ctl_msg(mixer->chip->dev, - usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, - USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - value, index + 2, NULL, 0, 100); + if (mixer->chip->usb_id == USB_ID(0x041e, 0x3042)) + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + !value, 0, NULL, 0, 100); + else + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + value, index + 2, NULL, 0, 100); if (err < 0) return err; mixer->audigy2nx_leds[index] = value; @@ -224,8 +231,12 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer) int i, err; for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) { + /* USB X-Fi S51 doesn't have a CMSS LED */ + if ((mixer->chip->usb_id == USB_ID(0x041e, 0x3042)) && i == 0) + continue; if (i > 1 && /* Live24ext has 2 LEDs only */ (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3042) || mixer->chip->usb_id == USB_ID(0x041e, 0x3048))) break; err = snd_ctl_add(mixer->chip->card, @@ -364,6 +375,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) || mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3042) || mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) { if ((err = snd_audigy2nx_controls_create(mixer)) < 0) return err; -- 1.7.1

4 5

[alsa-devel] [PATCH 1/7] ASoC: WM8903: Fix mic detection register definitions
by Stephen Warren 13 Feb '11

13 Feb '11

* There is no hysteresis enable field in the current datasheet. * Mic detection threshold field is only 2 bits wide. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- include/sound/wm8903.h | 10 +++------- 1 files changed, 3 insertions(+), 7 deletions(-) diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h index 86172cf..cf7ccb7 100644 --- a/include/sound/wm8903.h +++ b/include/sound/wm8903.h @@ -17,13 +17,9 @@ /* * R6 (0x06) - Mic Bias Control 0 */ -#define WM8903_MICDET_HYST_ENA 0x0080 /* MICDET_HYST_ENA */ -#define WM8903_MICDET_HYST_ENA_MASK 0x0080 /* MICDET_HYST_ENA */ -#define WM8903_MICDET_HYST_ENA_SHIFT 7 /* MICDET_HYST_ENA */ -#define WM8903_MICDET_HYST_ENA_WIDTH 1 /* MICDET_HYST_ENA */ -#define WM8903_MICDET_THR_MASK 0x0070 /* MICDET_THR - [6:4] */ -#define WM8903_MICDET_THR_SHIFT 4 /* MICDET_THR - [6:4] */ -#define WM8903_MICDET_THR_WIDTH 3 /* MICDET_THR - [6:4] */ +#define WM8903_MICDET_THR_MASK 0x0030 /* MICDET_THR - [5:4] */ +#define WM8903_MICDET_THR_SHIFT 4 /* MICDET_THR - [5:4] */ +#define WM8903_MICDET_THR_WIDTH 2 /* MICDET_THR - [5:4] */ #define WM8903_MICSHORT_THR_MASK 0x000C /* MICSHORT_THR - [3:2] */ #define WM8903_MICSHORT_THR_SHIFT 2 /* MICSHORT_THR - [3:2] */ #define WM8903_MICSHORT_THR_WIDTH 2 /* MICSHORT_THR - [3:2] */ -- 1.7.1

3 12

[alsa-devel] [PATCH v2 1/2 2.6.38] ASoC: WM8994: Improve robustness in some use cases
by Dimitris Papastamos 13 Feb '11

13 Feb '11

Ensure that on disabling certain registers such as AIF1DAC1L, AIF1DAC1R etc. the AIF1CLK and AIF2CLK remain enabled. Similarly when enabling those registers, AIF1CLK and AIF2CLK will remain disabled. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 142 ++++++++++++++++++++++++++++++++++++++++++--- 1 files changed, 133 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 37b8aa8..bd0cfdd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -107,6 +107,9 @@ struct wm8994_priv { int revision; struct wm8994_pdata *pdata; + + unsigned int aif1clk_enable:1; + unsigned int aif2clk_enable:1; }; static int wm8994_readable(unsigned int reg) @@ -1004,6 +1007,82 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) } } +static int late_enable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (wm8994->aif1clk_enable) + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, + WM8994_AIF1CLK_ENA); + if (wm8994->aif2clk_enable) + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, + WM8994_AIF2CLK_ENA); + break; + } + + return 0; +} + +static int late_disable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + if (wm8994->aif1clk_enable) { + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, 0); + wm8994->aif1clk_enable = 0; + } + if (wm8994->aif2clk_enable) { + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, 0); + wm8994->aif2clk_enable = 0; + } + break; + } + + return 0; +} + +static int aif1clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wm8994->aif1clk_enable = 1; + break; + } + + return 0; +} + +static int aif2clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wm8994->aif2clk_enable = 1; + break; + } + + return 0; +} + static const char *hp_mux_text[] = { "Mixer", "DAC", @@ -1272,6 +1351,29 @@ static const struct soc_enum aif2dacr_src_enum = static const struct snd_kcontrol_new aif2dacr_src_mux = SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum); +static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + +SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC1R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC2L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), + +SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) +}; + +static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0) +}; + static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), @@ -1284,9 +1386,6 @@ SND_SOC_DAPM_SUPPLY("DSP1CLK", WM8994_CLOCKING_1, 3, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), - SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 9, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, @@ -1516,14 +1615,12 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2ADC Mux", "AIF3DACDAT", "AIF3ADCDAT" }, /* DAC1 inputs */ - { "DAC1L", NULL, "DAC1L Mixer" }, { "DAC1L Mixer", "AIF2 Switch", "AIF2DACL" }, { "DAC1L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, { "DAC1L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, { "DAC1L Mixer", "Left Sidetone Switch", "Left Sidetone" }, { "DAC1L Mixer", "Right Sidetone Switch", "Right Sidetone" }, - { "DAC1R", NULL, "DAC1R Mixer" }, { "DAC1R Mixer", "AIF2 Switch", "AIF2DACR" }, { "DAC1R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, { "DAC1R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, @@ -1532,7 +1629,6 @@ static const struct snd_soc_dapm_route intercon[] = { /* DAC2/AIF2 outputs */ { "AIF2ADCL", NULL, "AIF2DAC2L Mixer" }, - { "DAC2L", NULL, "AIF2DAC2L Mixer" }, { "AIF2DAC2L Mixer", "AIF2 Switch", "AIF2DACL" }, { "AIF2DAC2L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, { "AIF2DAC2L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, @@ -1540,7 +1636,6 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2DAC2L Mixer", "Right Sidetone Switch", "Right Sidetone" }, { "AIF2ADCR", NULL, "AIF2DAC2R Mixer" }, - { "DAC2R", NULL, "AIF2DAC2R Mixer" }, { "AIF2DAC2R Mixer", "AIF2 Switch", "AIF2DACR" }, { "AIF2DAC2R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, { "AIF2DAC2R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, @@ -1584,6 +1679,24 @@ static const struct snd_soc_dapm_route intercon[] = { { "Right Headphone Mux", "DAC", "DAC1R" }, }; +static const struct snd_soc_dapm_route wm8994_lateclk_revd_intercon[] = { + { "DAC1L", NULL, "Late DAC1L Enable PGA" }, + { "Late DAC1L Enable PGA", NULL, "DAC1L Mixer" }, + { "DAC1R", NULL, "Late DAC1R Enable PGA" }, + { "Late DAC1R Enable PGA", NULL, "DAC1R Mixer" }, + { "DAC2L", NULL, "Late DAC2L Enable PGA" }, + { "Late DAC2L Enable PGA", NULL, "AIF2DAC2L Mixer" }, + { "DAC2R", NULL, "Late DAC2R Enable PGA" }, + { "Late DAC2R Enable PGA", NULL, "AIF2DAC2R Mixer" } +}; + +static const struct snd_soc_dapm_route wm8994_lateclk_intercon[] = { + { "DAC1L", NULL, "DAC1L Mixer" }, + { "DAC1R", NULL, "DAC1R Mixer" }, + { "DAC2L", NULL, "AIF2DAC2L Mixer" }, + { "DAC2R", NULL, "AIF2DAC2R Mixer" }, +}; + static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { { "AIF1DACDAT", NULL, "AIF2DACDAT" }, { "AIF2DACDAT", NULL, "AIF1DACDAT" }, @@ -3125,6 +3238,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8994: snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets, ARRAY_SIZE(wm8994_specific_dapm_widgets)); + if (wm8994->revision < 4) + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, + ARRAY_SIZE(wm8994_lateclk_revd_widgets)); + else + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, + ARRAY_SIZE(wm8994_lateclk_widgets)); break; case WM8958: snd_soc_add_controls(codec, wm8958_snd_controls, @@ -3143,10 +3262,15 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, wm8994_intercon, ARRAY_SIZE(wm8994_intercon)); - if (wm8994->revision < 4) + if (wm8994->revision < 4) { snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); - + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, + ARRAY_SIZE(wm8994_lateclk_revd_intercon)); + } else { + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, + ARRAY_SIZE(wm8994_lateclk_intercon)); + } break; case WM8958: snd_soc_dapm_add_routes(dapm, wm8958_intercon, -- 1.7.4

3 3

[alsa-devel] [PATCH] ASoC: Ensure supplies are maintained for force enabled widgets
by Mark Brown 13 Feb '11

13 Feb '11

If a widget has been force enabled then not only do we need to keep the widget itself enabled, we also need to keep any supplies the widget requires enabled. The user could force all the individual widgets on but this requires too much knowledge of device internals. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This is only compile tested so far. sound/soc/soc-dapm.c | 10 +++++++++- 1 files changed, 9 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b30eda6..1bd24d5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -712,7 +712,15 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) !path->connected(path->source, path->sink)) continue; - if (path->sink && path->sink->power_check && + if (!path->sink) + continue; + + if (path->sink->force) { + power = 1; + break; + } + + if (path->sink->power_check && path->sink->power_check(path->sink)) { power = 1; break; -- 1.7.2.3

3 2

[alsa-devel] [PATCH] ASoC: Warn if WM8903 platform data is used to enable microphone IRQ
by Mark Brown 13 Feb '11

13 Feb '11

The WM8903 interrupts are clear on read so if the WM8903 detection is enabled from platform data when the IRQ is in use (rather than using a direct signal from a GPIO) status may be lost during startup. Help users spot this misconfiguration by adding a WARN_ON(). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 22 ++++++++++++++++++++++ 1 files changed, 22 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 20325b3..b61fe76 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2119,12 +2119,26 @@ static int wm8903_probe(struct snd_soc_codec *codec) /* Set up GPIOs and microphone detection */ if (pdata) { + bool mic_gpio = false; + for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { if (pdata->gpio_cfg[i] == WM8903_GPIO_NO_CONFIG) continue; snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, pdata->gpio_cfg[i] & 0xffff); + + val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) + >> WM8903_GP1_FN_SHIFT; + + switch (val) { + case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT: + case WM8903_GPn_FN_MICBIAS_SHORT_DETECT: + mic_gpio = true; + break; + default: + break; + } } snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0, @@ -2135,6 +2149,14 @@ static int wm8903_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0, WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); + /* If microphone detection is enabled by pdata but + * detected via IRQ then interrupts can be lost before + * the machine driver has set up microphone detection + * IRQs as the IRQs are clear on read. The detection + * will be enabled when the machine driver configures. + */ + WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); + wm8903->mic_delay = pdata->micdet_delay; } -- 1.7.2.3

2 1

[PATCH] Add a quirk to ignore the Master volume control on 08bb:2704
by Omari Stephens 12 Feb '11

12 Feb '11

--- mixer.c | 6 ++++++ mixer_maps.c | 19 ++++++++++++++++++- 2 files changed, 24 insertions(+), 1 deletions(-) diff --git a/mixer.c b/mixer.c index 3ed3901..38d5d77 100644 --- a/mixer.c +++ b/mixer.c @@ -1192,6 +1192,12 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void /* disable non-functional volume control */ master_bits &= ~UAC_CONTROL_BIT(UAC_FU_VOLUME); break; + case USB_ID(0x08bb, 0x2704): + snd_printk(KERN_INFO + "usbmixer: master volume quirk for PCM2704 chip\n"); + /* disable non-functional volume control */ + master_bits &= ~UAC_CONTROL_BIT(UAC_FU_VOLUME); + break; } if (channels > 0) first_ch_bits = snd_usb_combine_bytes(bmaControls + csize, csize); diff --git a/mixer_maps.c b/mixer_maps.c index f1324c4..40edd9b 100644 --- a/mixer_maps.c +++ b/mixer_maps.c @@ -208,11 +208,23 @@ static struct usbmix_name_map live24ext_map[] = { /* LineX FM Transmitter entry - needed to bypass controls bug */ static struct usbmix_name_map linex_map[] = { /* 1: IT pcm */ - /* 2: OT Speaker */ + /* 2: OT Speaker */ { 3, "Master" }, /* FU: master volume - left / right / mute */ { 0 } /* terminator */ }; +/* This DAC (08bb:2704) has a single "Volume" control which isn't actually + * connected to anything. The 08bb:2702 device is based off the LineX map, so + * presumably the 2704 one is similar but more broken. + */ +static struct usbmix_name_map ti_broken_usb_dac_map[] = { + /* 1: IT pcm */ + /* 2: OT Speaker */ + { 3, NULL }, /* Master (not connected to anything) */ + { 0 } +}; + + static struct usbmix_name_map maya44_map[] = { /* 1: IT line */ { 2, "Line Playback" }, /* FU */ @@ -355,6 +367,11 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .ignore_ctl_error = 1, }, { + .id = USB_ID(0x08bb, 0x2704), + .map = ti_broken_usb_dac_map, + .ignore_ctl_error = 1, + }, + { .id = USB_ID(0x0a92, 0x0091), .map = maya44_map, }, -- 1.7.2.3 --------------070202010806020703010904--

1 0

[alsa-devel] [PATCH 13/17] twl4030: mfd_cell is now implicitly available to drivers
by Andres Salomon 12 Feb '11

12 Feb '11

The cell's platform_data is now accessed with a helper function; change clients to use that, and remove the now-unused data_size. Signed-off-by: Andres Salomon <dilinger(a)queued.net> --- drivers/input/misc/twl4030-vibra.c | 2 +- drivers/mfd/twl4030-codec.c | 2 -- sound/soc/codecs/twl4030.c | 5 +++-- 3 files changed, 4 insertions(+), 5 deletions(-) diff --git a/drivers/input/misc/twl4030-vibra.c b/drivers/input/misc/twl4030-vibra.c index 014dd4a..1246ae1 100644 --- a/drivers/input/misc/twl4030-vibra.c +++ b/drivers/input/misc/twl4030-vibra.c @@ -196,7 +196,7 @@ static SIMPLE_DEV_PM_OPS(twl4030_vibra_pm_ops, static int __devinit twl4030_vibra_probe(struct platform_device *pdev) { - struct twl4030_codec_vibra_data *pdata = pdev->dev.platform_data; + struct twl4030_codec_vibra_data *pdata = mfd_get_data(pdev); struct vibra_info *info; int ret; diff --git a/drivers/mfd/twl4030-codec.c b/drivers/mfd/twl4030-codec.c index 9a4b196..0f57426 100644 --- a/drivers/mfd/twl4030-codec.c +++ b/drivers/mfd/twl4030-codec.c @@ -209,14 +209,12 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) cell = &codec->cells[childs]; cell->name = "twl4030-codec"; cell->platform_data = pdata->audio; - cell->data_size = sizeof(*pdata->audio); childs++; } if (pdata->vibra) { cell = &codec->cells[childs]; cell->name = "twl4030-vibra"; cell->platform_data = pdata->vibra; - cell->data_size = sizeof(*pdata->vibra); childs++; } diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e4d464b..37004afb 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -732,7 +732,8 @@ static int aif_event(struct snd_soc_dapm_widget *w, static void headset_ramp(struct snd_soc_codec *codec, int ramp) { - struct twl4030_codec_audio_data *pdata = codec->dev->platform_data; + struct twl4030_codec_audio_data *pdata = + mfd_get_data(to_platform_device(codec->dev)); unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); /* Base values for ramp delay calculation: 2^19 - 2^26 */ @@ -2297,7 +2298,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { static int __devinit twl4030_codec_probe(struct platform_device *pdev) { - struct twl4030_codec_audio_data *pdata = pdev->dev.platform_data; + struct twl4030_codec_audio_data *pdata = mfd_get_data(pdev); if (!pdata) { dev_err(&pdev->dev, "platform_data is missing\n"); -- 1.7.2.3

2 1

[alsa-devel] [PATCH 10/17] wl1273: mfd_cell is now implicitly available to drivers
by Andres Salomon 12 Feb '11

12 Feb '11

The cell's platform_data is now accessed with a helper function; change clients to use that, and remove the now-unused data_size. Signed-off-by: Andres Salomon <dilinger(a)queued.net> --- drivers/media/radio/radio-wl1273.c | 2 +- drivers/mfd/wl1273-core.c | 2 -- sound/soc/codecs/wl1273.c | 3 ++- 3 files changed, 3 insertions(+), 4 deletions(-) diff --git a/drivers/media/radio/radio-wl1273.c b/drivers/media/radio/radio-wl1273.c index 7ecc8e6..4698eb0 100644 --- a/drivers/media/radio/radio-wl1273.c +++ b/drivers/media/radio/radio-wl1273.c @@ -2138,7 +2138,7 @@ static int wl1273_fm_radio_remove(struct platform_device *pdev) static int __devinit wl1273_fm_radio_probe(struct platform_device *pdev) { - struct wl1273_core **core = pdev->dev.platform_data; + struct wl1273_core **core = mfd_get_data(pdev); struct wl1273_device *radio; struct v4l2_ctrl *ctrl; int r = 0; diff --git a/drivers/mfd/wl1273-core.c b/drivers/mfd/wl1273-core.c index d2ecc24..703085e 100644 --- a/drivers/mfd/wl1273-core.c +++ b/drivers/mfd/wl1273-core.c @@ -80,7 +80,6 @@ static int __devinit wl1273_core_probe(struct i2c_client *client, cell = &core->cells[children]; cell->name = "wl1273_fm_radio"; cell->platform_data = &core; - cell->data_size = sizeof(core); children++; if (pdata->children & WL1273_CODEC_CHILD) { @@ -89,7 +88,6 @@ static int __devinit wl1273_core_probe(struct i2c_client *client, dev_dbg(&client->dev, "%s: Have codec.\n", __func__); cell->name = "wl1273-codec"; cell->platform_data = &core; - cell->data_size = sizeof(core); children++; } diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 861b28f..1ad0d5a 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -436,7 +436,8 @@ EXPORT_SYMBOL_GPL(wl1273_get_format); static int wl1273_probe(struct snd_soc_codec *codec) { - struct wl1273_core **core = codec->dev->platform_data; + struct wl1273_core **core = + mfd_get_data(to_platform_device(codec->dev)); struct wl1273_priv *wl1273; int r; -- 1.7.2.3

1 0

[alsa-devel] [PATCH 1/2 2.6.38] ASoC: soc-cache: Introduce the cache_bypass option
by Dimitris Papastamos 11 Feb '11

11 Feb '11

This is primarily needed to avoid writing back to the cache whenever we are syncing the cache with the hardware. This gives a performance benefit especially for large register maps. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-cache.c | 36 ++++++++++++++++++++++++------------ 2 files changed, 25 insertions(+), 12 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 74921f2..5e19644 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -463,6 +463,7 @@ struct snd_soc_codec { /* runtime */ struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ unsigned int active; + unsigned int cache_bypass:1; /* Suppress access to the cache */ unsigned int cache_only:1; /* Suppress writes to hardware */ unsigned int cache_sync:1; /* Cache needs to be synced to hardware */ unsigned int suspended:1; /* Codec is in suspend PM state */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 8c2a21a..799e376 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -25,7 +25,8 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -49,7 +50,8 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0x00ff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -106,7 +108,8 @@ static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -130,7 +133,8 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0x00ff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -191,7 +195,8 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -216,7 +221,8 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, reg &= 0xff; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -271,7 +277,8 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, data[2] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -295,7 +302,8 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -450,7 +458,8 @@ static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, reg &= 0xff; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -476,7 +485,8 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, reg &= 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -568,7 +578,8 @@ static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -595,7 +606,8 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, data[3] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; -- 1.7.4

2 6

Re: [alsa-devel] [PATCH] at73c213: add support for at73c240 devices
by Grant Likely 11 Feb '11

11 Feb '11

On Fri, Feb 11, 2011 at 8:56 AM, Andreas Bießmann <biessmann(a)corscience.de> wrote: > at73c240 is a successor of mature at73c213 and nearly register compatible. > > See http://www.atmel.com/dyn/resources/prod_documents/doc6484.pdf for > comparison of these two devices. > > Signed-off-by: Andreas Bießmann <biessmann(a)corscience.de> Looks okay to me. Briefly-glanced-at-by: Grant Likely <grant.likely(a)secretlab.ca> > --- > include/linux/spi/at73c213.h | 1 + > sound/spi/at73c213.c | 122 +++++++++++++++++++++++++++++++++--------- > sound/spi/at73c213.h | 3 + > 3 files changed, 101 insertions(+), 25 deletions(-) > > diff --git a/include/linux/spi/at73c213.h b/include/linux/spi/at73c213.h > index 0f20a70e..26d1f38 100644 > --- a/include/linux/spi/at73c213.h > +++ b/include/linux/spi/at73c213.h > @@ -19,6 +19,7 @@ > struct at73c213_board_info { > int ssc_id; > struct clk *dac_clk; > + bool is_at73c240; > char shortname[32]; > }; > > diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c > index 503ffb0..711aae4 100644 > --- a/sound/spi/at73c213.c > +++ b/sound/spi/at73c213.c > @@ -709,10 +709,44 @@ AT73C213_MONO_SWITCH("Aux Capture Switch", 0, DAC_CTRL, DAC_CTRL_ONAUXIN, > AT73C213_MONO_SWITCH("Line Capture Switch", 0, DAC_CTRL, 0, 0x03, 0), > }; > > +static struct snd_kcontrol_new snd_at73c240_controls[] __devinitdata = { > +AT73C213_STEREO("Master Playback Volume", 0, DAC_LMPG, DAC_RMPG, 0, 0, 0x1f, 1), > +AT73C213_STEREO("Master Playback Switch", 0, DAC_LMPG, DAC_RMPG, 5, 5, 1, 1), > +AT73C213_STEREO("PCM Playback Volume", 0, DAC_LLOG, DAC_RLOG, 0, 0, 0x1f, 1), > +AT73C213_STEREO("PCM Playback Switch", 0, DAC_LLOG, DAC_RLOG, 5, 5, 1, 1), > +AT73C213_MONO_SWITCH("Mono PA Playback Switch", 0, DAC_CTRL, > + (DAC_CTRL_ONPADRV-1), 0x01, 0), > +{ > + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .name = "PA Playback Volume", > + .index = 0, > + .info = snd_at73c213_pa_volume_info, > + .get = snd_at73c213_mono_get, > + .put = snd_at73c213_mono_put, > + .private_value = PA_CTRL | (PA_CTRL_APAGAIN << 8) | \ > + (0x0f << 16) | (1 << 24), > +}, > +AT73C213_MONO_SWITCH("PA Playback Switch", 0, PA_CTRL, (PA_CTRL_APAON-1), > + 0x01, 0), > +{ > + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .name = "Line Capture Volume", > + .index = 0, > + .info = snd_at73c213_line_capture_volume_info, > + .get = snd_at73c213_stereo_get, > + .put = snd_at73c213_stereo_put, > + .private_value = DAC_LLIG | (DAC_RLIG << 8) | (0 << 16) | (0 << 19) > + | (0x1f << 24) | (1 << 22), > +}, > +AT73C213_MONO_SWITCH("Line Capture Switch", 0, DAC_CTRL, 0, 0x03, 0), > +}; > + > static int __devinit snd_at73c213_mixer(struct snd_at73c213 *chip) > { > struct snd_card *card; > int errval, idx; > + struct snd_kcontrol_new *controls; > + int controls_size = 0; > > if (chip == NULL || chip->pcm == NULL) > return -EINVAL; > @@ -721,9 +755,17 @@ static int __devinit snd_at73c213_mixer(struct snd_at73c213 *chip) > > strcpy(card->mixername, chip->pcm->name); > > - for (idx = 0; idx < ARRAY_SIZE(snd_at73c213_controls); idx++) { > + if (chip->board->is_at73c240) { > + controls_size = ARRAY_SIZE(snd_at73c240_controls); > + controls = snd_at73c240_controls; > + } else { > + controls_size = ARRAY_SIZE(snd_at73c213_controls); > + controls = snd_at73c213_controls; > + } > + > + for (idx = 0; idx < controls_size; idx++) { > errval = snd_ctl_add(card, > - snd_ctl_new1(&snd_at73c213_controls[idx], > + snd_ctl_new1(&controls[idx], > chip)); > if (errval < 0) > goto cleanup; > @@ -732,7 +774,7 @@ static int __devinit snd_at73c213_mixer(struct snd_at73c213 *chip) > return 0; > > cleanup: > - for (idx = 1; idx < ARRAY_SIZE(snd_at73c213_controls) + 1; idx++) { > + for (idx = 1; idx < controls_size + 1; idx++) { > struct snd_kcontrol *kctl; > kctl = snd_ctl_find_numid(card, idx); > if (kctl) > @@ -777,7 +819,7 @@ static int __devinit snd_at73c213_ssc_init(struct snd_at73c213 *chip) > static int __devinit snd_at73c213_chip_init(struct snd_at73c213 *chip) > { > int retval; > - unsigned char dac_ctrl = 0; > + unsigned char tmp_reg; > > retval = snd_at73c213_set_bitrate(chip); > if (retval) > @@ -799,7 +841,11 @@ static int __devinit snd_at73c213_chip_init(struct snd_at73c213 *chip) > retval = snd_at73c213_write_reg(chip, DAC_PRECH, 0xff); > if (retval) > goto out_clk; > - retval = snd_at73c213_write_reg(chip, PA_CTRL, (1<<PA_CTRL_APAPRECH)); > + if (chip->board->is_at73c240) > + tmp_reg = (1<<(PA_CTRL_APAPRECH-1)); > + else > + tmp_reg = (1<<PA_CTRL_APAPRECH); > + retval = snd_at73c213_write_reg(chip, PA_CTRL, tmp_reg); > if (retval) > goto out_clk; > retval = snd_at73c213_write_reg(chip, DAC_CTRL, > @@ -809,11 +855,13 @@ static int __devinit snd_at73c213_chip_init(struct snd_at73c213 *chip) > > msleep(50); > > - /* Stop precharging PA. */ > - retval = snd_at73c213_write_reg(chip, PA_CTRL, > - (1<<PA_CTRL_APALP) | 0x0f); > - if (retval) > - goto out_clk; > + if (!chip->board->is_at73c240) { > + /* Stop precharging PA. */ > + retval = snd_at73c213_write_reg(chip, PA_CTRL, > + (1<<PA_CTRL_APALP) | 0x0f); > + if (retval) > + goto out_clk; > + } > > msleep(450); > > @@ -825,12 +873,18 @@ static int __devinit snd_at73c213_chip_init(struct snd_at73c213 *chip) > msleep(1); > > /* Turn on DAC. */ > - dac_ctrl = (1<<DAC_CTRL_ONDACL) | (1<<DAC_CTRL_ONDACR) > - | (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR) > - /* enable MONO PA */ > - | (1<<DAC_CTRL_ONPADRV); > + if (chip->board->is_at73c240) > + tmp_reg = (1<<DAC_CTRL_ONDACL) | (1<<DAC_CTRL_ONDACR) > + | (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR) > + /* enable MONO PA */ > + | (1<<(DAC_CTRL_ONPADRV-1)); > + else > + tmp_reg = (1<<DAC_CTRL_ONDACL) | (1<<DAC_CTRL_ONDACR) > + | (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR) > + /* enable MONO PA */ > + | (1<<DAC_CTRL_ONPADRV); > > - retval = snd_at73c213_write_reg(chip, DAC_CTRL, dac_ctrl); > + retval = snd_at73c213_write_reg(chip, DAC_CTRL, tmp_reg); > if (retval) > goto out_clk; > > @@ -859,9 +913,19 @@ static int __devinit snd_at73c213_chip_init(struct snd_at73c213 *chip) > retval = snd_at73c213_write_reg(chip, DAC_RLIG, 0x11); > if (retval) > goto out_clk; > - retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); > - if (retval) > - goto out_clk; > + if (!chip->board->is_at73c240) { > + retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); > + if (retval) > + goto out_clk; > + } > + > + /* enable I2S clock */ > + if (chip->board->is_at73c240) { > + retval = snd_at73c213_write_reg(chip, DAC_IS_CTRL, > + (1<<DAC_IS_CTRL_IS_CTRL)); > + if (retval) > + goto out_clk; > + } > > /* Enable I2S device, i.e. clock output. */ > ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN)); > @@ -915,6 +979,10 @@ static int __devinit snd_at73c213_dev_init(struct snd_card *card, > memcpy(&chip->reg_image, &snd_at73c213_original_image, > sizeof(snd_at73c213_original_image)); > > + /* at73c240 has another reset value here */ > + if (chip->board->is_at73c240) > + chip->reg_image[PA_CTRL] = 0x0f; > + > retval = snd_at73c213_ssc_init(chip); > if (retval) > goto out_irq; > @@ -1043,9 +1111,11 @@ static int __devexit snd_at73c213_remove(struct spi_device *spi) > retval = snd_at73c213_write_reg(chip, DAC_RLIG, 0x11); > if (retval) > goto out; > - retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); > - if (retval) > - goto out; > + if (!chip->board->is_at73c240) { > + retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); > + if (retval) > + goto out; > + } > > /* Turn off PA. */ > retval = snd_at73c213_write_reg(chip, PA_CTRL, > @@ -1053,10 +1123,12 @@ static int __devexit snd_at73c213_remove(struct spi_device *spi) > if (retval) > goto out; > msleep(10); > - retval = snd_at73c213_write_reg(chip, PA_CTRL, > - (1 << PA_CTRL_APALP) | 0x0f); > - if (retval) > - goto out; > + if (!chip->board->is_at73c240) { > + retval = snd_at73c213_write_reg(chip, PA_CTRL, > + (1 << PA_CTRL_APALP) | 0x0f); > + if (retval) > + goto out; > + } > > /* Turn off external DAC. */ > retval = snd_at73c213_write_reg(chip, DAC_CTRL, 0x0c); > diff --git a/sound/spi/at73c213.h b/sound/spi/at73c213.h > index fd8b372..3712d6e 100644 > --- a/sound/spi/at73c213.h > +++ b/sound/spi/at73c213.h > @@ -88,6 +88,9 @@ > #define DAC_MISC_DEEMPEN 2 > #define DAC_MISC_NBITS 0 > > +#define DAC_IS_CTRL 0x0B > +#define DAC_IS_CTRL_IS_CTRL 2 > + > /* DAC Precharge Control Register */ > #define DAC_PRECH 0x0C > #define DAC_PRECH_PRCHGPDRV 7 > -- > 1.7.2.3 > > > ------------------------------------------------------------------------------ > The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE: > Pinpoint memory and threading errors before they happen. > Find and fix more than 250 security defects in the development cycle. > Locate bottlenecks in serial and parallel code that limit performance. > http://p.sf.net/sfu/intel-dev2devfeb > _______________________________________________ > spi-devel-general mailing list > spi-devel-general(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/spi-devel-general > -- Grant Likely, B.Sc., P.Eng. Secret Lab Technologies Ltd.

1 0

[alsa-devel] finally, ALSA 1.0.24 is out
by Jaroslav Kysela 11 Feb '11

11 Feb '11

Hello all, after very intensive testing, I released all ALSA packages with version number 1.0.24(+) : * alsa-driver-1.0.24 * alsa-lib-1.0.24.1 * alsa-utils-1.0.24.2 * alsa-tools-1.0.24 * alsa-firmware-1.0.24 * alsa-plugins-1.0.24 * pyalsa-1.0.24 The full list of changes is here: http://www.alsa-project.org/main/index.php/Changes_v1.0.23_v1.0.24 There is a ton of work in these parts (among others): * alsa-driver - HDA driver - ASoC tree - USB driver * alsa-library - Use Case Manager * alsa-utils - alsaloop and alsaucm introduction - alsactl improvements * alsa-tools - updated hdspmixer Please, report compilation problems directly to me or to the alsa-devel mailing list. Enjoy this release, Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] QasMixer 0.7.1 now available
by Sebastian H. 11 Feb '11

11 Feb '11

Hello Alsa and Qt users QasMixer version 0.7.1 is now available. This point release brings persistent application settings and some GUI cleanups. New features * Persistent application settings managed by QT's QSettings class * A settings dialog * Icons from the system theme are used for menus * A sort option for the Alsa configuration tree Changes * The Alsa configuration view has moved to a dialog * System tray command line options were removed in favour of application settings * Many small GUI tweaks * Some bug fixes The homepage has been updated with screenshots http://xwmw.org/qasmixer and files are hosted at sourceforge http://sourceforge.net/projects/qasmixer/ Happy volume changing! -- Sebastian Holtermann

1 0

[alsa-devel] [PATCH 1/2 2.6.38] ASoC: WM8994: Improve robustness in some use cases
by Dimitris Papastamos 11 Feb '11

11 Feb '11

Ensure that on disabling certain registers such as AIF1DAC1L, AIF1DAC1R etc. the AIF1CLK and AIF2CLK remain enabled. Similarly when enabling those registers, AIF1CLK and AIF2CLK will remain disabled. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 142 ++++++++++++++++++++++++++++++++++++++++++--- 1 files changed, 133 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 37b8aa8..bd0cfdd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -107,6 +107,9 @@ struct wm8994_priv { int revision; struct wm8994_pdata *pdata; + + unsigned int aif1clk_enable:1; + unsigned int aif2clk_enable:1; }; static int wm8994_readable(unsigned int reg) @@ -1004,6 +1007,82 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) } } +static int late_enable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (wm8994->aif1clk_enable) + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, + WM8994_AIF1CLK_ENA); + if (wm8994->aif2clk_enable) + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, + WM8994_AIF2CLK_ENA); + break; + } + + return 0; +} + +static int late_disable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + if (wm8994->aif1clk_enable) { + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, 0); + wm8994->aif1clk_enable = 0; + } + if (wm8994->aif2clk_enable) { + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, 0); + wm8994->aif2clk_enable = 0; + } + break; + } + + return 0; +} + +static int aif1clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wm8994->aif1clk_enable = 1; + break; + } + + return 0; +} + +static int aif2clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wm8994->aif2clk_enable = 1; + break; + } + + return 0; +} + static const char *hp_mux_text[] = { "Mixer", "DAC", @@ -1272,6 +1351,29 @@ static const struct soc_enum aif2dacr_src_enum = static const struct snd_kcontrol_new aif2dacr_src_mux = SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum); +static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + +SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC1R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC2L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), + +SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) +}; + +static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0) +}; + static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), @@ -1284,9 +1386,6 @@ SND_SOC_DAPM_SUPPLY("DSP1CLK", WM8994_CLOCKING_1, 3, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), - SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 9, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, @@ -1516,14 +1615,12 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2ADC Mux", "AIF3DACDAT", "AIF3ADCDAT" }, /* DAC1 inputs */ - { "DAC1L", NULL, "DAC1L Mixer" }, { "DAC1L Mixer", "AIF2 Switch", "AIF2DACL" }, { "DAC1L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, { "DAC1L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, { "DAC1L Mixer", "Left Sidetone Switch", "Left Sidetone" }, { "DAC1L Mixer", "Right Sidetone Switch", "Right Sidetone" }, - { "DAC1R", NULL, "DAC1R Mixer" }, { "DAC1R Mixer", "AIF2 Switch", "AIF2DACR" }, { "DAC1R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, { "DAC1R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, @@ -1532,7 +1629,6 @@ static const struct snd_soc_dapm_route intercon[] = { /* DAC2/AIF2 outputs */ { "AIF2ADCL", NULL, "AIF2DAC2L Mixer" }, - { "DAC2L", NULL, "AIF2DAC2L Mixer" }, { "AIF2DAC2L Mixer", "AIF2 Switch", "AIF2DACL" }, { "AIF2DAC2L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, { "AIF2DAC2L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, @@ -1540,7 +1636,6 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2DAC2L Mixer", "Right Sidetone Switch", "Right Sidetone" }, { "AIF2ADCR", NULL, "AIF2DAC2R Mixer" }, - { "DAC2R", NULL, "AIF2DAC2R Mixer" }, { "AIF2DAC2R Mixer", "AIF2 Switch", "AIF2DACR" }, { "AIF2DAC2R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, { "AIF2DAC2R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, @@ -1584,6 +1679,24 @@ static const struct snd_soc_dapm_route intercon[] = { { "Right Headphone Mux", "DAC", "DAC1R" }, }; +static const struct snd_soc_dapm_route wm8994_lateclk_revd_intercon[] = { + { "DAC1L", NULL, "Late DAC1L Enable PGA" }, + { "Late DAC1L Enable PGA", NULL, "DAC1L Mixer" }, + { "DAC1R", NULL, "Late DAC1R Enable PGA" }, + { "Late DAC1R Enable PGA", NULL, "DAC1R Mixer" }, + { "DAC2L", NULL, "Late DAC2L Enable PGA" }, + { "Late DAC2L Enable PGA", NULL, "AIF2DAC2L Mixer" }, + { "DAC2R", NULL, "Late DAC2R Enable PGA" }, + { "Late DAC2R Enable PGA", NULL, "AIF2DAC2R Mixer" } +}; + +static const struct snd_soc_dapm_route wm8994_lateclk_intercon[] = { + { "DAC1L", NULL, "DAC1L Mixer" }, + { "DAC1R", NULL, "DAC1R Mixer" }, + { "DAC2L", NULL, "AIF2DAC2L Mixer" }, + { "DAC2R", NULL, "AIF2DAC2R Mixer" }, +}; + static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { { "AIF1DACDAT", NULL, "AIF2DACDAT" }, { "AIF2DACDAT", NULL, "AIF1DACDAT" }, @@ -3125,6 +3238,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8994: snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets, ARRAY_SIZE(wm8994_specific_dapm_widgets)); + if (wm8994->revision < 4) + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, + ARRAY_SIZE(wm8994_lateclk_revd_widgets)); + else + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, + ARRAY_SIZE(wm8994_lateclk_widgets)); break; case WM8958: snd_soc_add_controls(codec, wm8958_snd_controls, @@ -3143,10 +3262,15 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, wm8994_intercon, ARRAY_SIZE(wm8994_intercon)); - if (wm8994->revision < 4) + if (wm8994->revision < 4) { snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); - + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, + ARRAY_SIZE(wm8994_lateclk_revd_intercon)); + } else { + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, + ARRAY_SIZE(wm8994_lateclk_intercon)); + } break; case WM8958: snd_soc_dapm_add_routes(dapm, wm8958_intercon, -- 1.7.4

2 2

Re: [alsa-devel] [PATCH 3/3] ASoC: Add support for AIF channel muxing on WM8903
by Stephen Warren 11 Feb '11

11 Feb '11

Stephen Warren wrote at Wednesday, February 09, 2011 9:33 AM: > > Mark Brown wrote at Wednesday, February 09, 2011 6:47 AM: > > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > --- > > sound/soc/codecs/wm8903.c | 66 ++++++++++++++++++++++++++++++++++++++++++--- > > 1 files changed, 62 insertions(+), 4 deletions(-) > > ...I'll make sure to test it within the next few days, once it's applied. Mark, I did find one issue with this. With the default Left/Right Capture Mux settings, and the patch series I just posted, mic detection now works fine in all the scenarios I tested: 1) Mic plugged in before loading driver, or not. 2) Mic plugged in when starting to capture, or not. 3) Mic removed and plugged back in while capture active. However, if I flip the Right Capture Mux to setting Left (while not capturing), then this breaks case (3) above; the initial portion of the recording is fine, but when I remove and plug the mic back in, something isn't turned on, and so nothing is recorded after that. I haven't had a chance to look into this yet, but can later; probably on Saturday evening. -- nvpublic

2 3

[alsa-devel] [PATCH] ASoC: Allow use sleeping gpio in soc-jack
by Jarkko Nikula 11 Feb '11

11 Feb '11

It is safe to use sleeping gpio in snd_soc_jack_gpio_detect as it is not called from interrupt context. This avoids WARN_ON from __gpio_get_value if sleeping gpio is registered for jack. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- Only compile tested. --- sound/soc/soc-jack.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 99dbaf7..4579ee0 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -240,7 +240,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) int enable; int report; - enable = gpio_get_value(gpio->gpio); + enable = gpio_get_value_cansleep(gpio->gpio); if (gpio->invert) enable = !enable; -- 1.7.2.3

3 2

[alsa-devel] [PATCH] ASoC: mid-x86: Use the soc-jack apis for jack type detection
by Koul, Vinod 11 Feb '11

11 Feb '11

From: Vinod Koul <vinod.koul(a)intel.com> This patch modifies the mfld_machine to use the new jack apis for adding the voltage zones for jack type detection. It also modifed TI sn95031 codec driver to use these new apis Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/codecs/sn95031.c | 4 +--- sound/soc/mid-x86/mfld_machine.c | 12 ++++++++++++ 2 files changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 4cc0017..d0b7802 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -782,9 +782,7 @@ static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack) { int micbias = sn95031_get_mic_bias(mfld_jack->codec); - /* Defaulting to HEADSET for now. - * will change after adding soc-jack detection apis */ - int jack_type = SND_JACK_HEADSET; + int jack_type = snd_soc_jack_get_type(mfld_jack, micbias); pr_debug("jack type detected = %d\n", jack_type); if (jack_type == SND_JACK_HEADSET) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 45a0067..96487fb 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -74,6 +74,12 @@ static struct snd_soc_jack_pin mfld_jack_pins[] = { }, }; +/* jack detection voltage zones */ +static struct snd_soc_jack_zone mfld_zones[] = { + {MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE}, + {MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET}, +}; + /* sound card controls */ static const char *headset_switch_text[] = {"Earpiece", "Headset"}; @@ -264,6 +270,12 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) pr_err("adding jack pins failed\n"); return ret_val; } + ret_val = snd_soc_jack_add_zones(&mfld_jack, + ARRAY_SIZE(mfld_zones), mfld_zones); + if (ret_val) { + pr_err("adding jack zones failed\n"); + return ret_val; + } /* we want to check if anything is inserted at boot, * so send a fake event to codec and it will read adc -- 1.7.2.3

3 2

[alsa-devel] [PATCH 1/2] ASoC: WM8994: Improve robustness in some use cases
by Dimitris Papastamos 11 Feb '11

11 Feb '11

Ensure that on disabling certain registers such as AIF1DAC1L, AIF1DAC1R etc. the AIF1CLK and AIF2CLK remain enabled. Similarly when enabling those registers, AIF1CLK and AIF2CLK will remain disabled. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 142 ++++++++++++++++++++++++++++++++++++++++++--- 1 files changed, 133 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0024b4c..055ad8f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -107,6 +107,9 @@ struct wm8994_priv { int revision; struct wm8994_pdata *pdata; + + unsigned int aif1clk_enable:1; + unsigned int aif2clk_enable:1; }; static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) @@ -1004,6 +1007,82 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) } } +static int late_enable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (wm8994->aif1clk_enable) + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, + WM8994_AIF1CLK_ENA); + if (wm8994->aif2clk_enable) + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, + WM8994_AIF2CLK_ENA); + break; + } + + return 0; +} + +static int late_disable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + if (wm8994->aif1clk_enable) { + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, 0); + wm8994->aif1clk_enable = 0; + } + if (wm8994->aif2clk_enable) { + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, 0); + wm8994->aif2clk_enable = 0; + } + break; + } + + return 0; +} + +static int aif1clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wm8994->aif1clk_enable = 1; + break; + } + + return 0; +} + +static int aif2clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wm8994->aif2clk_enable = 1; + break; + } + + return 0; +} + static const char *hp_mux_text[] = { "Mixer", "DAC", @@ -1272,6 +1351,29 @@ static const struct soc_enum aif2dacr_src_enum = static const struct snd_kcontrol_new aif2dacr_src_mux = SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum); +static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + +SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC1R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC2L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), + +SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) +}; + +static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0) +}; + static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), @@ -1284,9 +1386,6 @@ SND_SOC_DAPM_SUPPLY("DSP1CLK", WM8994_CLOCKING_1, 3, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), - SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 9, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, @@ -1516,14 +1615,12 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2ADC Mux", "AIF3DACDAT", "AIF3ADCDAT" }, /* DAC1 inputs */ - { "DAC1L", NULL, "DAC1L Mixer" }, { "DAC1L Mixer", "AIF2 Switch", "AIF2DACL" }, { "DAC1L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, { "DAC1L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, { "DAC1L Mixer", "Left Sidetone Switch", "Left Sidetone" }, { "DAC1L Mixer", "Right Sidetone Switch", "Right Sidetone" }, - { "DAC1R", NULL, "DAC1R Mixer" }, { "DAC1R Mixer", "AIF2 Switch", "AIF2DACR" }, { "DAC1R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, { "DAC1R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, @@ -1532,7 +1629,6 @@ static const struct snd_soc_dapm_route intercon[] = { /* DAC2/AIF2 outputs */ { "AIF2ADCL", NULL, "AIF2DAC2L Mixer" }, - { "DAC2L", NULL, "AIF2DAC2L Mixer" }, { "AIF2DAC2L Mixer", "AIF2 Switch", "AIF2DACL" }, { "AIF2DAC2L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, { "AIF2DAC2L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, @@ -1540,7 +1636,6 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2DAC2L Mixer", "Right Sidetone Switch", "Right Sidetone" }, { "AIF2ADCR", NULL, "AIF2DAC2R Mixer" }, - { "DAC2R", NULL, "AIF2DAC2R Mixer" }, { "AIF2DAC2R Mixer", "AIF2 Switch", "AIF2DACR" }, { "AIF2DAC2R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, { "AIF2DAC2R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, @@ -1584,6 +1679,24 @@ static const struct snd_soc_dapm_route intercon[] = { { "Right Headphone Mux", "DAC", "DAC1R" }, }; +static const struct snd_soc_dapm_route wm8994_lateclk_revd_intercon[] = { + { "DAC1L", NULL, "Late DAC1L Enable PGA" }, + { "Late DAC1L Enable PGA", NULL, "DAC1L Mixer" }, + { "DAC1R", NULL, "Late DAC1R Enable PGA" }, + { "Late DAC1R Enable PGA", NULL, "DAC1R Mixer" }, + { "DAC2L", NULL, "Late DAC2L Enable PGA" }, + { "Late DAC2L Enable PGA", NULL, "AIF2DAC2L Mixer" }, + { "DAC2R", NULL, "Late DAC2R Enable PGA" }, + { "Late DAC2R Enable PGA", NULL, "AIF2DAC2R Mixer" } +}; + +static const struct snd_soc_dapm_route wm8994_lateclk_intercon[] = { + { "DAC1L", NULL, "DAC1L Mixer" }, + { "DAC1R", NULL, "DAC1R Mixer" }, + { "DAC2L", NULL, "AIF2DAC2L Mixer" }, + { "DAC2R", NULL, "AIF2DAC2R Mixer" }, +}; + static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { { "AIF1DACDAT", NULL, "AIF2DACDAT" }, { "AIF2DACDAT", NULL, "AIF1DACDAT" }, @@ -3125,6 +3238,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8994: snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets, ARRAY_SIZE(wm8994_specific_dapm_widgets)); + if (wm8994->revision < 4) + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, + ARRAY_SIZE(wm8994_lateclk_revd_widgets)); + else + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, + ARRAY_SIZE(wm8994_lateclk_widgets)); break; case WM8958: snd_soc_add_controls(codec, wm8958_snd_controls, @@ -3143,10 +3262,15 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, wm8994_intercon, ARRAY_SIZE(wm8994_intercon)); - if (wm8994->revision < 4) + if (wm8994->revision < 4) { snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); - + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, + ARRAY_SIZE(wm8994_lateclk_revd_intercon)); + } else { + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, + ARRAY_SIZE(wm8994_lateclk_intercon)); + } break; case WM8958: snd_soc_dapm_add_routes(dapm, wm8958_intercon, -- 1.7.3.5

3 3

[alsa-devel] [PATCH 2.6.38] ASoC: CX20442: fix wrong reg_cache_default content
by Janusz Krzysztofik 11 Feb '11

11 Feb '11

Content of the CX20442's snd_soc_codec_driver.reg_cache_default pointed area, introduced with my recent NULL pointer dereferece fix (commit f019ee5feb344ff0b22b58df4568676295aae14f), occured wrong after further testing, more thorough than just booting successfully. There are two problems with it: 1) It should read (1 << CX20442_TELOUT) | (1 << CX20442_MIC), not CX20442_TELOUT | CX20442_MIC. 2) While correctly matching actual codec hardware state on boot when fixed per 1), a few more code modifications would still be required to reflect that state not only into register cache, but also force them into DAPM pins state, otherwise an inconsitency occures which may prevent further codec state changes from being applied correctly. As a result, the phone stops ringing after reboot, until someone picks up the handset for the first time. Revert that reg_cache_default content to a working, previous de facto default value of 0, in hope this change can still be accepted as an rc cycle fix. Created and tested against linux-2.6.38-rc4 Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- sound/soc/codecs/cx20442.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- git/sound/soc/codecs/cx20442.c.orig 2011-02-10 00:52:33.000000000 +0100 +++ git/sound/soc/codecs/cx20442.c 2011-02-10 12:17:31.000000000 +0100 @@ -367,7 +367,7 @@ static int cx20442_codec_remove(struct s return 0; } -static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC; +static const u8 cx20442_reg; static struct snd_soc_codec_driver cx20442_codec_dev = { .probe = cx20442_codec_probe,

3 2

[alsa-devel] C API, usage of function _snd_pcm_file_open()
by Jon Harald Øymyr 11 Feb '11

11 Feb '11

I had a look at the Spotify C API, and found that they were using the following code to output music from Spotify: snd_pcm_open(&h, dev, SND_PCM_STREAM_PLAYBACK, 0). I want use this function instead: int _snd_pcm_file_open(snd_pcm_t **pcmp, const char *name, snd_config_t *root, snd_config_t *conf, snd_pcm_stream_t stream, int mode) The function is found here: http://www.alsa-project.org/alsa-doc/alsa-lib/pcm__file_8c.html#94d0dc725e5… I'm not able to find anything in the documentation explaining what the two parameters snd_config_t *root and snd_config_t *conf is. What I'm trying to achieve is to write the stream to a file, and then read it from another application. But it would be even better if I could make the C application write to standard output. Can anybody help me with this? I'm not very experienced with either C programming or streaming, but I would really like to make this work.

2 1

[alsa-devel] [PATCH resend] ALSA: hda - add quirk for Ordissimo EVE using a realtek ALC662
by Anisse Astier 11 Feb '11

11 Feb '11

This netbook has a only one jack output and an internal mic. By default, mic and jack sense aren't working. Using lenovo-101e parameters makes both work. The device seems based on a Sharetronic Q70, so this should fix audio for this model too. Signed-off-by: Anisse Astier <anisse(a)astier.eu> --- Hi, Sorry for the delay. As usual, this is based on sound-2.6/for-next. Regards, Anisse --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1e092b3..6a43c1c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -18934,6 +18934,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { ALC662_3ST_6ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E), {} }; -- 1.7.3.2

2 1

Re: [alsa-devel] [PATCH] asihpi: Update from 4.04 to 4.06
by Takashi Iwai 11 Feb '11

11 Feb '11

At Fri, 11 Feb 2011 09:20:29 +1300, Eliot Blennerhassett wrote: > > On 11/02/11 06:52, Takashi Iwai wrote: > > At Thu, 10 Feb 2011 17:25:53 +1300, > > linux(a)audioscience.com wrote: > >> > >> This patch series updates asihpi driver from version 4.04 to 4.06 > > Thanks for the patches. > > Many of them include nice cleanups indeed. > > Thanks Takashi. > > Please also update alsa-firmware/asihpi with the contents of: > > http://audioscience.com/internet/download/drivers/released/v4/06/00/asihpi-… > > This update is necessary for driver and dsp firmware to function > correctly together. OK, done. Let me know if everything is working with the latest sound git tree. thanks, Takashi

1 0

[alsa-devel] [PATCH] asihpi: Update from 4.04 to 4.06
by linux＠audioscience.com 11 Feb '11

11 Feb '11

This patch series updates asihpi driver from version 4.04 to 4.06 Major changes: - hpi6205 adapters no longer signal interrupts on message transfer, freeing the interrupt for use by streams. This requires matching DSP firmware. - rewrote alsa driver pcm stream timer function so it works better with small buffer sizes e.g. JACK. - update alsa control names to better conform to 'standard' style. Minor changes: add low level volume mute control, allow adapters with duplicate hardware index jumpers to be discovered, fix playback start for non-mmap adapters. Cosmetic changes: removal of many unused functions and data. removal of vestigial hpi subsys pointer from hpi functions. conversion of many defines to enums, various cleanups to comment text and some badly munged identifiers. asihpi.c | 781 ++++++++--------- hpi.h | 1214 ++++++++++----------------- hpi6000.c | 299 +++--- hpi6205.c | 603 +++++-------- hpi6205.h | 7 hpi_internal.h | 1118 ++++++++++++------------- hpicmn.c | 480 ++++++---- hpicmn.h | 24 hpidebug.c | 157 --- hpidebug.h | 323 ------- hpidspcd.c | 37 hpidspcd.h | 2 hpifunc.c | 2498 ++++++++++++++++----------------------------------------- hpimsginit.c | 18 hpimsginit.h | 12 hpimsgx.c | 203 +--- hpioctl.c | 85 - hpios.h | 10 18 files changed, 2914 insertions(+), 4957 deletions(-)

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[alsa-devel] [PATCH 1/3] ASoC: Convert WM8903 bias management to use snd_soc_update_bits()
by Mark Brown 11 Feb '11

11 Feb '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 17 +++++++---------- 1 files changed, 7 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index ff1371f..9156086 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1144,15 +1144,13 @@ static int wm8903_add_widgets(struct snd_soc_codec *codec) static int wm8903_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: + break; case SND_SOC_BIAS_PREPARE: - reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0); - reg &= ~(WM8903_VMID_RES_MASK); - reg |= WM8903_VMID_RES_50K; - snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg); + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_RES_MASK, + WM8903_VMID_RES_50K); break; case SND_SOC_BIAS_STANDBY: @@ -1179,10 +1177,9 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, WM8903_CP_DYN_V); } - reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0); - reg &= ~(WM8903_VMID_RES_MASK); - reg |= WM8903_VMID_RES_250K; - snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg); + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_RES_MASK, + WM8903_VMID_RES_250K); break; case SND_SOC_BIAS_OFF: -- 1.7.2.3

2 3

[alsa-devel] [PATCH 0/2] hdspmixer: RME AES/AES32 support
by Adrian Knoth 10 Feb '11

10 Feb '11

Hi! Here's another round of patches for hdspmixer. They add support for the RME AES and RME AES32 cards (same ID, only difference is PCIe vs. PCI). Also includes some minor style fixes and formerly missing commas. Cheers Adrian Knoth (2): hdspmixer: Add RME HDSPe AES and AES32 support. hdspmixer: Style and purity fixes for name arrays. hdspmixer/src/HDSPMixerCard.cxx | 22 ++++++ hdspmixer/src/HDSPMixerOutput.cxx | 7 ++ hdspmixer/src/HDSPMixerSelector.cxx | 124 +++++++++++++++++++--------------- hdspmixer/src/HDSPMixerWindow.cxx | 10 +++ hdspmixer/src/defines.h | 1 + hdspmixer/src/hdspmixer.cxx | 4 + 6 files changed, 113 insertions(+), 55 deletions(-) -- 1.7.2.3

2 3

[alsa-devel] [PATCH] ALSA: HDA: Add subwoofer quirk for Acer Aspire 8942G
by David Henningsson 10 Feb '11

10 Feb '11

According to the reporter, node 0x15 needs to be muted for subwoofer to stop sounding. This pin is marked as unused by BIOS, so fix that. BugLink: http://bugs.launchpad.net/bugs/715877 -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[PATCH] ALSA: HDA: Add subwoofer quirk for Acer Aspire 8942G
by David Henningsson 10 Feb '11

10 Feb '11

According to the reporter, node 0x15 needs to be muted for subwoofer to stop sounding. This pin is marked as unused by BIOS, so fix that. BugLink: http://bugs.launchpad.net/bugs/715877 Cc: stable(a)kernel.org (2.6.37+) Reported-by: Hans Peter Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2571d97..089a7de 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19517,6 +19517,7 @@ static const struct alc_fixup alc662_fixups[] = { }; static struct snd_pci_quirk alc662_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), -- 1.7.1 --------------010304080202090108010307--

1 0

Re: [alsa-devel] WL1273 FM Radio driver...
by Matti J. Aaltonen 10 Feb '11

10 Feb '11

Hello. Mark Brown wrote: > On Wed, Feb 02, 2011 at 01:35:01PM -0200, Mauro > Carvalho Chehab wrote: > > [Reflowed into 80 columns.] >> My concerns is that the V4L2-specific part of the code should be at >> drivers/media. I prefer that the specific MFD I/O part to be at >> drivers/mfd, just like the other drivers. > > Currently that's not the case - the I/O functionality is not in any > meaningful sense included in the MFD, it's provided by the V4L portion. I've been away for two and a half weeks so I haven't been able to comment... But before I start to make changes, I'd still like to ask for a comment on my original plan, which was to have the I/O functions in the MFD driver and also have there things like interrupt handling etc. My vision was that the MFD part would have the application logic and the child drivers would be just true interfaces to the core functionality, because I kind of saw the children to be of equal importance and because the codec and the v4l2 driver share some controls like for example the volume control. If you'd care to take a look an earlier version of the MFD driver here: http://article.gmane.org/gmane.linux.drivers.video-input-infrastructure/236… So the question is if I put only the I/O stuff into the MFD driver or can I have the other application logic there as well? Thanks, Matti

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[alsa-devel] list *all* PCMs
by Aleksandar Kostadinov 10 Feb '11

10 Feb '11

Hello, I searched through the list archive and I can't find a way to list all PCM devices. aplay/arecord -L show only these that have hint show on. Is there such command that lists all PCMs? While on it, it is also hard to obtain a list of capabilities of a device. For example I don't see a way user can obtain the supported sample rates by a sound device. I have found a program (see it attached to http://www.spinics.net/linux/fedora/alsa-user/msg07230.html) that seems to work fine. Could that program or similar functionality be implemented in arecord/aplay so user can easily check device capabilities? Thank you, Aleksandar P.S. please cc me in replies because I'm not a subscriber

3 3

[alsa-devel] [PATCH] ALSA: usb - prevent kernel panic on disconnect
by Pierre-Louis Bossart 10 Feb '11

10 Feb '11

Fix 100% reproducible kernel panic when a USB audio device is disconnected during playback. When the PCM substream is NULL, any dereference causes a kernel panic. This isn't a recent problem, the crash happened with a 2.6.35 kernel as well Note that this is only a work-around, it does not address the root cause of this inconsistency between urbs and PCM states. The dmesg below shows two calls to snd_urb_complete, the substream is NULL and the state is either running or stopped. This doesn't make any sense. Thanks to Sarah Sharp <sarah.a.sharp(a)linux.intel.com> for her help. usb 6-1: USB disconnect, address 16 ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:186: NULL substream (subs->running 1) <- How is this possible? ALSA urb.c:186: NULL substream (subs->running 0) Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)intel.com> --- usb/urb.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/usb/urb.c b/usb/urb.c index e184349..1a264c2 100644 --- a/usb/urb.c +++ b/usb/urb.c @@ -170,6 +170,12 @@ static void snd_complete_urb(struct urb *urb) struct snd_pcm_substream *substream = ctx->subs->pcm_substream; int err = 0; + if (substream == NULL) { + snd_printd(KERN_ERR "NULL substream (subs->running %d)\n", subs->running); + clear_bit(ctx->index, &subs->active_mask); + return; + } + if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) || !subs->running || /* can be stopped during retire callback */ (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 || @@ -193,6 +199,12 @@ static void snd_complete_sync_urb(struct urb *urb) struct snd_pcm_substream *substream = ctx->subs->pcm_substream; int err = 0; + if (substream == NULL) { + snd_printd(KERN_ERR "NULL substream (subs->running %d)\n", subs->running); + clear_bit(ctx->index + 16, &subs->active_mask); + return; + } + if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) || !subs->running || /* can be stopped during retire callback */ (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 || -- 1.7.4

2 1

[alsa-devel] [PATCH] ASoC: add support for multiple jack types
by Koul, Vinod 10 Feb '11

10 Feb '11

From: Vinod Koul <vinod.koul(a)intel.com> This patch adds soc-jack support for adding voltage zones and for detecting jack type Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- include/sound/soc.h | 23 +++++++++++++++++++++++ sound/soc/soc-jack.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 69 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4b6c0a8..4ccf1e4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -234,6 +234,7 @@ struct snd_soc_codec; struct snd_soc_codec_driver; struct soc_enum; struct snd_soc_jack; +struct snd_soc_jack_zone; struct snd_soc_jack_pin; struct snd_soc_cache_ops; #include <sound/soc-dapm.h> @@ -307,6 +308,9 @@ void snd_soc_jack_notifier_register(struct snd_soc_jack *jack, struct notifier_block *nb); void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack, struct notifier_block *nb); +int snd_soc_jack_add_zones(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_zone *zones); +int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage); #ifdef CONFIG_GPIOLIB int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); @@ -407,6 +411,24 @@ struct snd_soc_jack_pin { }; /** + * struct snd_soc_jack_zone - Describes voltage zones of jack detection + * + * @min_mv: start voltage in mv + * @max_mv: end voltage in mv + * @jack_type: type of jack that is expected for this voltage + * @debounce_time: debounce_time for jack, codec driver should wait for this + * duration before reading the adc for voltages + * @:list: list container + */ +struct snd_soc_jack_zone { + unsigned int min_mv; + unsigned int max_mv; + unsigned int jack_type; + unsigned int debounce_time; + struct list_head list; +}; + +/** * struct snd_soc_jack_gpio - Describes a gpio pin for jack detection * * @gpio: gpio number @@ -435,6 +457,7 @@ struct snd_soc_jack { struct list_head pins; int status; struct blocking_notifier_head notifier; + struct list_head jack_zones; }; /* SoC PCM stream information */ diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ac5a5bc..99dbaf7 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -37,6 +37,7 @@ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, { jack->codec = codec; INIT_LIST_HEAD(&jack->pins); + INIT_LIST_HEAD(&jack->jack_zones); BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); return snd_jack_new(codec->card->snd_card, id, type, &jack->jack); @@ -112,6 +113,51 @@ out: EXPORT_SYMBOL_GPL(snd_soc_jack_report); /** + * snd_soc_jack_add_zones - Associate voltage zones with jack + * + * @jack: ASoC jack + * @count: Number of zones + * @zone: Array of zones + * + * After this function has been called the zones specified in the + * array will be associated with the jack. + */ +int snd_soc_jack_add_zones(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_zone *zones) +{ + int i; + + for (i = 0; i < count; i++) { + INIT_LIST_HEAD(&zones[i].list); + list_add(&(zones[i].list), &jack->jack_zones); + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_add_zones); + +/** + * snd_soc_jack_get_type - Based on the mic bias value, this function returns + * the type of jack from the zones delcared in the jack type + * + * @micbias_voltage: mic bias voltage at adc channel when jack is plugged in + * + * Based on the mic bias value passed, this function helps identify + * the type of jack from the already delcared jack zones + */ +int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage) +{ + struct snd_soc_jack_zone *zone; + + list_for_each_entry(zone, &jack->jack_zones, list) { + if (micbias_voltage >= zone->min_mv && + micbias_voltage < zone->max_mv) + return zone->jack_type; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_get_type); + +/** * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack * * @jack: ASoC jack -- 1.7.2.3

3 2

[alsa-devel] [PATCH 0/3] add support for jack detection in mid-x86
by Koul, Vinod 10 Feb '11

10 Feb '11

From: Vinod Koul <vinod.koul(a)intel.com> This patch set adds support for jack detection in mid-x6 machine and TI sn95031 codec drivers. This also adds support to read mic bias values thru ADC This version fixes namespace issue in patch1, and removes gpio bits in machine driver pending gpiolib support, which will be added separately Vinod Koul (3): ASoC: sn95031: Add jack support in the codec ASoC: mfld_machine: Add support for jack detection ASoC: sn95031: Add support for reading mic bias sound/soc/codecs/sn95031.c | 183 +++++++++++++++++++++++++++++++++- sound/soc/codecs/sn95031.h | 33 ++++++ sound/soc/mid-x86/mfld_machine.c | 203 +++++++++++++++++++++++++++++++++----- 3 files changed, 388 insertions(+), 31 deletions(-) -- 1.7.2.3

3 5

[alsa-devel] [PATCH 1/3] ASoC: Davinci: Use resource_size() helper function
by Vaibhav Bedia 10 Feb '11

10 Feb '11

This patch modifies the Davinci i2s and mcasp drivers to make use of the resource_size() helper function for readability. Signed-off-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> --- sound/soc/davinci/davinci-i2s.c | 6 +++--- sound/soc/davinci/davinci-mcasp.c | 6 +++--- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 9e0e565..f8c8487 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -658,7 +658,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) return -ENODEV; } - ioarea = request_mem_region(mem->start, (mem->end - mem->start) + 1, + ioarea = request_mem_region(mem->start, resource_size(mem), pdev->name); if (!ioarea) { dev_err(&pdev->dev, "McBSP region already claimed\n"); @@ -731,7 +731,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) err_free_mem: kfree(dev); err_release_region: - release_mem_region(mem->start, (mem->end - mem->start) + 1); + release_mem_region(mem->start, resource_size(mem)); return ret; } @@ -747,7 +747,7 @@ static int davinci_i2s_remove(struct platform_device *pdev) dev->clk = NULL; kfree(dev); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(mem->start, (mem->end - mem->start) + 1); + release_mem_region(mem->start, resource_size(mem)); return 0; } diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index fb55d2c..9b5d5ca 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -868,7 +868,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } ioarea = request_mem_region(mem->start, - (mem->end - mem->start) + 1, pdev->name); + resource_size(mem), pdev->name); if (!ioarea) { dev_err(&pdev->dev, "Audio region already claimed\n"); ret = -EBUSY; @@ -933,7 +933,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return 0; err_release_region: - release_mem_region(mem->start, (mem->end - mem->start) + 1); + release_mem_region(mem->start, resource_size(mem)); err_release_data: kfree(dev); @@ -951,7 +951,7 @@ static int davinci_mcasp_remove(struct platform_device *pdev) dev->clk = NULL; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(mem->start, (mem->end - mem->start) + 1); + release_mem_region(mem->start, resource_size(mem)); kfree(dev); -- 1.6.2.4

3 4

[alsa-devel] [PATCH 1/3] ASoC: Remove redundant -codec from WM8903 driver name
by Mark Brown 09 Feb '11

09 Feb '11

It causes noisy -codecs to appear in things like .codec_name. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 2 +- sound/soc/tegra/harmony.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 27cd8f5..04c3256 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2134,7 +2134,7 @@ MODULE_DEVICE_TABLE(i2c, wm8903_i2c_id); static struct i2c_driver wm8903_i2c_driver = { .driver = { - .name = "wm8903-codec", + .name = "wm8903", .owner = THIS_MODULE, }, .probe = wm8903_i2c_probe, diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index 11e2cb8..d702abe 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -189,7 +189,7 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link harmony_wm8903_dai = { .name = "WM8903", .stream_name = "WM8903 PCM", - .codec_name = "wm8903-codec.0-001a", + .codec_name = "wm8903.0-001a", .platform_name = "tegra-pcm-audio", .cpu_dai_name = "tegra-i2s.0", .codec_dai_name = "wm8903-hifi", -- 1.7.2.3

3 5

[alsa-devel] [PATCH 1/3] ASoC: Convert WM8903 to use PGA_S for output stage enables
by Mark Brown 09 Feb '11

09 Feb '11

This simplfies the code and slightly reduces the startup time. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 180 +++++++++++++++------------------------------ 1 files changed, 60 insertions(+), 120 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b20611b..a5b2d07 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -251,6 +251,8 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re case WM8903_REVISION_NUMBER: case WM8903_INTERRUPT_STATUS_1: case WM8903_WRITE_SEQUENCER_4: + case WM8903_POWER_MANAGEMENT_3: + case WM8903_POWER_MANAGEMENT_2: return 1; default: @@ -309,11 +311,6 @@ static void wm8903_reset(struct snd_soc_codec *codec) sizeof(wm8903_reg_defaults)); } -#define WM8903_OUTPUT_SHORT 0x8 -#define WM8903_OUTPUT_OUT 0x4 -#define WM8903_OUTPUT_INT 0x2 -#define WM8903_OUTPUT_IN 0x1 - static int wm8903_cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -324,99 +321,6 @@ static int wm8903_cp_event(struct snd_soc_dapm_widget *w, } /* - * Event for headphone and line out amplifier power changes. Special - * power up/down sequences are required in order to maximise pop/click - * performance. - */ -static int wm8903_output_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - u16 val; - u16 reg; - u16 dcs_reg; - u16 dcs_bit; - int shift; - - switch (w->reg) { - case WM8903_POWER_MANAGEMENT_2: - reg = WM8903_ANALOGUE_HP_0; - dcs_bit = 0 + w->shift; - break; - case WM8903_POWER_MANAGEMENT_3: - reg = WM8903_ANALOGUE_LINEOUT_0; - dcs_bit = 2 + w->shift; - break; - default: - BUG(); - return -EINVAL; /* Spurious warning from some compilers */ - } - - switch (w->shift) { - case 0: - shift = 0; - break; - case 1: - shift = 4; - break; - default: - BUG(); - return -EINVAL; /* Spurious warning from some compilers */ - } - - if (event & SND_SOC_DAPM_PRE_PMU) { - val = snd_soc_read(codec, reg); - - /* Short the output */ - val &= ~(WM8903_OUTPUT_SHORT << shift); - snd_soc_write(codec, reg, val); - } - - if (event & SND_SOC_DAPM_POST_PMU) { - val = snd_soc_read(codec, reg); - - val |= (WM8903_OUTPUT_IN << shift); - snd_soc_write(codec, reg, val); - - val |= (WM8903_OUTPUT_INT << shift); - snd_soc_write(codec, reg, val); - - /* Turn on the output ENA_OUTP */ - val |= (WM8903_OUTPUT_OUT << shift); - snd_soc_write(codec, reg, val); - - /* Enable the DC servo */ - dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0); - dcs_reg |= dcs_bit; - snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg); - - /* Remove the short */ - val |= (WM8903_OUTPUT_SHORT << shift); - snd_soc_write(codec, reg, val); - } - - if (event & SND_SOC_DAPM_PRE_PMD) { - val = snd_soc_read(codec, reg); - - /* Short the output */ - val &= ~(WM8903_OUTPUT_SHORT << shift); - snd_soc_write(codec, reg, val); - - /* Disable the DC servo */ - dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0); - dcs_reg &= ~dcs_bit; - snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg); - - /* Then disable the intermediate and output stages */ - val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT | - WM8903_OUTPUT_IN) << shift); - snd_soc_write(codec, reg, val); - } - - return 0; -} - -/* * When used with DAC outputs only the WM8903 charge pump supports * operation in class W mode, providing very low power consumption * when used with digital sources. Enable and disable this mode @@ -918,23 +822,40 @@ SND_SOC_DAPM_MIXER("Left Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 1, 0, SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), -SND_SOC_DAPM_PGA_E("Left Headphone Output PGA", WM8903_POWER_MANAGEMENT_2, - 1, 0, NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_PGA_E("Right Headphone Output PGA", WM8903_POWER_MANAGEMENT_2, - 0, 0, NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), - -SND_SOC_DAPM_PGA_E("Left Line Output PGA", WM8903_POWER_MANAGEMENT_3, 1, 0, - NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_PGA_E("Right Line Output PGA", WM8903_POWER_MANAGEMENT_3, 0, 0, - NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0, + 4, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0, + 0, 0, NULL, 0), + +SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0, + NULL, 0), + +SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0), + +SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0, + NULL, 0), + +SND_SOC_DAPM_PGA_S("HPL_DCS", 3, WM8903_DC_SERVO_0, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_DCS", 3, WM8903_DC_SERVO_0, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTL_DCS", 3, WM8903_DC_SERVO_0, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_DCS", 3, WM8903_DC_SERVO_0, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0, NULL, 0), @@ -1050,11 +971,30 @@ static const struct snd_soc_dapm_route intercon[] = { { "Left Speaker PGA", NULL, "Left Speaker Mixer" }, { "Right Speaker PGA", NULL, "Right Speaker Mixer" }, - { "HPOUTL", NULL, "Left Headphone Output PGA" }, - { "HPOUTR", NULL, "Right Headphone Output PGA" }, - - { "LINEOUTL", NULL, "Left Line Output PGA" }, - { "LINEOUTR", NULL, "Right Line Output PGA" }, + { "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" }, + { "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" }, + { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" }, + { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" }, + + { "HPL_DCS", NULL, "HPL_ENA_DLY" }, + { "HPR_DCS", NULL, "HPR_ENA_DLY" }, + { "LINEOUTL_DCS", NULL, "LINEOUTL_ENA_DLY" }, + { "LINEOUTR_DCS", NULL, "LINEOUTR_ENA_DLY" }, + + { "HPL_ENA_OUTP", NULL, "HPL_DCS" }, + { "HPR_ENA_OUTP", NULL, "HPR_DCS" }, + { "LINEOUTL_ENA_OUTP", NULL, "LINEOUTL_DCS" }, + { "LINEOUTR_ENA_OUTP", NULL, "LINEOUTR_DCS" }, + + { "HPL_RMV_SHORT", NULL, "HPL_ENA_OUTP" }, + { "HPR_RMV_SHORT", NULL, "HPR_ENA_OUTP" }, + { "LINEOUTL_RMV_SHORT", NULL, "LINEOUTL_ENA_OUTP" }, + { "LINEOUTR_RMV_SHORT", NULL, "LINEOUTR_ENA_OUTP" }, + + { "HPOUTL", NULL, "HPL_RMV_SHORT" }, + { "HPOUTR", NULL, "HPR_RMV_SHORT" }, + { "LINEOUTL", NULL, "LINEOUTL_RMV_SHORT" }, + { "LINEOUTR", NULL, "LINEOUTR_RMV_SHORT" }, { "LOP", NULL, "Left Speaker PGA" }, { "LON", NULL, "Left Speaker PGA" }, -- 1.7.2.3

2 5

[alsa-devel] [PATCH] ASoC: Sync initial widget state with hardware
by Mark Brown 09 Feb '11

09 Feb '11

ASoC generally uses the register defaults for everything, but in some cases the hardware will default to enabling some of the DAPM widgets (clocks for example). Ensure that DAPM knows about the actual widget state at initialisation by reading the enable bits after instantiating the widgets so they don't get left enabled needlessly. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d0342aa..b30eda6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1664,6 +1664,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *w; + unsigned int val; list_for_each_entry(w, &dapm->card->widgets, list) { @@ -1712,6 +1713,18 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_post: break; } + + /* Read the initial power state from the device */ + if (w->reg >= 0) { + val = snd_soc_read(w->codec, w->reg); + val &= 1 << w->shift; + if (w->invert) + val = !val; + + if (val) + w->power = 1; + } + w->new = 1; } -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Actively manage WM8903 DC servo configuration
by Mark Brown 09 Feb '11

09 Feb '11

Explicitly cache the DC servo offsets for digital paths in the driver, allowing them to be preserved over suspend and resume, and ensure that we recalibrate analogue outputs paths when they are in use so that we cover any changes in the input offset. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 123 +++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm8903.h | 8 +++ 2 files changed, 127 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 0ade99d..ff1371f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -224,6 +224,9 @@ struct wm8903_priv { int fs; int deemph; + int dcs_pending; + int dcs_cache[4]; + /* Reference count */ int class_w_users; @@ -253,6 +256,10 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re case WM8903_WRITE_SEQUENCER_4: case WM8903_POWER_MANAGEMENT_3: case WM8903_POWER_MANAGEMENT_2: + case WM8903_DC_SERVO_READBACK_1: + case WM8903_DC_SERVO_READBACK_2: + case WM8903_DC_SERVO_READBACK_3: + case WM8903_DC_SERVO_READBACK_4: return 1; default: @@ -320,6 +327,103 @@ static int wm8903_cp_event(struct snd_soc_dapm_widget *w, return 0; } +static int wm8903_dcs_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + wm8903->dcs_pending |= 1 << w->shift; + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, WM8903_DC_SERVO_0, + 1 << w->shift, 0); + break; + } + + return 0; +} + +#define WM8903_DCS_MODE_WRITE_STOP 0 +#define WM8903_DCS_MODE_START_STOP 2 + +static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, + enum snd_soc_dapm_type event, int subseq) +{ + struct snd_soc_codec *codec = container_of(dapm, + struct snd_soc_codec, dapm); + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + int dcs_mode = WM8903_DCS_MODE_WRITE_STOP; + int i, val; + + /* Complete any pending DC servo starts */ + if (wm8903->dcs_pending) { + dev_dbg(codec->dev, "Starting DC servo for %x\n", + wm8903->dcs_pending); + + /* If we've no cached values then we need to do startup */ + for (i = 0; i < ARRAY_SIZE(wm8903->dcs_cache); i++) { + if (!(wm8903->dcs_pending & (1 << i))) + continue; + + if (wm8903->dcs_cache[i]) { + dev_dbg(codec->dev, + "Restore DC servo %d value %x\n", + 3 - i, wm8903->dcs_cache[i]); + + snd_soc_write(codec, WM8903_DC_SERVO_4 + i, + wm8903->dcs_cache[i] & 0xff); + } else { + dev_dbg(codec->dev, + "Calibrate DC servo %d\n", 3 - i); + dcs_mode = WM8903_DCS_MODE_START_STOP; + } + } + + /* Don't trust the cache for analogue */ + if (wm8903->class_w_users) + dcs_mode = WM8903_DCS_MODE_START_STOP; + + snd_soc_update_bits(codec, WM8903_DC_SERVO_2, + WM8903_DCS_MODE_MASK, dcs_mode); + + snd_soc_update_bits(codec, WM8903_DC_SERVO_0, + WM8903_DCS_ENA_MASK, wm8903->dcs_pending); + + switch (dcs_mode) { + case WM8903_DCS_MODE_WRITE_STOP: + break; + + case WM8903_DCS_MODE_START_STOP: + msleep(270); + + /* Cache the measured offsets for digital */ + if (wm8903->class_w_users) + break; + + for (i = 0; i < ARRAY_SIZE(wm8903->dcs_cache); i++) { + if (!(wm8903->dcs_pending & (1 << i))) + continue; + + val = snd_soc_read(codec, + WM8903_DC_SERVO_READBACK_1 + i); + dev_dbg(codec->dev, "DC servo %d: %x\n", + 3 - i, val); + wm8903->dcs_cache[i] = val; + } + break; + + default: + pr_warn("DCS mode %d delay not set\n", dcs_mode); + break; + } + + wm8903->dcs_pending = 0; + } +} + /* * When used with DAC outputs only the WM8903 charge pump supports * operation in class W mode, providing very low power consumption @@ -852,10 +956,15 @@ SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0, SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPL_DCS", 3, WM8903_DC_SERVO_0, 3, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPR_DCS", 3, WM8903_DC_SERVO_0, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTL_DCS", 3, WM8903_DC_SERVO_0, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTR_DCS", 3, WM8903_DC_SERVO_0, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_DCS", 3, SND_SOC_NOPM, 3, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("HPR_DCS", 3, SND_SOC_NOPM, 2, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("LINEOUTL_DCS", 3, SND_SOC_NOPM, 1, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("LINEOUTR_DCS", 3, SND_SOC_NOPM, 0, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0, NULL, 0), @@ -984,6 +1093,11 @@ static const struct snd_soc_dapm_route intercon[] = { { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" }, { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" }, + { "HPL_DCS", NULL, "DCS Master" }, + { "HPR_DCS", NULL, "DCS Master" }, + { "LINEOUTL_DCS", NULL, "DCS Master" }, + { "LINEOUTR_DCS", NULL, "DCS Master" }, + { "HPL_DCS", NULL, "HPL_ENA_DLY" }, { "HPR_DCS", NULL, "HPR_ENA_DLY" }, { "LINEOUTL_DCS", NULL, "LINEOUTL_ENA_DLY" }, @@ -2103,6 +2217,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8903 = { .reg_word_size = sizeof(u16), .reg_cache_default = wm8903_reg_defaults, .volatile_register = wm8903_volatile_register, + .seq_notifier = wm8903_seq_notifier, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h index a6d63e4..7d96646 100644 --- a/sound/soc/codecs/wm8903.h +++ b/sound/soc/codecs/wm8903.h @@ -81,6 +81,14 @@ extern int wm8903_mic_detect(struct snd_soc_codec *codec, #define WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0 0x41 #define WM8903_DC_SERVO_0 0x43 #define WM8903_DC_SERVO_2 0x45 +#define WM8903_DC_SERVO_4 0x47 +#define WM8903_DC_SERVO_5 0x48 +#define WM8903_DC_SERVO_6 0x49 +#define WM8903_DC_SERVO_7 0x4A +#define WM8903_DC_SERVO_READBACK_1 0x51 +#define WM8903_DC_SERVO_READBACK_2 0x52 +#define WM8903_DC_SERVO_READBACK_3 0x53 +#define WM8903_DC_SERVO_READBACK_4 0x54 #define WM8903_ANALOGUE_HP_0 0x5A #define WM8903_ANALOGUE_LINEOUT_0 0x5E #define WM8903_CHARGE_PUMP_0 0x62 -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Sync initial widget state with hardware
by Mark Brown 09 Feb '11

09 Feb '11

ASoC generally uses the register defaults for everything, but in some cases the hardware will default to enabling some of the DAPM widgets (clocks for example). Ensure that DAPM knows about the actual widget state at initialisation by reading the enable bits after instantiating the widgets so they don't get left enabled needlessly. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d0342aa..b4e8755 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1664,6 +1664,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *w; + unsigned int val; list_for_each_entry(w, &dapm->card->widgets, list) { @@ -1712,6 +1713,18 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_post: break; } + + /* Read the initial power state from the device */ + if (w->reg >= 0) { + val = snd_soc_read(w->codec, w->reg); + val &= 1 << w->shift; + if (w->invert) + val = !val; + + if (val) + w->active = 1; + } + w->new = 1; } -- 1.7.2.3

1 1

[alsa-devel] [PATCH 0/3] add support for jack detection in mid-x86
by Koul, Vinod 09 Feb '11

09 Feb '11

From: Vinod Koul <vinod.koul(a)intel.com> This patch set adds support for jack detection in mid-x6 machine and TI sn95031 codec drivers. This also adds support to read mic bias values thru ADC The comments from previous submission are addressed and patches split Vinod Koul (3): ASoC: sn95031: Add jack support in the codec ASoC: mfld_machine: Add support for jack detection ASoC: sn95031: Add support for reading mic bias sound/soc/codecs/sn95031.c | 184 ++++++++++++++++++++++++++++++++- sound/soc/codecs/sn95031.h | 35 ++++++ sound/soc/mid-x86/mfld_machine.c | 211 +++++++++++++++++++++++++++++++++----- 3 files changed, 399 insertions(+), 31 deletions(-) -- 1.7.2.3

2 12

[alsa-devel] [PATCH 1/6] ASoC: Tegra: Harmony: Add headphone jack detection
by Stephen Warren 09 Feb '11

09 Feb '11

Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/harmony.c | 30 +++++++++++++++++++++++++++++- 1 files changed, 29 insertions(+), 1 deletions(-) diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index 1b80c4f..26ab4fd 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -38,6 +38,7 @@ #include <mach/harmony_audio.h> #include <sound/core.h> +#include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -123,6 +124,24 @@ static struct snd_soc_ops harmony_asoc_ops = { .hw_params = harmony_asoc_hw_params, }; +static struct snd_soc_jack harmony_hp_jack; + +static struct snd_soc_jack_pin harmony_hp_jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack_gpio harmony_hp_jack_gpios[] = { + { + .name = "headphone detect", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, + .invert = 1, + } +}; + static int harmony_event_int_spk(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { @@ -178,11 +197,20 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_add_routes(dapm, harmony_audio_map, ARRAY_SIZE(harmony_audio_map)); - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Int Spk"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_sync(dapm); + harmony_hp_jack_gpios[0].gpio = pdata->gpio_hp_det; + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &harmony_hp_jack); + snd_soc_jack_add_pins(&harmony_hp_jack, + ARRAY_SIZE(harmony_hp_jack_pins), + harmony_hp_jack_pins); + snd_soc_jack_add_gpios(&harmony_hp_jack, + ARRAY_SIZE(harmony_hp_jack_gpios), + harmony_hp_jack_gpios); + return 0; } -- 1.7.1

3 25

[alsa-devel] [PATCH] alsa-utils: Spelling fixes
by David Henningsson 09 Feb '11

09 Feb '11

While looking into the Debian packaging of alsa-utils, I noticed a few patches which fix spelling errors. I'm attaching them here. If the patches attached are good for just applying, please do so or report back if there's anything I can do to get them in upstream. Thanks! -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

1 0

Re: [alsa-devel] ALC861-VD bad recording issue
by Raymond Yau 09 Feb '11

09 Feb '11

> This is the output of the alsa script to ge information about my system and the driver http://www.alsa-project.org/db/?f=9871b38d99bdd4cc91e398b60eb31361eb305395 Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Input Source", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b if Input source is using [Audio Mixer], the driver may need to unmute only the seleted source and the other sources may need to be muted

1 0

Re: [alsa-devel] MIDI playback not keeping steady time with recent kernels
by Clemens Ladisch 09 Feb '11

09 Feb '11

Arthur Marsh wrote: >>> Arthur Marsh wrote, on 05/02/11 17:30: >>>> MIDI playback either through xmms or aplaymidi won't keep regular time, >>>> sounding weird going faster and slower under any kind of system load >>>> except when nothing else is running. > > I've tried setting "options snd-timer timer_limit=x" where x is anywhere > from 5 down to 0, and seeing either: > > $ cat /proc/asound/seq/timer > Timer for queue 0 : HR timer > Period time : 0.004000250 > Skew : 65536 / 65536 > > or "system timer" in place of "HR timer". > > In all cases, under sufficient load (which might be just running > aptitude -u), the tempo of the MIDI file play-back slows right down. The ALSA interfaces of both the system timer and the HR timer do not handle delayed interrupts correctly. Please try the patch below, and if it fixes the HR timer, tell me if you're OK with the published tag Reported-and-tested-by: Arthur Marsh <arthur.marsh(a)internode.on.net>. Regards, Clemens --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -45,12 +45,13 @@ static enum hrtimer_restart snd_hrtimer_ { struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt); struct snd_timer *t = stime->timer; + unsigned long oruns; if (!atomic_read(&stime->running)) return HRTIMER_NORESTART; - hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution)); - snd_timer_interrupt(stime->timer, t->sticks); + oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution)); + snd_timer_interrupt(stime->timer, t->sticks * oruns); if (!atomic_read(&stime->running)) return HRTIMER_NORESTART;

2 3

[alsa-devel] [PATCH 0/2] LaCie Firewire Speakers/Griffin FireWave support
by Clemens Ladisch 08 Feb '11

08 Feb '11

This driver version is, I hope, finished enough so that I can commit it soon. The FireWave's low-frequency processing makes me believe that this device is prepared to accept two- and four-channel data, so I've enabled this in the driver; but I'd still like to actually know if it works. (If it does, adding support for the AC-3 decoding will be possible too.) I haven't yet looked into volume controls. Jay, are you OK with "Tested-by: Jay Fenlason <fenlason(a)redhat.com>"? Stefan, if those firewire core bits are worth your notice :) , please give an Acked-by for them. Regards, Clemens

2 10

[alsa-devel] ASoC updates for 2.6.39
by Mark Brown 08 Feb '11

08 Feb '11

The following changes since commit 273de37655369e57919f19e157fafb2b4b1e4472: Merge branch 'for-2.6.39' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into for-2.6.39 (2011-02-01 14:55:10 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Alexander Sverdlin (1): ASoC: CS4271: Move Chip Select control out of the CODEC code. Dan Carpenter (1): ASoC: soc-cache: dereferencing before checking Dimitris Papastamos (1): ASoC: soc-core: Support debugfs entries larger than PAGE_SIZE bytes Janusz Krzysztofik (1): ASoC: fill in snd_soc_pcm_runtime.card before calling snd_soc_dai_link.init() Lars-Peter Clausen (1): ASoC: codecs: wm8753: Fix DAI mode switching Mark Brown (6): ASoC: Update PM ifdefs for exported suspend/resume Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Sort ALC5623 in Kconfig and Makefile ASoC: Create an AIF1ADCDAT signal widget to match AIF2 ASoC: Improve WM8994 digital power sequencing Merge branch 'for-2.6.38' into for-2.6.39 Mika Westerberg (1): ASoC: ep93xx-ac97: remove extra empty line Sven Neumann (1): ASoC: PXA: formatting Vaibhav Bedia (1): asoc: davinci: da830/omap-l137: correct cpu_dai_name Vinod Koul (1): ASoC: mid-x86: Fix dependency on intel_sst driver include/sound/cs4271.h | 1 - sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/cs4271.c | 22 +--- sound/soc/codecs/wm8753.c | 296 ++++++++++++++++----------------------- sound/soc/codecs/wm8994.c | 26 +++- sound/soc/davinci/davinci-evm.c | 18 ++- sound/soc/ep93xx/ep93xx-ac97.c | 1 - sound/soc/mid-x86/Kconfig | 2 +- sound/soc/pxa/raumfeld.c | 16 +- sound/soc/soc-cache.c | 25 ++-- sound/soc/soc-core.c | 114 ++++++++++----- 11 files changed, 258 insertions(+), 265 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 08 Feb '11

08 Feb '11

The following changes since commit f019ee5feb344ff0b22b58df4568676295aae14f: ASoC: CX20442: fix NULL pointer dereference (2011-02-01 14:13:56 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Janusz Krzysztofik (1): ASoC: fill in snd_soc_pcm_runtime.card before calling snd_soc_dai_link.init() Mark Brown (2): ASoC: Create an AIF1ADCDAT signal widget to match AIF2 ASoC: Improve WM8994 digital power sequencing Vaibhav Bedia (1): asoc: davinci: da830/omap-l137: correct cpu_dai_name sound/soc/codecs/wm8994.c | 26 ++++++++++++++++++++++---- sound/soc/davinci/davinci-evm.c | 18 +++++++++++++++--- sound/soc/soc-core.c | 2 +- 3 files changed, 38 insertions(+), 8 deletions(-)

2 1

Re: [alsa-devel] [Alsa-user] hda intel alc889 broken since 2.6.36.2
by Paul Menzel 08 Feb '11

08 Feb '11

Dear Andrea, please do just send plain text messages to lists [2]. Also I am subscribed to the list, so you do not need to CC me. Am Dienstag, den 08.02.2011, 08:36 +0100 schrieb Andrea Rizzolo: > On Tue, Feb 8, 2011 at 8:29 AM, Paul Menzel wrote: > > Am Montag, den 07.02.2011, 18:12 +0100 schrieb Andrea Rizzolo: > > > Hi, my hda intel card is broken since kernel 2.6.36.2. it looks correctly > > > recognized, but it plays no sound at all. Same problem with 2.6.36.3 and > > > 2.6.37, instead it works correctly with kernels up to 2.6.36.1 (same > > > config). > > > my /proc/asound/cards: > > > 0 [Intel ]: HDA-Intel - HDA Intel > > > HDA Intel at 0xf0420000 irq 22 > > > > please read the Wiki pages [1] and attach the output of `alsa-info.sh` > > from the working setup and the non-working setup. > > > > What GNU/Linux distribution do you use? According to your report in the BTS, you are using Arch Linux. > Sorry, you are right. Here I provide full details: > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5267 Unfortunately most ALSA developers do not use the BTS but the documentation is not updated. :( I am attaching the differences to this message. With Linux kernel 2.6.37 your Webcam seems to be detected. Could you try if the volumes are up and the correct outputs are used. I am moving this thread to alsa-devel [3][4] since I have the experience that you will more likely get an answer there. Thanks, Paul > > [1] http://alsa-project.org/main/index.php/Help_To_Debug [2] http://www2.cit.cornell.edu/computer/elist/text/google.html [3] http://alsa-project.org/main/index.php/Mailing-lists [4] http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

1 0

[alsa-devel] ALSA: Reasonable buffer size / where should it be implemented?
by Knut Petersen 08 Feb '11

08 Feb '11

I use a RME Digi 96 PAD audio card, and I do have buffer overrun/underrun problems if I use the standard rme96c linux driver. I need a simple recording machine. It should not fail if e.g. cron starts updatedb or if I start a make -j 15 icecream compile job and decide to surf the internet while I record the digital satellite radio 48kHz stream. There is a lot of information about xrun problems, but somehow that information either does not help to prevent xruns on my system, is outdated, or asks for system changes I do not accept. No, JACK does not help. No, I do not need low latency. No, I don't want to switch to rt kernels. No, I don't want to use an audio PC without X, without cron, network, etc. The hardware provides independent 64k ringbuffers for capture and playback, that's not more than 85msec for a 96 kHz / 2 channel / 32 bit setup or ADAT. That's simply not enough for reliable operation. My private solution is a rme96.c that kmallocs 4 MB software buffers for capture and playback, data transfer between software and hardware buffer in the interrupt service routine. That does efficiently prevent xruns even on a really loaded system. But I don't know if that is the right way to go. Wouldn't it be better if there would be an (optional) software buffer one layer above the hardware driver? That would increase reliability for all audio hardware with insufficient hardware buffers. Is there any other audio hardware with similar small buffers? Has somebody already written an "extended alsa buffer" patch? Did I miss something? To put it into one question: How much buffer should be provided by an alsa hardware driver module? cu, Knut

2 2

[alsa-devel] [PATCH v8 00/12] Media controller (core and V4L2)
by Laurent Pinchart 08 Feb '11

08 Feb '11

Hi everybody, Here is the eighth version of the media controller core and V4L2 patches. Quick reminder for those who missed the previous version. let me quote the documentation (Documentation/DocBook/v4l/media-controller.xml). "Discovering a [media] device internal topology, and configuring it at runtime, is one of the goals of the media controller API. To achieve this, hardware devices are modelled as an oriented graph of building blocks called entities connected through pads." The code has been extensively reviewed by the V4L community, and this version is the first one to incorporate comments from the ALSA community (big thanks to Mark Brown and Clemens Ladisch). Two issues are not fully addressed yet, namely power management (I need to discuss this some more with the ALSA developers to really understand their requirements) and entities type names. I'm still posting this for review, as other developers have showed interest in commenting on the code. I want to emphasize once again that the media controller API does not replace the V4L, DVB or ALSA APIs. It complements them. The first user of the media controller API is the OMAP3 ISP driver. You can find it (as well as these patches and other V4L-specific patches) in a git tree at http://git.linuxtv.org/pinchartl/media.git (media-0005-omap3isp branch). The OMAP3 ISP driver patches are regularly posted for review on the linux-media list. Laurent Pinchart (11): media: Media device node support media: Media device media: Entities, pads and links media: Entity use count media: Media device information query media: Entities, pads and links enumeration media: Links setup media: Pipelines and media streams v4l: Add a media_device pointer to the v4l2_device structure v4l: Make video_device inherit from media_entity v4l: Make v4l2_subdev inherit from media_entity Sakari Ailus (1): media: Entity graph traversal Documentation/ABI/testing/sysfs-bus-media | 6 + Documentation/DocBook/media-entities.tmpl | 24 + Documentation/DocBook/media.tmpl | 3 + Documentation/DocBook/v4l/media-controller.xml | 89 ++++ Documentation/DocBook/v4l/media-func-close.xml | 59 +++ Documentation/DocBook/v4l/media-func-ioctl.xml | 116 +++++ Documentation/DocBook/v4l/media-func-open.xml | 94 ++++ .../DocBook/v4l/media-ioc-device-info.xml | 133 +++++ .../DocBook/v4l/media-ioc-enum-entities.xml | 308 +++++++++++ Documentation/DocBook/v4l/media-ioc-enum-links.xml | 207 ++++++++ Documentation/DocBook/v4l/media-ioc-setup-link.xml | 93 ++++ Documentation/media-framework.txt | 353 +++++++++++++ Documentation/video4linux/v4l2-framework.txt | 72 +++- drivers/media/Kconfig | 13 + drivers/media/Makefile | 10 +- drivers/media/media-device.c | 382 ++++++++++++++ drivers/media/media-devnode.c | 321 ++++++++++++ drivers/media/media-entity.c | 536 ++++++++++++++++++++ drivers/media/video/v4l2-dev.c | 49 ++- drivers/media/video/v4l2-device.c | 52 ++- drivers/media/video/v4l2-subdev.c | 41 ++- include/linux/Kbuild | 1 + include/linux/media.h | 132 +++++ include/media/media-device.h | 95 ++++ include/media/media-devnode.h | 97 ++++ include/media/media-entity.h | 147 ++++++ include/media/v4l2-dev.h | 7 + include/media/v4l2-device.h | 4 + include/media/v4l2-subdev.h | 10 + 29 files changed, 3428 insertions(+), 26 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-media create mode 100644 Documentation/DocBook/v4l/media-controller.xml create mode 100644 Documentation/DocBook/v4l/media-func-close.xml create mode 100644 Documentation/DocBook/v4l/media-func-ioctl.xml create mode 100644 Documentation/DocBook/v4l/media-func-open.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-device-info.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-enum-entities.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-enum-links.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-setup-link.xml create mode 100644 Documentation/media-framework.txt create mode 100644 drivers/media/media-device.c create mode 100644 drivers/media/media-devnode.c create mode 100644 drivers/media/media-entity.c create mode 100644 include/linux/media.h create mode 100644 include/media/media-device.h create mode 100644 include/media/media-devnode.h create mode 100644 include/media/media-entity.h -- Regards, Laurent Pinchart

4 18

[alsa-devel] [PATCH 1/7] ASoC: neo1973_wm8753: Remove scenario management code.
by Lars-Peter Clausen 08 Feb '11

08 Feb '11

It has been proven to be inflexible to do scenario management in kernel space. Since actual neo1973 board support has not been merged in mainline and this patch has been in the neo1973 tree for some time now it should be safe to remove this functionality without breaking existing userspace. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/neo1973_wm8753.c | 142 +---------------------------------- 1 files changed, 5 insertions(+), 137 deletions(-) diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index d3cd688..1ea6e61 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -37,17 +37,6 @@ #include "dma.h" #include "s3c24xx-i2s.h" -/* define the scenarios */ -#define NEO_AUDIO_OFF 0 -#define NEO_GSM_CALL_AUDIO_HANDSET 1 -#define NEO_GSM_CALL_AUDIO_HEADSET 2 -#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3 -#define NEO_STEREO_TO_SPEAKERS 4 -#define NEO_STEREO_TO_HEADPHONES 5 -#define NEO_CAPTURE_HANDSET 6 -#define NEO_CAPTURE_HEADSET 7 -#define NEO_CAPTURE_BLUETOOTH 8 - static struct snd_soc_card neo1973; static struct i2c_client *i2c; @@ -224,113 +213,6 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; -static int neo1973_scenario; - -static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = neo1973_scenario; - return 0; -} - -static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - pr_debug("Entered %s\n", __func__); - - switch (neo1973_scenario) { - case NEO_AUDIO_OFF: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HANDSET: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HEADSET: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_BLUETOOTH: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_STEREO_TO_SPEAKERS: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_STEREO_TO_HEADPHONES: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_CAPTURE_HANDSET: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Call Mic"); - break; - case NEO_CAPTURE_HEADSET: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_CAPTURE_BLUETOOTH: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - default: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - } - - snd_soc_dapm_sync(dapm); - - return 0; -} - -static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - - pr_debug("Entered %s\n", __func__); - - if (neo1973_scenario == ucontrol->value.integer.value[0]) - return 0; - - neo1973_scenario = ucontrol->value.integer.value[0]; - set_scenario_endpoints(codec, neo1973_scenario); - return 1; -} - static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; static void lm4857_write_regs(void) @@ -454,22 +336,6 @@ static const struct soc_enum lm4857_mode_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode), }; -static const char *neo_scenarios[] = { - "Off", - "GSM Handset", - "GSM Headset", - "GSM Bluetooth", - "Speakers", - "Headphones", - "Capture Handset", - "Capture Headset", - "Capture Bluetooth" -}; - -static const struct soc_enum neo_scenario_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), -}; - static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); @@ -482,8 +348,6 @@ static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { lm4857_get_reg, lm4857_set_reg, mono_tlv), SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], lm4857_get_mode, lm4857_set_mode), - SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], - neo1973_get_scenario, neo1973_set_scenario), SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0, lm4857_get_reg, lm4857_set_reg), SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0, @@ -520,7 +384,11 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(wm8753_dapm_widgets)); /* set endpoints to default mode */ - set_scenario_endpoints(codec, NEO_AUDIO_OFF); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); /* add neo1973 specific controls */ err = snd_soc_add_controls(codec, wm8753_neo1973_controls, -- 1.7.2.3

3 19

Re: [alsa-devel] [PATCH v8 05/12] media: Entity use count
by Hans Verkuil 08 Feb '11

08 Feb '11

On Friday, February 04, 2011 13:34:02 Sakari Ailus wrote: > Hi, > > And many thanks for the comments! > > Hans Verkuil wrote: > ... > >> diff --git a/include/media/media-entity.h b/include/media/media-entity.h > >> index b82f824..114541a 100644 > >> --- a/include/media/media-entity.h > >> +++ b/include/media/media-entity.h > >> @@ -81,6 +81,8 @@ struct media_entity { > >> struct media_pad *pads; /* Pads array (num_pads elements) */ > >> struct media_link *links; /* Links array (max_links elements)*/ > >> > >> + int use_count; /* Use count for the entity. */ > > > > Isn't unsigned better? > > Could be. The result, though, will be slightly more difficult checking > for bad use count --- which always is a driver bug. > > me->use_count += change; > WARN_ON(me->use_count < 0); > > we must do something like this: > > if (change < 0) > WARN_ON(me->use_count < (unsigned)-change); > me->use_count += change; > > I'd perhaps also go with unsigned int; the choice for signed was made > mainly since the above check and with signed int the check was more trivial. I saw this WARN_ON as well. I think there is a good reason for that WARN_ON, but I think a comment in the header explaining why it is an int will be useful. If I trip over it, then others will as well :-) Regards, Hans > > Regards, > > -- Hans Verkuil - video4linux developer - sponsored by Cisco

2 1

[alsa-devel] [patch] ALSA: soc-cache: dereferencing before checking
by Dan Carpenter 08 Feb '11

08 Feb '11

The patch c358e640a66 "ASoC: soc-cache: Add trace event for snd_soc_cache_sync()" introduced a dereference of "codec->cache_ops" before we had checked it for NULL. I pulled the check forward, and then pulled everything in an indent level. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index db66dc4..5d76da4 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1609,24 +1609,23 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return 0; } + if (!codec->cache_ops || !codec->cache_ops->sync) + return -EINVAL; + if (codec->cache_ops->name) name = codec->cache_ops->name; else name = "unknown"; - if (codec->cache_ops && codec->cache_ops->sync) { - if (codec->cache_ops->name) - dev_dbg(codec->dev, "Syncing %s cache for %s codec\n", - codec->cache_ops->name, codec->name); - trace_snd_soc_cache_sync(codec, name, "start"); - ret = codec->cache_ops->sync(codec); - if (!ret) - codec->cache_sync = 0; - trace_snd_soc_cache_sync(codec, name, "end"); - return ret; - } - - return -EINVAL; + if (codec->cache_ops->name) + dev_dbg(codec->dev, "Syncing %s cache for %s codec\n", + codec->cache_ops->name, codec->name); + trace_snd_soc_cache_sync(codec, name, "start"); + ret = codec->cache_ops->sync(codec); + if (!ret) + codec->cache_sync = 0; + trace_snd_soc_cache_sync(codec, name, "end"); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_cache_sync);

4 3

[alsa-devel] [PATCH] ASoC: CS4271: Move Chip Select control out of the CODEC code.
by Alexander Sverdlin 08 Feb '11

08 Feb '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Move Chip Select control out of the CODEC code for CS4271. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- include/sound/cs4271.h | 1 - sound/soc/codecs/cs4271.c | 22 +++------------------- 2 files changed, 3 insertions(+), 20 deletions(-) diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index 16f8d32..50a059e 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -19,7 +19,6 @@ struct cs4271_platform_data { int gpio_nreset; /* GPIO driving Reset pin, if any */ - int gpio_disable; /* GPIO that disable serial bus, if any */ }; #endif /* __CS4271_H */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 9c5b7db..1791796 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -441,22 +441,11 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; int gpio_nreset = -EINVAL; - int gpio_disable = -EINVAL; codec->control_data = cs4271->control_data; - if (cs4271plat) { - if (gpio_is_valid(cs4271plat->gpio_nreset)) - gpio_nreset = cs4271plat->gpio_nreset; - if (gpio_is_valid(cs4271plat->gpio_disable)) - gpio_disable = cs4271plat->gpio_disable; - } - - if (gpio_disable >= 0) - if (gpio_request(gpio_disable, "CS4271 Disable")) - gpio_disable = -EINVAL; - if (gpio_disable >= 0) - gpio_direction_output(gpio_disable, 0); + if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) + gpio_nreset = cs4271plat->gpio_nreset; if (gpio_nreset >= 0) if (gpio_request(gpio_nreset, "CS4271 Reset")) @@ -471,7 +460,6 @@ static int cs4271_probe(struct snd_soc_codec *codec) } cs4271->gpio_nreset = gpio_nreset; - cs4271->gpio_disable = gpio_disable; /* * In case of I2C, chip address specified in board data. @@ -509,10 +497,9 @@ static int cs4271_probe(struct snd_soc_codec *codec) static int cs4271_remove(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - int gpio_nreset, gpio_disable; + int gpio_nreset; gpio_nreset = cs4271->gpio_nreset; - gpio_disable = cs4271->gpio_disable; if (gpio_is_valid(gpio_nreset)) { /* Set codec to the reset state */ @@ -520,9 +507,6 @@ static int cs4271_remove(struct snd_soc_codec *codec) gpio_free(gpio_nreset); } - if (gpio_is_valid(gpio_disable)) - gpio_free(gpio_disable); - return 0; };

4 3

Re: [alsa-devel] issue with AD1989A HDA codec and alsamixer
by Raymond Yau 08 Feb '11

08 Feb '11

2011/2/7 Vincent Meserette <vmeserette(a)aldebaran-robotics.com> > > I found from where came my problem with the capture gains by adding some > logs in the hda_codec.c file. > When I set one of the capture gain, the put_vol_mute function well write in > the AMP_GAIN_MUTE verb of the corresponding ADC selector (0x0c, 0x0d or 0x0e > widget) but the mute bit is always set to 1. > Indeed, the value associated to the AMP_GAIN_MUTE verb is formated as folow > : > bits 0 to 6 : gain > bit 7 : mute > bits 8 to 11 : index > bit 12 : set right amp > bit 13 : set left amp > bit 14 : set input amp > bit 15 : set output amp > > and alsa sets the value of this register from a080 / b080 to a0b6 / b0b6 > when i change the gain from -58.5 dB to 22.5dB > > Do you know how alsa and thus the alsamixer determine the minimum and > maximum values for the capture gain, and how to say to alsa the the gain is > only a 7bits step value ? > Take a look at HDA Specification 7.3.4.10 Amplifier Capabilities StepSize (7 bits) indicates the size of each step in the gain range. Each individual step may be 0-32 dB specified in 0.25-dB steps. A value of 0 indicates 0.25-dB steps, while a value of 127d indicates a 32-dB step. NumSteps (7 bits) indicates the number of steps in the gain range. There may be from 1 to 128 steps in the amplifier gain range. (0d means 1 step, 127d means 128 steps). A value of 0 (1 step) means that the gain is fixed and may not be changed. Offset (7 bits) indicates which step is 0 dB. If there are two or more steps, one of the step values must correspond to a value of 0 dB. The “Offset” value reflects the value which, if programmed in to the Amplifier Gain control, would result in a gain of 0 dB. Mute Capable (1 bit) reports if the respective amplifier is capable of muting. Muting implies a –infinity gain (no sound passes), but the actual performance is determined by the hardware. Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Input Source", index=0, device=0 Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0x25 0x25] Connection: 10 0x38 0x39* 0x3a 0x3b 0x3c 0x18 0x24 0x25 0x3d 0x20 Node 0x0d [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Capture Volume", index=1, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Capture Switch", index=1, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Input Source", index=1, device=0 Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0xa7 0xa7] Connection: 10 0x38 0x39* 0x3a 0x3b 0x3c 0x18 0x24 0x25 0x3d 0x20 Node 0x0e [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Capture Volume", index=2, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Capture Switch", index=2, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Input Source", index=2, device=0 Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0xa7 0xa7] Connection: 10 0x38 0x39* 0x3a 0x3b 0x3c 0x18 0x24 0x25 0x3d 0x20

1 0

Re: [alsa-devel] [PATCH] [ALSA] Enable the E-MU 0204 USB
by Takashi Iwai 08 Feb '11

08 Feb '11

At Tue, 8 Feb 2011 01:22:36 -0500, jteich3(a)gmail.com wrote: > > From: Joseph Teichman <josteich(a)gmail.com> > > > Signed-off-by: Joseph Teichman <josteich(a)gmail.com> Applied now. Thanks! Takashi > diff --git a/usb/mixer.c b/usb/mixer.c > index 7df89b3..85af605 100644 > --- a/usb/mixer.c > +++ b/usb/mixer.c > @@ -95,7 +95,7 @@ enum { > }; > > > -/*E-mu 0202(0404) eXtension Unit(XU) control*/ > +/*E-mu 0202/0404/0204 eXtension Unit(XU) control*/ > enum { > USB_XU_CLOCK_RATE = 0xe301, > USB_XU_CLOCK_SOURCE = 0xe302, > @@ -1566,7 +1566,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw > cval->initialized = 1; > } else { > if (type == USB_XU_CLOCK_RATE) { > - /* E-Mu USB 0404/0202/TrackerPre > + /* E-Mu USB 0404/0202/TrackerPre/0204 > * samplerate control quirk > */ > cval->min = 0; > diff --git a/usb/quirks-table.h b/usb/quirks-table.h > index 3599987..921a86f 100644 > --- a/usb/quirks-table.h > +++ b/usb/quirks-table.h > @@ -79,6 +79,13 @@ > .idProduct = 0x3f0a, > .bInterfaceClass = USB_CLASS_AUDIO, > }, > +{ > + /* E-Mu 0204 USB */ > + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, > + .idVendor = 0x041e, > + .idProduct = 0x3f19, > + .bInterfaceClass = USB_CLASS_AUDIO, > +}, > > /* > * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface > diff --git a/usb/quirks.c b/usb/quirks.c > index cf8bf08..e314cdb 100644 > --- a/usb/quirks.c > +++ b/usb/quirks.c > @@ -532,7 +532,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat > } > > /* > - * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, > + * For E-Mu 0404USB/0202USB/TrackerPre/0204 sample rate should be set for device, > * not for interface. > */ > > @@ -589,6 +589,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, > case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ > case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ > case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ > + case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ > set_format_emu_quirk(subs, fmt); > break; > } > -- > 1.7.0.4 >

1 0

[alsa-devel] [PATCH] [ALSA] Enable the E-MU 0204 USB
by josteich＠gmail.com 08 Feb '11

08 Feb '11

From: Joseph Teichman <josteich(a)gmail.com> Signed-off-by: Joseph Teichman <josteich(a)gmail.com> diff --git a/usb/mixer.c b/usb/mixer.c index 7df89b3..85af605 100644 --- a/usb/mixer.c +++ b/usb/mixer.c @@ -95,7 +95,7 @@ enum { }; -/*E-mu 0202(0404) eXtension Unit(XU) control*/ +/*E-mu 0202/0404/0204 eXtension Unit(XU) control*/ enum { USB_XU_CLOCK_RATE = 0xe301, USB_XU_CLOCK_SOURCE = 0xe302, @@ -1566,7 +1566,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw cval->initialized = 1; } else { if (type == USB_XU_CLOCK_RATE) { - /* E-Mu USB 0404/0202/TrackerPre + /* E-Mu USB 0404/0202/TrackerPre/0204 * samplerate control quirk */ cval->min = 0; diff --git a/usb/quirks-table.h b/usb/quirks-table.h index 3599987..921a86f 100644 --- a/usb/quirks-table.h +++ b/usb/quirks-table.h @@ -79,6 +79,13 @@ .idProduct = 0x3f0a, .bInterfaceClass = USB_CLASS_AUDIO, }, +{ + /* E-Mu 0204 USB */ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x041e, + .idProduct = 0x3f19, + .bInterfaceClass = USB_CLASS_AUDIO, +}, /* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface diff --git a/usb/quirks.c b/usb/quirks.c index cf8bf08..e314cdb 100644 --- a/usb/quirks.c +++ b/usb/quirks.c @@ -532,7 +532,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat } /* - * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, + * For E-Mu 0404USB/0202USB/TrackerPre/0204 sample rate should be set for device, * not for interface. */ @@ -589,6 +589,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ + case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; } -- 1.7.0.4

1 0

[PATCH 1/2] ALSA: HDA: Add a new Conexant codec 506e (20590)
by David Henningsson 08 Feb '11

08 Feb '11

Conexant 506e/20590 has the same graph as the rest of the 5066 family. BugLink: http://bugs.launchpad.net/bugs/723672 Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index dd7c5c1..909ce9e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3937,6 +3937,8 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5066 }, { .id = 0x14f15069, .name = "CX20585", .patch = patch_cxt5066 }, + { .id = 0x14f1506e, .name = "CX20590", + .patch = patch_cxt5066 }, { .id = 0x14f15097, .name = "CX20631", .patch = patch_conexant_auto }, { .id = 0x14f15098, .name = "CX20632", @@ -3963,6 +3965,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15066"); MODULE_ALIAS("snd-hda-codec-id:14f15067"); MODULE_ALIAS("snd-hda-codec-id:14f15068"); MODULE_ALIAS("snd-hda-codec-id:14f15069"); +MODULE_ALIAS("snd-hda-codec-id:14f1506e"); MODULE_ALIAS("snd-hda-codec-id:14f15097"); MODULE_ALIAS("snd-hda-codec-id:14f15098"); MODULE_ALIAS("snd-hda-codec-id:14f150a1"); -- 1.7.1 --------------020500020206030309030703--

1 0

[alsa-devel] ANNOUNCE: NVIDIA HDMI audio documentation
by Stephen Warren 07 Feb '11

07 Feb '11

I have written a document re: how to understand and trouble-shoot issues with audio-over-HDMI on NVIDIA GPUs. It's available at: ftp://download.nvidia.com/XFree86/gpu-hdmi-audio-document/gpu-hdmi-audio.ht… The intent is to provide all the support information I've given various people on support forums in a centralized place. Hope this helps! -- nvpublic

2 4

[alsa-devel] ALSA: HDA ad1989 on Intel SCH interrupt issues
by Aleksandar Mihaylov 07 Feb '11

07 Feb '11

Hello, I have been working on a custom audio interface, utilizing the Intel SCH and an ad1989 codec. the goal is to provide 1 stereo headphone output and 3 stereo microphone inputs. As the Intel SCH only provides two DMA engines per direction my first question will be: How do i setup interleaving of the 3 stereo ADCs provided by the ad1989, through the two DMA engines. Additionally the more intriguing problem involves interrupt handling during capture(using APIC). Upon boot (or module reload) playback works just fine, however recording fails with "ALSA pcm_lib.c:1787: capture write error (DMA or IRQ trouble?)". This behaviour is consistent if one tries to continuously record form one stereo channel and is applicable to all 3 PCM-substreams as long as they are accessed one at a time. However if i try to record from 2 or more PCM substreams simultaneously, after a few failed attempts(same issue as above: DMA/IRQ trouble) one of the channels will engage, after which any subsequent recording request will start without any problems. The above issues arise when i force the HDA controller in APIC mode. If the controller is left in the default MSI interrupt mode then the kernel receives no interrupts on that IRQ, resulting in playback only playing the first buffer repetitively, until timing out with "DMA or IRQ trouble?". However recording exhibits a similar behaviour to above, where after a few attempts it successfully begins to record, inspection of /proc/interrupts still returns 0 for the assigned IRQ line. The same behaviour is exhibited when i use the default patch_analog.c module (snd-hda-codec-analog). Playback and recording through other media such as usb seem to work fine. Attached i am sending the alsa-info outputs immediately upon boot, and after the capture streams have begun recording properly, i have included detailed lspci output at the end of each file. Additionally i am sending the patch for alsa-driver 1.0.24 that contains the code for the new module as well as the PCI_QUIRKS i have added to hda_intel.c. My working directory is available through at: git://github.com/sashkosnail/AD1989_HDA.git I will much appreciate any input on the problem. -- Aleksandar Mihaylov, EIT

1 0

[alsa-devel] [PATCH] ASoC: codecs: wm8753: Fix DAI mode switching
by Lars-Peter Clausen 07 Feb '11

07 Feb '11

The wm8753 codec supports switching between different DAI modes. The current drivers tries to implement this by changing the DAI driver at runtime. But to properly work this would require support from the ASoC core. So this patch takes a different approch on how the DAI mode switching is implemented. The only difference, from a driver point of view, between the different DAI modes is how to program the DAI format to the hardware. So what this patch is, it stores the current format for each DAI in the drivers private struct and when the DAI mode is changed the format gets simply reprogrammed according to the new DAI mode. Futhermore this patch restricts the changing of the DAI format to when the codec is inactive. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8753.c | 296 ++++++++++++++++++-------------------------- 1 files changed, 121 insertions(+), 175 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 79b02ae..3f09dee 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -55,8 +55,10 @@ static int caps_charge = 2000; module_param(caps_charge, int, 0); MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)"); -static void wm8753_set_dai_mode(struct snd_soc_codec *codec, - struct snd_soc_dai *dai, unsigned int hifi); +static int wm8753_hifi_write_dai_fmt(struct snd_soc_codec *codec, + unsigned int fmt); +static int wm8753_voice_write_dai_fmt(struct snd_soc_codec *codec, + unsigned int fmt); /* * wm8753 register cache @@ -87,6 +89,10 @@ struct wm8753_priv { enum snd_soc_control_type control_type; unsigned int sysclk; unsigned int pcmclk; + + unsigned int voice_fmt; + unsigned int hifi_fmt; + int dai_func; }; @@ -170,9 +176,9 @@ static int wm8753_get_dai(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int mode = snd_soc_read(codec, WM8753_IOCTL); + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.integer.value[0] = (mode & 0xc) >> 2; + ucontrol->value.integer.value[0] = wm8753->dai_func; return 0; } @@ -180,16 +186,26 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int mode = snd_soc_read(codec, WM8753_IOCTL); struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); + u16 ioctl; + + if (codec->active) + return -EBUSY; + + ioctl = snd_soc_read(codec, WM8753_IOCTL); + + wm8753->dai_func = ucontrol->value.integer.value[0]; + + if (((ioctl >> 2) & 0x3) == wm8753->dai_func) + return 1; + + ioctl = (ioctl & 0x1f3) | (wm8753->dai_func << 2); + snd_soc_write(codec, WM8753_IOCTL, ioctl); - if (((mode & 0xc) >> 2) == ucontrol->value.integer.value[0]) - return 0; - mode &= 0xfff3; - mode |= (ucontrol->value.integer.value[0] << 2); + wm8753_hifi_write_dai_fmt(codec, wm8753->hifi_fmt); + wm8753_voice_write_dai_fmt(codec, wm8753->voice_fmt); - wm8753->dai_func = ucontrol->value.integer.value[0]; return 1; } @@ -828,10 +844,9 @@ static int wm8753_set_dai_sysclk(struct snd_soc_dai *codec_dai, /* * Set's ADC and Voice DAC format. */ -static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 voice = snd_soc_read(codec, WM8753_PCM) & 0x01ec; /* interface format */ @@ -858,13 +873,6 @@ static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -static int wm8753_pcm_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - wm8753_set_dai_mode(dai->codec, dai, 0); - return 0; -} - /* * Set PCM DAI bit size and sample rate. */ @@ -905,10 +913,9 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, /* * Set's PCM dai fmt and BCLK. */ -static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 voice, ioctl; voice = snd_soc_read(codec, WM8753_PCM) & 0x011f; @@ -999,10 +1006,9 @@ static int wm8753_set_dai_clkdiv(struct snd_soc_dai *codec_dai, /* * Set's HiFi DAC format. */ -static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 hifi = snd_soc_read(codec, WM8753_HIFI) & 0x01e0; /* interface format */ @@ -1032,10 +1038,9 @@ static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai, /* * Set's I2S DAI format. */ -static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 ioctl, hifi; hifi = snd_soc_read(codec, WM8753_HIFI) & 0x011f; @@ -1098,13 +1103,6 @@ static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -static int wm8753_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - wm8753_set_dai_mode(dai->codec, dai, 1); - return 0; -} - /* * Set PCM DAI bit size and sample rate. */ @@ -1147,61 +1145,117 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 clock; /* set clk source as pcmclk */ clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb; snd_soc_write(codec, WM8753_CLOCK, clock); - if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0) - return -EINVAL; - return wm8753_pcm_set_dai_fmt(codec_dai, fmt); + return wm8753_vdac_adc_set_dai_fmt(codec, fmt); } -static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0) - return -EINVAL; - return wm8753_i2s_set_dai_fmt(codec_dai, fmt); + return wm8753_hdac_set_dai_fmt(codec, fmt); } -static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 clock; /* set clk source as pcmclk */ clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb; snd_soc_write(codec, WM8753_CLOCK, clock); - if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0) - return -EINVAL; - return wm8753_i2s_set_dai_fmt(codec_dai, fmt); + return wm8753_vdac_adc_set_dai_fmt(codec, fmt); } -static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 clock; /* set clk source as mclk */ clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb; snd_soc_write(codec, WM8753_CLOCK, clock | 0x4); - if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0) + if (wm8753_hdac_set_dai_fmt(codec, fmt) < 0) return -EINVAL; - if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0) - return -EINVAL; - return wm8753_i2s_set_dai_fmt(codec_dai, fmt); + return wm8753_vdac_adc_set_dai_fmt(codec, fmt); } +static int wm8753_hifi_write_dai_fmt(struct snd_soc_codec *codec, + unsigned int fmt) +{ + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (wm8753->dai_func) { + case 0: + ret = wm8753_mode1h_set_dai_fmt(codec, fmt); + break; + case 1: + ret = wm8753_mode2_set_dai_fmt(codec, fmt); + break; + case 2: + case 3: + ret = wm8753_mode3_4_set_dai_fmt(codec, fmt); + break; + default: + break; + } + if (ret) + return ret; + + return wm8753_i2s_set_dai_fmt(codec, fmt); +} + +static int wm8753_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); + + wm8753->hifi_fmt = fmt; + + return wm8753_hifi_write_dai_fmt(codec, fmt); +}; + +static int wm8753_voice_write_dai_fmt(struct snd_soc_codec *codec, + unsigned int fmt) +{ + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if (wm8753->dai_func != 0) + return 0; + + ret = wm8753_mode1v_set_dai_fmt(codec, fmt); + if (ret) + return ret; + ret = wm8753_pcm_set_dai_fmt(codec, fmt); + if (ret) + return ret; + + return 0; +}; + +static int wm8753_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); + + wm8753->voice_fmt = fmt; + + return wm8753_voice_write_dai_fmt(codec, fmt); +}; + static int wm8753_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; @@ -1268,57 +1322,25 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, * 3. Voice disabled - HIFI over HIFI * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture */ -static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode1 = { - .startup = wm8753_i2s_startup, +static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode = { .hw_params = wm8753_i2s_hw_params, .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode1h_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, -}; - -static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode1 = { - .startup = wm8753_pcm_startup, - .hw_params = wm8753_pcm_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode1v_set_dai_fmt, + .set_fmt = wm8753_hifi_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, .set_pll = wm8753_set_dai_pll, .set_sysclk = wm8753_set_dai_sysclk, }; -static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode2 = { - .startup = wm8753_pcm_startup, +static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode = { .hw_params = wm8753_pcm_hw_params, .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode2_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, -}; - -static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode3 = { - .startup = wm8753_i2s_startup, - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode3_4_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, -}; - -static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode4 = { - .startup = wm8753_i2s_startup, - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode3_4_set_dai_fmt, + .set_fmt = wm8753_voice_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, .set_pll = wm8753_set_dai_pll, .set_sysclk = wm8753_set_dai_sysclk, }; -static struct snd_soc_dai_driver wm8753_all_dai[] = { +static struct snd_soc_dai_driver wm8753_dai[] = { /* DAI HiFi mode 1 */ { .name = "wm8753-hifi", .playback = { @@ -1326,14 +1348,16 @@ static struct snd_soc_dai_driver wm8753_all_dai[] = { .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, - .formats = WM8753_FORMATS}, + .formats = WM8753_FORMATS + }, .capture = { /* dummy for fast DAI switching */ .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, - .formats = WM8753_FORMATS}, - .ops = &wm8753_dai_ops_hifi_mode1, + .formats = WM8753_FORMATS + }, + .ops = &wm8753_dai_ops_hifi_mode, }, /* DAI Voice mode 1 */ { .name = "wm8753-voice", @@ -1342,97 +1366,19 @@ static struct snd_soc_dai_driver wm8753_all_dai[] = { .channels_min = 1, .channels_max = 1, .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .ops = &wm8753_dai_ops_voice_mode1, -}, -/* DAI HiFi mode 2 - dummy */ -{ .name = "wm8753-hifi", -}, -/* DAI Voice mode 2 */ -{ .name = "wm8753-voice", - .playback = { - .stream_name = "Voice Playback", - .channels_min = 1, - .channels_max = 1, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .ops = &wm8753_dai_ops_voice_mode2, -}, -/* DAI HiFi mode 3 */ -{ .name = "wm8753-hifi", - .playback = { - .stream_name = "HiFi Playback", - .channels_min = 1, - .channels_max = 2, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .ops = &wm8753_dai_ops_hifi_mode3, -}, -/* DAI Voice mode 3 - dummy */ -{ .name = "wm8753-voice", -}, -/* DAI HiFi mode 4 */ -{ .name = "wm8753-hifi", - .playback = { - .stream_name = "HiFi Playback", - .channels_min = 1, - .channels_max = 2, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, + .formats = WM8753_FORMATS, + }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .ops = &wm8753_dai_ops_hifi_mode4, -}, -/* DAI Voice mode 4 - dummy */ -{ .name = "wm8753-voice", -}, -}; - -static struct snd_soc_dai_driver wm8753_dai[] = { - { - .name = "wm8753-aif0", - }, - { - .name = "wm8753-aif1", + .formats = WM8753_FORMATS, }, + .ops = &wm8753_dai_ops_voice_mode, +}, }; -static void wm8753_set_dai_mode(struct snd_soc_codec *codec, - struct snd_soc_dai *dai, unsigned int hifi) -{ - struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); - - if (wm8753->dai_func < 4) { - if (hifi) - dai->driver = &wm8753_all_dai[wm8753->dai_func << 1]; - else - dai->driver = &wm8753_all_dai[(wm8753->dai_func << 1) + 1]; - } - snd_soc_write(codec, WM8753_IOCTL, wm8753->dai_func); -} - static void wm8753_work(struct work_struct *work) { struct snd_soc_dapm_context *dapm = -- 1.7.2.3

3 2

[alsa-devel] [patch] USB: 6fire: signedness bug in usb6fire_pcm_prepare()
by Dan Carpenter 07 Feb '11

07 Feb '11

rt->rate is an unsigned char so it's never equal to -1. It's not a huge problem because the invalid rate is caught inside the call to usb6fire_pcm_set_rate() which returns -EINVAL. But if we fix the test then it prints out the correct error message so that's good. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 705c88e..ba62c74 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -493,13 +493,12 @@ static int usb6fire_pcm_prepare(struct snd_pcm_substream *alsa_sub) sub->period_off = 0; if (rt->stream_state == STREAM_DISABLED) { - rt->rate = -1; for (i = 0; i < ARRAY_SIZE(rates); i++) if (alsa_rt->rate == rates[i]) { rt->rate = i; break; } - if (rt->rate == -1) { + if (i == ARRAY_SIZE(rates)) { mutex_unlock(&rt->stream_mutex); snd_printk("invalid rate %d in prepare.\n", alsa_rt->rate);

2 1

Re: [alsa-devel] issue with AD1989A HDA codec and alsamixer
by Raymond Yau 07 Feb '11

07 Feb '11

2011/2/7 Vincent Meserette <vmeserette(a)aldebaran-robotics.com> > > To record four channels simultaneously, I have created the following alsa configuration file : > > *# output device > pcm.AD1989A{ > type hw > card 0 > } > > # input device* * > pcm.quad { > type multi > > slaves.a.pcm "hw:0,0,0" # ADC 0 of the codec (left and right mics) > slaves.a.channels 2 > slaves.b.pcm "hw:0,0,1" # ADC 1 of the codec (front and rear mics) > slaves.b.channels 2 > > bindings.0.slave a > bindings.0.channel 0 > bindings.1.slave a > bindings.1.channel 1 > bindings.2.slave b > bindings.2.channel 0 > bindings.3.slave b > bindings.3.channel 1 > } > > # input device containing interleaved data > pcm.quad2 { > type route > slave.pcm "quad" > ttable.0.0 1 > ttable.1.1 1 > ttable.2.2 1 > ttable.3.3 1 > } > > ### CTL DEVICES (controls) > ctl.mixer0 { > type hw > card 0 > }* > > With this configuration file, I am able to record the four channels > simultaneously with the following command : > > arecord -c4 -fS16_LE -r48000 -D «quad2 » -d5 /tmp/recording.wav > > it is strange that I alway got overrun only at the first time of running the above command but no overrun on the second time arecord -v -c4 -fS16_LE -r48000 -D quad2 -d5 /tmp/recording.wav Recording WAVE '/tmp/recording.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Channels 4 Route conversion PCM Transformation table: 0 <- 0 1 <- 1 2 <- 2 3 <- 3 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 4 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 85333 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Multi PCM Channel bindings: 0: slave 0, channel 0 1: slave 0, channel 1 2: slave 1, channel 0 3: slave 1, channel 1 Its setup is: stream : CAPTURE access : MMAP_COMPLEX format : S16_LE subformat : STD channels : 4 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 85333 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave #0: Hardware PCM card 1 'HDA Intel' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 85333 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 Slave #1: Hardware PCM card 1 'HDA Intel' device 0 subdevice 1 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 85333 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 overrun!!! (at least 0.001 ms long) Status: state : XRUN trigger_time: 1077.282286934 tstamp : 1077.282292637 delay : 0 avail : 16384 avail_max : 16384 overrun!!! (at least 0.001 ms long) Status: state : XRUN trigger_time: 1077.623810402 tstamp : 1077.623819150 delay : 0 avail : 16384

1 0

[PATCH] ALSA: HDA: New AD1984A model for Dell Precision R5500
by David Henningsson 07 Feb '11

07 Feb '11

For codec AD1984A, add a new model to support Dell Precision R5500 or the microphone jack won't work correctly. BugLink: http://bugs.launchpad.net/bugs/741516 Tested-by: Kent Baxley <kent.baxley(a)canonical.com> Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_analog.c | 89 ++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 89 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 734c6ee..2942d2a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -4256,6 +4256,84 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec) } /* + * Precision R5500 + * 0x12 - HP/line-out + * 0x13 - speaker (mono) + * 0x15 - mic-in + */ + +static struct hda_verb ad1984a_precision_verbs[] = { + /* Unmute main output path */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x1f}, /* 0dB */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) + 0x17}, /* 0dB */ + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* Select mic as input */ + {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x27}, /* 0dB */ + /* Configure as mic */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ + /* HP unmute */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* turn on EAPD */ + {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + /* unsolicited event for pin-sense */ + {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984a_precision_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + { } /* end */ +}; + + +/* mute internal speaker if HP is plugged */ +static void ad1984a_precision_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_jack_detect(codec, 0x12); + snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + + +/* unsolicited event for HP jack sensing */ +static void ad1984a_precision_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != AD1884A_HP_EVENT) + return; + ad1984a_precision_automute(codec); +} + +/* initialize jack-sensing, too */ +static int ad1984a_precision_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1984a_precision_automute(codec); + return 0; +} + + +/* * HP Touchsmart * port-A (0x11) - front hp-out * port-B (0x14) - unused @@ -4384,6 +4462,7 @@ enum { AD1884A_MOBILE, AD1884A_THINKPAD, AD1984A_TOUCHSMART, + AD1984A_PRECISION, AD1884A_MODELS }; @@ -4393,9 +4472,11 @@ static const char * const ad1884a_models[AD1884A_MODELS] = { [AD1884A_MOBILE] = "mobile", [AD1884A_THINKPAD] = "thinkpad", [AD1984A_TOUCHSMART] = "touchsmart", + [AD1984A_PRECISION] = "precision", }; static struct snd_pci_quirk ad1884a_cfg_tbl[] = { + SND_PCI_QUIRK(0x1028, 0x04ac, "Precision R5500", AD1984A_PRECISION), SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), @@ -4489,6 +4570,14 @@ static int patch_ad1884a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; codec->patch_ops.init = ad1984a_thinkpad_init; break; + case AD1984A_PRECISION: + spec->mixers[0] = ad1984a_precision_mixers; + spec->init_verbs[spec->num_init_verbs++] = + ad1984a_precision_verbs; + spec->multiout.dig_out_nid = 0; + codec->patch_ops.unsol_event = ad1984a_precision_unsol_event; + codec->patch_ops.init = ad1984a_precision_init; + break; case AD1984A_TOUCHSMART: spec->mixers[0] = ad1984a_touchsmart_mixers; spec->init_verbs[0] = ad1984a_touchsmart_verbs; -- 1.7.1 --------------040907030604030002090009--

1 0

[alsa-devel] [PATCH - hdspmixer 1/1] hdspmixer: Fix detection for non-MADIface RME cards
by Adrian Knoth 07 Feb '11

07 Feb '11

When adding RayDAT support, detection for cards like the 9652 was accidently lost. This patch re-adds the missing bits. Signed-off-by: Jörn Nettingsmeier <nettings(a)stackingdwarves.net> Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/hdspmixer/src/hdspmixer.cxx b/hdspmixer/src/hdspmixer.cxx index 7fd50e9..2a7c5c9 100644 --- a/hdspmixer/src/hdspmixer.cxx +++ b/hdspmixer/src/hdspmixer.cxx @@ -51,7 +51,23 @@ int main(int argc, char **argv) } else { snd_card_get_longname(card, &name); printf("Card %d : %s\n", card, name); - if (!strncmp(name, "RME MADIface", 12)) { + if (!strncmp(name, "RME Hammerfall DSP + Multiface", 30)) { + printf("Multiface found !\n"); + hdsp_cards[cards] = new HDSPMixerCard(Multiface, card); + cards++; + } else if (!strncmp(name, "RME Hammerfall DSP + Digiface", 29)) { + printf("Digiface found !\n"); + hdsp_cards[cards] = new HDSPMixerCard(Digiface, card); + cards++; + } else if (!strncmp(name, "RME Hammerfall HDSP 9652", 24)) { + printf("HDSP 9652 found !\n"); + hdsp_cards[cards] = new HDSPMixerCard(H9652, card); + cards++; + } else if (!strncmp(name, "RME Hammerfall HDSP 9632", 24)) { + printf("HDSP 9632 found !\n"); + hdsp_cards[cards] = new HDSPMixerCard(H9632, card); + cards++; + } else if (!strncmp(name, "RME MADIface", 12)) { printf("RME MADIface found !\n"); hdsp_cards[cards] = new HDSPMixerCard(HDSPeMADI, card); cards++; -- 1.7.2.3

2 2

[alsa-devel] ALSA: hda - 4930g switch lfe with side in mixer
by Łukasz Wojniłowicz 07 Feb '11

07 Feb '11

Hello, I'm sending patch that switches lfe with side in mixer for Acer Aspire 5930g. Currently I use Alsa 1.0.24 and if I want to change the volume of built-in subwoofer I have to adjust left channel of side in mixer, which isn't correct. I'm sending patch which corrects this behavior. BTW. If someone would like to investigate more then acer-aspire-4930g is ALC888S-VC. Cheers

2 1

[alsa-devel] [PATCH] ASoC: mid-x86: Fix dependency on intel_sst driver
by Koul, Vinod 07 Feb '11

07 Feb '11

From: Vinod Koul <vinod.koul(a)intel.com> Enabling medfield asoc driver causes compliation error when intel_sst is not selected ERROR: "register_sst_card" [sound/soc/mid-x86/snd-soc-sst-platform.ko] undefined! This patch puts proper dependency to elimate build error Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Reported-by: Andrew Morton <akpm(a)linux-foundation.org> --- sound/soc/mid-x86/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/mid-x86/Kconfig index 1ad7538..2935042 100644 --- a/sound/soc/mid-x86/Kconfig +++ b/sound/soc/mid-x86/Kconfig @@ -1,6 +1,7 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" depends on INTEL_SCU_IPC + depends on SND_INTEL_SST select SND_SOC_SN95031 select SND_SST_PLATFORM help @@ -11,4 +12,3 @@ config SND_MFLD_MACHINE config SND_SST_PLATFORM tristate - depends on SND_INTEL_SST -- 1.7.2.3

3 2

[alsa-devel] Nvidia HDMI and four codecs
by David Henningsson 07 Feb '11

07 Feb '11

I've seen a few Nvidia HDMI's recently which have four HDMI codecs connected to the same HDA controller. The problem is that we don't know which one(s) of these four that is the real one. So people will have to try devices "hdmi:Nvidia,0", "hdmi:Nvidia,1", "hdmi:Nvidia,2" and "hdmi:Nvidia,3" to figure out which one is the right one. (And by empirical testing we know that it isn't always the first one.) Thinking PulseAudio and/or "Just Works", this is not good enough, so is there a way to know beforehand which one of these four that is actually physically connected? I'd love BIOS to provide a good codec probe mask, a pin config default, or something similar, but given a codec-proc I can't see any difference between the real ones and the unconnected ones, so I don't know how to fix up the driver without hardcoding/quirking every one of them. Any hints? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 6

[alsa-devel] Conexant 5069: internal speakers are not muted when headphones are plugged in
by Paul Menzel 07 Feb '11

07 Feb '11

Dear ALSA folks, I encounter a problem with a Lenovo G555 where the internal speakers are not muted when headphones are plugged in. Ubuntu 10.10 with Linux 2.6.35 [1] runs the Lenovo, but this seems to be a common problem [2][3]. I tried to load the module with `model=lenovo`, but the only result was, that the external speakers did not work at all. Do you have another idea what to use for the `model`? The system has to rebooted every time, since I cannot unload the module using sudo modprobe -r snd-hda-intel or sudo rmmod snd-hda-intel since it seems to be in use by the system. Seems like I missed `model=thinkpad` in [3]. But I do not have access to the system right now, so I am sending this report anyway. Please find attached the output of `alsa-info.sh` with the default behavior (headphones work, but internal ones are not muted) and with `model=lenovo` (headphones do not work). David, do you know if your recent patches sent to this list [PATCH 2/2] ALSA: HDA: Add a new model "asus" for Conexant 5066/205xx [5] [PATCH] ALSA: HDA: Add Lenovo vendor quirk for Conexant 205xx [6] will also address this issue¹? They are for 5066 and I have 5069 according to [2]. If yes, is there a package repository with a package having this patches applied I can install using the package management system? Thanks, Paul ¹ Maybe you could add »5066« also to the commit message of the second patch. Since the data sheets seem to use this name.? But this string is not shown in `alsa-info.sh` so it seems it is not important. [1] http://packages.ubuntu.com/maverick-updates/linux-image-2.6.35-25-generic [2] http://www.insidesocal.com/click/2011/01/living-on-the-edge-with-the-li.html [3] http://www.insidesocal.com/click/2010/10/fedora-13-fixing-sound-and-vid.html [4] http://monespaceperso.org/blog-en/2010/05/02/upgrade-alsa-1-0-23-on-ubuntu-… [5] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-January/035954.ht… [6] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-January/035400.ht…

3 3

[alsa-devel] ALC861-VD bad recording issue
by Nowardev-Team 06 Feb '11

06 Feb '11

fixed my cards eats only 48000 with this ffmpeg -f alsa -ac 2 -ar 48000 -i hw:0,0 -s 1024x768 -f x11grab -r 30 -i :0.0 -acodec pcm_s16le -vcodec libx264 -vpre lossless_ultrafast -threads 0 output. works note ar 48000 with this ffmpeg -f alsa -ac 2 -i hw:0,0 -s 1024x768 -f x11grab -r 30 -i :0.0 -acodec pcm_s16le -vcodec libx264 -vpre lossless_ultrafast -threads 0 output. nope

1 0

[alsa-devel] Jack sense fails on Fujitsu p7120
by Robin Neatherway 06 Feb '11

06 Feb '11

I posted regarding this issue to alsa-users, but haven't had any response. I've come to the conclusion that I will have to modify the driver myself to rectify this problem, so I would appreciate some pointers as to how to get started with that. Running Xubuntu 10.10 with Alsa 1.0.23 The sound chip is Intel HDA, ALC260. I have tried setting options snd-hda-intel model=fujitsu in /etc/modprobe.d/alsa-base.conf, but this hasn't made any difference. There is no "Jack Sense" option in alsamixer. I've also tried a couple of values for fix_position. I've tried following the ubuntu wiki pages for Intel HDA and sound debugging in general, to no avail. When doing the jack sense test recommended on https://wiki.ubuntu.com/ALSA/JackSense the two files generated with and without the headphones plugged in were identical. This makes me think it is a driver problem. I pasted the output of `cat /proc/asound/card*/codec*` at http://pastebin.com/4NyvdsXf in case this is of any help.

3 15

[alsa-devel] ALC861-VD bad recording issue
by Nowardev-Team 05 Feb '11

05 Feb '11

This is the output of the alsa script to ge information about my system and the driver http://www.alsa-project.org/db/?f=9871b38d99bdd4cc91e398b60eb31361eb305395 <http://www.alsa-project.org/db/?f=9871b38d99bdd4cc91e398b60eb31361eb305395>the issue : when i record with mic i get always a bad recording with some tic tic tic tic , does not matter if i use low volume for the mic i can hear them anyway... if i use louder volume i can hear them ... i tried with arecord --quiet --file-type wav --rate=44100 -D plughw:Intel test.wav with this there is less rumors when i play with arecord --quiet --file-type wav --rate=44100 test.wav i got a lots of problems ... alsa (snd_pcm_recover) underrun occurred the strange thing is that on kubuntu 8.04 it was workin better ... that was my screenshot of alsamixer in the terminal http://nowardev.files.wordpress.com/2009/11/alsamixer.png now to get the " mic volume controller" i have to use options snd-hda-intel model=lenovo tried with all these lenovo is the only i can use ALC861VD/660VD ============== 3stack 3-jack 3stack-dig 3-jack with SPDIF OUT 6stack-dig 6-jack with SPDIF OUT 3stack-660 3-jack (for ALC660VD) 3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD) lenovo Lenovo 3000 C200 dallas Dallas laptops hp HP TX1000 asus-v1s ASUS V1Sn auto auto-config reading BIOS (default) but i get ... a lots of "controllers" like you can see here http://i0.simplest-image-hosting.net/168bf183b2abe8bc9188aacc163dd507/alsam… it seems that capture is the controller that can increase the mic volume ... :S i have tried every damned combinations of volumes but i get a lots of rumors always

1 0

[alsa-devel] issue with AD1989A HDA codec and alsamixer
by Vincent Meserette 05 Feb '11

05 Feb '11

Hi all, I am working on a system which includes a connexant AD1989A HDA codec connected to a ATOM processor. I have four microphones connected to the B and C ports of this codec. If I change one of the 3 capture gains present in the alsamixer (Capture, Capture 1 or Capture 3), I can't have any more signal on the B and C ports It seems to be due to the fact that ALSA breaks (for example for the Capture gain) the link between the ADC selector 0 and ADC_0 widgets of the codec, what can be easily seen with codecgraph. Have you ever had this kind of problem? Do you know how can I resolve it ? Many thanks for your asnwers and best regards Vincent

2 4

[alsa-devel] [PATCH] EDB93xx: Add support for CS4271 SPI-connected CODEC
by Alexander Sverdlin 04 Feb '11

04 Feb '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Add support for CS4271 SPI-connected CODEC to EDB93xx. Modified according to latest comments by H Hartley Sweeten. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- arch/arm/mach-ep93xx/edb93xx.c | 83 ++++++++++++++++++++++++++++++++++++++++ 1 files changed, 83 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-ep93xx/edb93xx.c b/arch/arm/mach-ep93xx/edb93xx.c index 3f37567..2c2f84d 100644 --- a/arch/arm/mach-ep93xx/edb93xx.c +++ b/arch/arm/mach-ep93xx/edb93xx.c @@ -30,9 +30,13 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/i2c-gpio.h> +#include <linux/spi/spi.h> + +#include <sound/cs4271.h> #include <mach/hardware.h> #include <mach/fb.h> +#include <mach/ep93xx_spi.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> @@ -94,6 +98,83 @@ static void __init edb93xx_register_i2c(void) /************************************************************************* + * EDB93xx SPI peripheral handling + *************************************************************************/ +static struct cs4271_platform_data edb93xx_cs4271_data = { + .gpio_nreset = -EINVAL, /* filled in later */ +}; + +static int edb93xx_cs4271_hw_setup(struct spi_device *spi) +{ + return gpio_request_one(EP93XX_GPIO_LINE_EGPIO6, + GPIOF_OUT_INIT_HIGH, spi->modalias); +} + +static void edb93xx_cs4271_hw_cleanup(struct spi_device *spi) +{ + gpio_free(EP93XX_GPIO_LINE_EGPIO6); +} + +static void edb93xx_cs4271_hw_cs_control(struct spi_device *spi, int value) +{ + gpio_set_value(EP93XX_GPIO_LINE_EGPIO6, value); +} + +static struct ep93xx_spi_chip_ops edb93xx_cs4271_hw = { + .setup = edb93xx_cs4271_hw_setup, + .cleanup = edb93xx_cs4271_hw_cleanup, + .cs_control = edb93xx_cs4271_hw_cs_control, +}; + +static struct spi_board_info edb93xx_spi_board_info[] __initdata = { + { + .modalias = "cs4271", + .platform_data = &edb93xx_cs4271_data, + .controller_data = &edb93xx_cs4271_hw, + .max_speed_hz = 6000000, + .bus_num = 0, + .chip_select = 0, + .mode = SPI_MODE_3, + }, +}; + +static struct ep93xx_spi_info edb93xx_spi_info __initdata = { + .num_chipselect = ARRAY_SIZE(edb93xx_spi_board_info), +}; + +static void __init edb93xx_register_spi(void) +{ + if (machine_is_edb9301() || machine_is_edb9302()) + edb93xx_cs4271_data.gpio_nreset = EP93XX_GPIO_LINE_EGPIO1; + else if (machine_is_edb9302a() || machine_is_edb9307a()) + edb93xx_cs4271_data.gpio_nreset = EP93XX_GPIO_LINE_H(2); + else if (machine_is_edb9315a()) + edb93xx_cs4271_data.gpio_nreset = EP93XX_GPIO_LINE_EGPIO14; + + ep93xx_register_spi(&edb93xx_spi_info, edb93xx_spi_board_info, + ARRAY_SIZE(edb93xx_spi_board_info)); +} + + +/************************************************************************* + * EDB93xx I2S + *************************************************************************/ +static int __init edb93xx_has_audio(void) +{ + return (machine_is_edb9301() || machine_is_edb9302() || + machine_is_edb9302a() || machine_is_edb9307a() || + machine_is_edb9315a()); +} + +static void __init edb93xx_register_i2s(void) +{ + if (edb93xx_has_audio()) { + ep93xx_register_i2s(); + } +} + + +/************************************************************************* * EDB93xx pwm *************************************************************************/ static void __init edb93xx_register_pwm(void) @@ -149,6 +230,8 @@ static void __init edb93xx_init_machine(void) edb93xx_register_flash(); ep93xx_register_eth(&edb93xx_eth_data, 1); edb93xx_register_i2c(); + edb93xx_register_spi(); + edb93xx_register_i2s(); edb93xx_register_pwm(); edb93xx_register_fb(); }

2 1

[alsa-devel] [PATCH] EDB93xx: Add support for CS4271 SPI-connected CODEC
by Alexander Sverdlin 04 Feb '11

04 Feb '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Add support for CS4271 SPI-connected CODEC to EDB93xx Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- arch/arm/mach-ep93xx/edb93xx.c | 85 ++++++++++++++++++++++++++++++++++++++++ 1 files changed, 85 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-ep93xx/edb93xx.c b/arch/arm/mach-ep93xx/edb93xx.c index 3f37567..c1e73ee 100644 --- a/arch/arm/mach-ep93xx/edb93xx.c +++ b/arch/arm/mach-ep93xx/edb93xx.c @@ -30,13 +30,17 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/i2c-gpio.h> +#include <linux/spi/spi.h> #include <mach/hardware.h> #include <mach/fb.h> +#include <mach/ep93xx_spi.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> +#include <sound/cs4271.h> + static void __init edb93xx_register_flash(void) { @@ -94,6 +98,85 @@ static void __init edb93xx_register_i2c(void) /************************************************************************* + * EDB93xx SPI peripheral handling + *************************************************************************/ +static struct cs4271_platform_data edb93xx_cs4271_data = { + .gpio_nreset = -EINVAL, /* filled in later */ +}; + +static int edb93xx_cs4271_hw_setup(struct spi_device *spi) +{ + return gpio_request_one(EP93XX_GPIO_LINE_EGPIO6, + GPIOF_OUT_INIT_HIGH, spi->modalias); +} + +static void edb93xx_cs4271_hw_cleanup(struct spi_device *spi) +{ + gpio_free(EP93XX_GPIO_LINE_EGPIO6); +} + +static void edb93xx_cs4271_hw_cs_control(struct spi_device *spi, int value) +{ + gpio_set_value(EP93XX_GPIO_LINE_EGPIO6, value); +} + +static struct ep93xx_spi_chip_ops edb93xx_cs4271_hw = { + .setup = edb93xx_cs4271_hw_setup, + .cleanup = edb93xx_cs4271_hw_cleanup, + .cs_control = edb93xx_cs4271_hw_cs_control, +}; + +static struct spi_board_info edb93xx_spi_board_info[] __initdata = { + { + .modalias = "cs4271", + .platform_data = &edb93xx_cs4271_data, + .controller_data = &edb93xx_cs4271_hw, + .max_speed_hz = 6000000, + .bus_num = 0, + .chip_select = 0, + .mode = SPI_MODE_3, + }, +}; + +static struct ep93xx_spi_info edb93xx_spi_info __initdata = { + .num_chipselect = ARRAY_SIZE(edb93xx_spi_board_info), +}; + +static void __init edb93xx_register_spi(void) +{ + if (machine_is_edb9301() || machine_is_edb9302()) + edb93xx_cs4271_data.gpio_nreset = EP93XX_GPIO_LINE_EGPIO1; + else if (machine_is_edb9302a() || machine_is_edb9307a()) + edb93xx_cs4271_data.gpio_nreset = EP93XX_GPIO_LINE_H(2); + else if (machine_is_edb9315a()) + edb93xx_cs4271_data.gpio_nreset = EP93XX_GPIO_LINE_EGPIO14; + else + return; + + ep93xx_register_spi(&edb93xx_spi_info, edb93xx_spi_board_info, + ARRAY_SIZE(edb93xx_spi_board_info)); +} + + +/************************************************************************* + * EDB93xx I2S + *************************************************************************/ +static int __init edb93xx_has_audio(void) +{ + return (machine_is_edb9301() || machine_is_edb9302() || + machine_is_edb9302a() || machine_is_edb9307a() || + machine_is_edb9315a()); +} + +static void __init edb93xx_register_i2s(void) +{ + if (edb93xx_has_audio()) { + ep93xx_register_i2s(); + } +} + + +/************************************************************************* * EDB93xx pwm *************************************************************************/ static void __init edb93xx_register_pwm(void) @@ -149,6 +232,8 @@ static void __init edb93xx_init_machine(void) edb93xx_register_flash(); ep93xx_register_eth(&edb93xx_eth_data, 1); edb93xx_register_i2c(); + edb93xx_register_spi(); + edb93xx_register_i2s(); edb93xx_register_pwm(); edb93xx_register_fb(); }

3 2

[alsa-devel] [PATCH] ASoC: ep93xx-ac97: remove extra empty line
by Mika Westerberg 04 Feb '11

04 Feb '11

Signed-off-by: Mika Westerberg <mika.westerberg(a)iki.fi> --- sound/soc/ep93xx/ep93xx-ac97.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c index 68a0bae..104e95c 100644 --- a/sound/soc/ep93xx/ep93xx-ac97.c +++ b/sound/soc/ep93xx/ep93xx-ac97.c @@ -253,7 +253,6 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, struct ep93xx_ac97_info *info = snd_soc_dai_get_drvdata(dai); unsigned v = 0; - switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: -- 1.7.2.3

3 2

[alsa-devel] [PATCH 2/3] ASoC: mid-x86 machine driver - add jack detect support
by Harsha Priya 04 Feb '11

04 Feb '11

This patch adds the jack detection in mid-86 machine driver Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/mid-x86/mfld_machine.c | 186 ++++++++++++++++++++++++++++++++------ 1 files changed, 159 insertions(+), 27 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 7925851..7abd3fc 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -27,6 +27,7 @@ #include <linux/init.h> #include <linux/device.h> #include <linux/slab.h> +#include <linux/io.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -35,10 +36,38 @@ #define MID_MONO 1 #define MID_STEREO 2 #define MID_MAX_CAP 5 +#define IRQ_BASE 0xFFFF7FCD +#define IRQ_SIZE 1 static unsigned int hs_switch; static unsigned int lo_dac; +struct mfld_mc_private { + struct platform_device *socdev; + struct workqueue_struct *mfld_jack_wq; + void __iomem *int_base; + struct snd_soc_codec *codec; +} mc_drv_ctx; + +struct snd_soc_jack mfld_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin mfld_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "AMIC1", + .mask = SND_JACK_MICROPHONE, + }, +}; +/* jack detection voltage zones */ +static struct snd_soc_jack_zone mfld_zones[] = { + {SN95031_VOL_START, SN95031_VOL_AM_HS, SND_JACK_HEADPHONE}, + {SN95031_VOL_AM_HS, SN95031_VOL_HS, SND_JACK_HEADSET}, +}; + /* sound card controls */ static const char *headset_switch_text[] = {"Earpiece", "Headset"}; @@ -67,13 +96,11 @@ static int headset_set_switch(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[0]) { pr_debug("hs_set HS path\n"); - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); } else { pr_debug("hs_set EP path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); } snd_soc_dapm_sync(&codec->dapm); @@ -91,12 +118,10 @@ static void lo_enable_out_pins(struct snd_soc_codec *codec) snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT"); snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT"); if (hs_switch) { - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); } else { - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); } } @@ -130,8 +155,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, case 1: pr_debug("set hs path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22); break; @@ -162,12 +186,36 @@ static const struct snd_kcontrol_new mfld_snd_controls[] = { lo_get_switch, lo_set_switch), }; +static const struct snd_soc_dapm_widget mfld_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route mfld_map[] = { + {"Headphones", NULL, "HPOUTR"}, + {"Headphones", NULL, "HPOUTL"}, + {"Mic", NULL, "AMIC1"}, +}; + static int mfld_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_codec *codec = runtime->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; int ret_val; + /* Add jack sense widgets */ + snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets)); + + /* Set up the map */ + snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map)); + + /* always connected */ + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Mic"); + snd_soc_dapm_sync(dapm); + + mc_drv_ctx.codec = codec; + ret_val = snd_soc_add_controls(codec, mfld_snd_controls, ARRAY_SIZE(mfld_snd_controls)); if (ret_val) { @@ -175,8 +223,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) return ret_val; } /* default is earpiece pin, userspace sets it explcitly */ - snd_soc_dapm_disable_pin(dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(dapm, "Headphones"); /* default is lineout NC, userspace sets it explcitly */ snd_soc_dapm_disable_pin(dapm, "LINEOUTL"); snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); @@ -185,7 +232,32 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) /* we dont use linein in this so set to NC */ snd_soc_dapm_disable_pin(dapm, "LINEINL"); snd_soc_dapm_disable_pin(dapm, "LINEINR"); - return snd_soc_dapm_sync(dapm); + snd_soc_dapm_sync(dapm); + + /* Headset jack detection */ + ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1, &mfld_jack); + + if (ret_val) { + pr_err("jack creation failed\n"); + return ret_val; + } + + ret_val = snd_soc_jack_add_pins(&mfld_jack, + ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins); + if (ret_val) { + pr_err("adding jack pins failed\n"); + return ret_val; + } + ret_val = snd_soc_jack_add_zones(&mfld_jack, + ARRAY_SIZE(mfld_zones), mfld_zones); + if (ret_val) { + pr_err("adding jack zones failed\n"); + return ret_val; + } + disable_jack_btn(codec); + return ret_val; } struct snd_soc_dai_link mfld_msic_dailink[] = { @@ -234,37 +306,97 @@ static struct snd_soc_card snd_soc_card_mfld = { .num_links = ARRAY_SIZE(mfld_msic_dailink), }; +void snd_mfld_jack_detection(struct work_struct *work) +{ + struct mfld_jack_msg_wq *jack_msg = + container_of(work, struct mfld_jack_msg_wq, wq); + + sn95031_jack_detection(&mfld_jack, jack_msg); + kfree(jack_msg); +} + +static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev) +{ + struct mfld_jack_msg_wq *jack_msg; + struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev; + + jack_msg = kzalloc(sizeof(*jack_msg), GFP_ATOMIC); + memcpy_fromio(&jack_msg->intr_id, + ((void *)(mc_private->int_base)), + sizeof(u8)); + jack_msg->codec = mc_private->codec; + pr_debug("snd_mfld_jack_intr_handler called value %x\n", + jack_msg->intr_id); + INIT_WORK(&jack_msg->wq, snd_mfld_jack_detection); + queue_work(mc_private->mfld_jack_wq, &jack_msg->wq); + + return IRQ_HANDLED; +} + static int __devinit snd_mfld_mc_probe(struct platform_device *pdev) { - struct platform_device *socdev; - int ret_val = 0; + int ret_val = 0, irq; pr_debug("snd_mfld_mc_probe called\n"); - socdev = platform_device_alloc("soc-audio", -1); - if (!socdev) { + /* retrive the irq number */ + irq = platform_get_irq(pdev, 0); + /* audio interrupt base of SRAM location where + * interrupts are stored by System FW */ + mc_drv_ctx.int_base = ioremap_nocache(IRQ_BASE, IRQ_SIZE); + if (!mc_drv_ctx.int_base) { + pr_err("Mapping of cache failed\n"); + ret_val = -ENOMEM; + goto unalloc; + } + /* create botoom half handler for interrupt */ + mc_drv_ctx.mfld_jack_wq = create_workqueue("mfld_jack_wq"); + if (!mc_drv_ctx.mfld_jack_wq) { + pr_err("cannot create jack workqueue\n"); + ret_val = -ENOMEM; + goto unalloc; + } + /* register for interrupt */ + ret_val = request_irq(irq, snd_mfld_jack_intr_handler, + IRQF_SHARED, pdev->dev.driver->name, + &mc_drv_ctx); + if (ret_val) { + pr_err("cannot register IRQ\n"); + goto destroy_wq; + } + /* create soc device */ + mc_drv_ctx.socdev = platform_device_alloc("soc-audio", -1); + if (!mc_drv_ctx.socdev) { pr_err("soc-audio device allocation failed\n"); - return -ENOMEM; + ret_val = -ENOMEM; + goto destroy_wq; } - platform_set_drvdata(socdev, &snd_soc_card_mfld); - ret_val = platform_device_add(socdev); + platform_set_drvdata(mc_drv_ctx.socdev, &snd_soc_card_mfld); + ret_val = platform_device_add(mc_drv_ctx.socdev); if (ret_val) { pr_err("Unable to add soc-audio device, err %d\n", ret_val); - platform_device_put(socdev); + goto unregister; } - - platform_set_drvdata(pdev, socdev); - + platform_set_drvdata(pdev, &mc_drv_ctx); pr_debug("successfully exited probe\n"); return ret_val; + +unregister: + platform_device_put(mc_drv_ctx.socdev); +destroy_wq: + destroy_workqueue(mc_drv_ctx.mfld_jack_wq); +unalloc: + return ret_val; } static int __devexit snd_mfld_mc_remove(struct platform_device *pdev) { - struct platform_device *socdev = platform_get_drvdata(pdev); - pr_debug("snd_mfld_mc_remove called\n"); + struct mfld_mc_private *mc_private_data = platform_get_drvdata(pdev); - platform_device_unregister(socdev); + pr_debug("snd_mfld_mc_remove called\n"); + free_irq(platform_get_irq(pdev, 0), mc_private_data); + destroy_workqueue(mc_private_data->mfld_jack_wq); + platform_device_unregister(mc_private_data->socdev); platform_set_drvdata(pdev, NULL); return 0; } -- 1.7.3.4

3 3

[alsa-devel] [PATCH 3/3] ASoC: sn95031 codec - adding jack detection/reporting
by Harsha Priya 04 Feb '11

04 Feb '11

This patch adds the jack detection logic in sn95031 codec. This also adds code for reading the ADC to figure out the mic bias and therby report jack type detected Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/sn95031.c | 185 +++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/sn95031.h | 56 +++++++++++++ 2 files changed, 237 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 40e285d..07b6ea7 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -27,6 +27,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> +#include <linux/input.h> #include <asm/intel_scu_ipc.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -39,12 +40,129 @@ #define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) #define SN95031_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) +/* adc helper functions */ + +/* enables mic bias voltage */ +void sn95031_enable_mic_bias(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_VAUD, BIT(2)|BIT(1)|BIT(0)); + snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(2), BIT(2)); +} + +/* Enable/Disable the ADC depending on the argument */ +static void configure_adc(struct snd_soc_codec *sn95031_codec, int val) +{ + int value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); + + if (val) { + /* Enable and start the ADC */ + value |= (SN95031_ADC_ENBL | SN95031_ADC_START); + value &= (~ADC_NO_LOOP); + } else + /* Just stop the ADC */ + value &= (~SN95031_ADC_START); + snd_soc_write(sn95031_codec, SN95031_ADC1CNTL1, value); +} + /* - * todo: - * capture paths - * jack detection - * PM functions + * finds an empty channel for conversion + * If the ADC is not enabled then start using 0th channel + * itself. Otherwise find an empty channel by looking for a + * channel in which the stopbit is set to 1. returns the index + * of the first free channel if succeeds or an error code. + * + * Context: can sleep + * */ +static int find_free_channel(struct snd_soc_codec *sn95031_codec) +{ + int ret = 0, i, value; + + /* check whether ADC is enabled */ + value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); + + if ((value & SN95031_ADC_ENBL) == 0) + return 0; + + /* ADC is already enabled; Looking for an empty channel */ + for (i = 0; i < ADC_CHANLS_MAX; i++) { + value = snd_soc_read(sn95031_codec, ADC_CHNL_START_ADDR + i); + if (value & SN95031_STOPBIT_MASK) { + ret = i; + break; + } + } + return (ret > ADC_LOOP_MAX) ? (-EINVAL) : ret; +} + +/* Initialize the ADC for reading thermistor values. Can sleep. */ +static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec) +{ + int base_addr, chnl_addr; + int value; + static int channel_index; + + /* Index of the first channel in which the stop bit is set */ + channel_index = find_free_channel(sn95031_codec); + if (channel_index < 0) { + pr_err("No free ADC channels"); + return channel_index; + } + + base_addr = ADC_CHNL_START_ADDR + channel_index; + + if (!(channel_index == 0 || channel_index == ADC_LOOP_MAX)) { + /* Reset stop bit for channels other than 0 and 12 */ + value = snd_soc_read(sn95031_codec, base_addr); + /* Set the stop bit to zero */ + snd_soc_write(sn95031_codec, base_addr, value & 0xEF); + /* Index of the first free channel */ + base_addr++; + channel_index++; + } + + /* Since this is the last channel, set the stop bit + to 1 by ORing the DIE_SENSOR_CODE with 0x10 */ + snd_soc_write(sn95031_codec, base_addr, AUDIO_DETECT_CODE | 0x10); + + chnl_addr = ADC_DATA_START_ADDR + 2 * channel_index; + pr_debug("mid_initialize : %x", chnl_addr); + configure_adc(sn95031_codec, 1); + return chnl_addr; +} + + +/* reads the ADC registers and gets the mic bias value in mV. */ +static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) +{ + u16 adc_adr = sn95031_initialize_adc(codec); + u16 adc_val1, adc_val2; + unsigned int mic_bias; + + sn95031_enable_mic_bias(codec); + + /* Enable the sound card for conversion before reading */ + snd_soc_write(codec, SN95031_ADC1CNTL3, 0x05); + /* Re-toggle the RRDATARD bit */ + snd_soc_write(codec, SN95031_ADC1CNTL3, 0x04); + + /* Read the higher bits of data */ + msleep(1000); + adc_val1 = snd_soc_read(codec, adc_adr); + adc_adr++; + adc_val2 = snd_soc_read(codec, adc_adr); + + /* Adding lower two bits to the higher bits */ + mic_bias = (adc_val1 << 2) + (adc_val2 & 3); + mic_bias = (mic_bias * ADC_ONE_LSB_MULTIPLIER) / 1000; + pr_debug("mic bias = %dmV\n", mic_bias); + return mic_bias; +} +EXPORT_SYMBOL_GPL(sn95031_get_mic_bias); + +/*end - adc helper functions */ + +/* codec read/write functions */ static inline unsigned int sn95031_read(struct snd_soc_codec *codec, unsigned int reg) @@ -70,6 +188,8 @@ static inline int sn95031_write(struct snd_soc_codec *codec, return ret; } +/* end - codec read/write functions */ + static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -649,6 +769,63 @@ struct snd_soc_dai_driver sn95031_dais[] = { }, }; +int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack, + struct snd_soc_codec *codec) +{ + int micbias = sn95031_get_mic_bias(codec); + + int jack_type = snd_soc_jack_get_type(mfld_jack, micbias); + + /* If american headest, program the gpio control registers */ + if (micbias >= SN95031_VOL_HP && micbias < SN95031_VOL_AM_HS) + snd_soc_update_bits(codec, AUDIO_GPIO_CTRL, BIT(1), BIT(1)); + else + snd_soc_update_bits(codec, AUDIO_GPIO_CTRL, BIT(1), BIT(0)); + + pr_debug("jack type detected = %d\n", jack_type); + if (jack_type == SND_JACK_HEADSET) + enable_jack_btn(codec); + return jack_type; +} + +void sn95031_jack_detection(struct snd_soc_jack *mfld_jack, + struct mfld_jack_msg_wq *jack_msg) +{ + unsigned int status, mask; + static int jack_state = SND_JACK_HEADSET; + struct snd_soc_codec *codec = jack_msg->codec; + + if (!codec) { + pr_err("codec is null\n"); + return; + } + pr_debug("interrupt id read in sram = 0x%x\n", jack_msg->intr_id); + if (jack_msg->intr_id & 0x1) { + pr_debug("short_push detected\n"); + mask = status = SND_JACK_BTN_0; + } else if (jack_msg->intr_id & 0x2) { + pr_debug("long_push detected\n"); + mask = status = SND_JACK_BTN_1; + } else if (jack_msg->intr_id & 0x4) { + mask = status = sn95031_get_headset_state(mfld_jack, codec); + jack_state = mask; + } else if (jack_msg->intr_id & 0x8) { + pr_debug("headset or headphones removed\n"); + status = 0; + mask = jack_state; + disable_jack_btn(codec); + } else { + pr_err("unidentified interrupt\n"); + return; + } + + snd_soc_jack_report(mfld_jack, status, mask); + /*button pressed and released */ + if (status == SND_JACK_BTN_0 || status == SND_JACK_BTN_1) + snd_soc_jack_report(mfld_jack, 0, mask); +} +EXPORT_SYMBOL_GPL(sn95031_jack_detection); + /* codec registration */ static int sn95031_codec_probe(struct snd_soc_codec *codec) { diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index e2b17d9..dbae58d 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -25,6 +25,7 @@ */ #ifndef _SN95031_H #define _SN95031_H +#include <sound/jack.h> /*register map*/ #define SN95031_VAUD 0xDB @@ -96,4 +97,59 @@ #define SN95031_SSR5 0x384 #define SN95031_SSR6 0x385 +/* ADC registers */ + +#define SN95031_ADC1CNTL1 0x1C0 +#define SN95031_ADC_ENBL 0x10 +#define SN95031_ADC_START 0x08 +#define SN95031_ADC1CNTL3 0x1C2 +#define SN95031_ADCTHERM_ENBL 0x04 +#define SN95031_ADCRRDATA_ENBL 0x05 +#define SN95031_STOPBIT_MASK 16 +#define SN95031_ADCTHERM_MASK 4 +#define ADC_CHANLS_MAX 15 /* Number of ADC channels */ +#define ADC_LOOP_MAX (ADC_CHANLS_MAX - 1) +#define ADC_NO_LOOP 0x07 + +/* ADC channel code values */ +#define AUDIO_DETECT_CODE 0x06 +#define AUDIO_GPIO_CTRL 0x070 +/* ADC base addresses */ +#define ADC_CHNL_START_ADDR 0x1C5 /* increments by 1 */ +#define ADC_DATA_START_ADDR 0x1D4 /* increments by 2 */ +/* multipier to convert to mV */ +#define ADC_ONE_LSB_MULTIPLIER 2346 + +enum soc_mic_bias_zones { + SN95031_VOL_START = 0, + /* mic bias volutage range for Headphones*/ + SN95031_VOL_HP = 400, + /* mic bias volutage range for American Headset*/ + SN95031_VOL_AM_HS = 650, + /* mic bias volutage range for Headset*/ + SN95031_VOL_HS = 2000, + SN95031_VOL_UNDEFINED, +}; + + +struct mfld_jack_msg_wq { + u8 intr_id; + struct snd_soc_codec *codec; + struct work_struct wq; +}; + +static inline void disable_jack_btn(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_BTNCTRL2, 0x00); +} + +static inline void enable_jack_btn(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_BTNCTRL1, 0x77); + snd_soc_write(codec, SN95031_BTNCTRL2, 0x01); +} + +extern void sn95031_jack_detection(struct snd_soc_jack *mfld_jack, + struct mfld_jack_msg_wq *jack_msg); + #endif -- 1.7.3.4

3 6

Re: [alsa-devel] [Alsa-devel] how to cross compile the alsa util?
by Leonid 04 Feb '11

04 Feb '11

Hello alls: Please help. I also cross compiled alsa utils trhough CodeSourcery: PATH=$PATH:/opt/arm-2007q1/bin CC=arm-none-linux-gnueabi-gcc ./configure --host=i686-pc-linux-gnu \ --target=arm-none-linux-gnueabi \ --with-alsa-prefix=/usr/lib \ --with-alsa-inc-prefix=/usr/include/alsa configure out... --cut checking for ncursesw5-config... yes checking for curses library... ncursesw checking for curses header name...<ncurses.h> checking for curses compiler flags... -I/usr/include/ncursesw checking for curses NLS support... yes checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking panel.h usability... no checking panel.h presence... no checking for panel.h... no configure: error: required curses helper header not found but libncursesw5 and aibncursesw5-dev is installed to host: more... I changet the configuration and add option --with-curses=ncurses (libncurses5-dev also installed): PATH=$PATH:/opt/arm-2007q1/bin CC=arm-none-linux-gnueabi-gcc ./configure --host=i686-pc-linux-gnu \ --target=arm-none-linux-gnueabi \ --with-alsa-prefix=/usr/lib \ --with-alsa-inc-prefix=/usr/include/alsa \ --with-curses=ncurses out... --cut checking for ncurses5-config... yes checking for curses library... ncurses checking for curses header name...<ncurses.h> checking for curses compiler flags... checking for curses NLS support... no checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking panel.h usability... no checking panel.h presence... no checking for panel.h... no configure: error: required curses helper header not found the same effect. what am i doing wrong? can be cross compiled ncurses? -- with best wishes Leonid On Mon, 31 May 2004, Roc Wu wrote: > Hello alls: > > Sorry for post the mail again. I posted it several > days ago, but no response. Anybody can give me some > hints? > > I cross compiled the alsa-lib-1.04 to arm platform. > The 2.6.6 kernel including alsa driver is ok on our > ARM board, so I want to test the driver. how to do it? > > I think maybe the first step is to crosscompile the > alsa utils to play a wave audio file. > > CC=arm-linux-gcc ./configure --host=i686-linux > --target=arm-linux ./configure --help --- cut --- --with-alsa-prefix=PFX Prefix where Alsa library is installed (optional) --with-alsa-inc-prefix=PFX Prefix where include libraries are (optional) --- cut --- Jaroslav ----- Jaroslav Kysela<[EMAIL PROTECTED]> Linux Kernel Sound Maintainer ALSA Project, SuSE Labs

2 1

[alsa-devel] [PATCH 12/19] twl4030: mfd_cell is now implicitly available to drivers
by Andres Salomon 04 Feb '11

04 Feb '11

No need to explicitly set the cell's platform_data/data_size. In this case, move the various platform_data pointers to driver_data. All of the clients which make use of it are also changed. Signed-off-by: Andres Salomon <dilinger(a)queued.net> --- drivers/input/misc/twl4030-vibra.c | 2 +- drivers/mfd/twl4030-codec.c | 6 ++---- sound/soc/codecs/twl4030.c | 9 ++++++--- 3 files changed, 9 insertions(+), 8 deletions(-) diff --git a/drivers/input/misc/twl4030-vibra.c b/drivers/input/misc/twl4030-vibra.c index 014dd4a..7d7602a 100644 --- a/drivers/input/misc/twl4030-vibra.c +++ b/drivers/input/misc/twl4030-vibra.c @@ -196,7 +196,7 @@ static SIMPLE_DEV_PM_OPS(twl4030_vibra_pm_ops, static int __devinit twl4030_vibra_probe(struct platform_device *pdev) { - struct twl4030_codec_vibra_data *pdata = pdev->dev.platform_data; + struct twl4030_codec_vibra_data *pdata = platform_get_drvdata(pdev); struct vibra_info *info; int ret; diff --git a/drivers/mfd/twl4030-codec.c b/drivers/mfd/twl4030-codec.c index 9a4b196..d44ad30 100644 --- a/drivers/mfd/twl4030-codec.c +++ b/drivers/mfd/twl4030-codec.c @@ -208,15 +208,13 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) if (pdata->audio) { cell = &codec->cells[childs]; cell->name = "twl4030-codec"; - cell->platform_data = pdata->audio; - cell->data_size = sizeof(*pdata->audio); + cell->driver_data = pdata->audio; childs++; } if (pdata->vibra) { cell = &codec->cells[childs]; cell->name = "twl4030-vibra"; - cell->platform_data = pdata->vibra; - cell->data_size = sizeof(*pdata->vibra); + cell->driver_data = pdata->vibra; childs++; } diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e4d464b..3721671 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -130,6 +130,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { /* codec private data */ struct twl4030_priv { struct snd_soc_codec codec; + struct twl4030_codec_audio_data *pdata; unsigned int codec_powered; @@ -297,8 +298,8 @@ static inline void twl4030_reset_registers(struct snd_soc_codec *codec) static void twl4030_init_chip(struct snd_soc_codec *codec) { - struct twl4030_codec_audio_data *pdata = dev_get_platdata(codec->dev); struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); + struct twl4030_codec_audio_data *pdata = twl4030->pdata; u8 reg, byte; int i = 0; @@ -732,9 +733,9 @@ static int aif_event(struct snd_soc_dapm_widget *w, static void headset_ramp(struct snd_soc_codec *codec, int ramp) { - struct twl4030_codec_audio_data *pdata = codec->dev->platform_data; unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); + struct twl4030_codec_audio_data *pdata = twl4030->pdata; /* Base values for ramp delay calculation: 2^19 - 2^26 */ unsigned int ramp_base[] = {524288, 1048576, 2097152, 4194304, 8388608, 16777216, 33554432, 67108864}; @@ -2251,6 +2252,7 @@ static int twl4030_soc_resume(struct snd_soc_codec *codec) static int twl4030_soc_probe(struct snd_soc_codec *codec) { + struct twl4030_codec_audio_data *pdata = dev_get_drvdata(codec->dev); struct twl4030_priv *twl4030; twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL); @@ -2258,6 +2260,7 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) printk("Can not allocate memroy\n"); return -ENOMEM; } + twl4030->pdata = pdata; snd_soc_codec_set_drvdata(codec, twl4030); /* Set the defaults, and power up the codec */ twl4030->sysclk = twl4030_codec_get_mclk() / 1000; @@ -2297,7 +2300,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { static int __devinit twl4030_codec_probe(struct platform_device *pdev) { - struct twl4030_codec_audio_data *pdata = pdev->dev.platform_data; + struct twl4030_codec_audio_data *pdata = platform_get_drvdata(pdev); if (!pdata) { dev_err(&pdev->dev, "platform_data is missing\n"); -- 1.7.2.3

5 9

[alsa-devel] [PATCH] ALSA: HDA: Fix microphone(s) on Lenovo Edge 13
by David Henningsson 04 Feb '11

04 Feb '11

This Edge 13 model has an internal mic at 0x1a and should therefore use the asus quirk. Since my backport of the asus model is now tested and working, would it be possible to apply that patch as well as this quirk to 2.6.38? Thanks Takashi for keeping up with all the patches coming in! :-) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

3 11

[alsa-devel] [PATCH 1/3] ASoC: soc_jack - add function to add jack zones and determine jack type
by Harsha Priya 04 Feb '11

04 Feb '11

Based on the voltage value in adc channel,jack plugged in needs to be determined This patch adds a helper functions that adds the adc voltage zones and determines the jack type given the mic bias value. When a jack is created, the voltage zones can to be added to it. Later when a jack is inserted, the micbias voltage can be measured and used to find the jack type For any codec that would give the mic bias value on a jack insertion, this function would return the type of jack based on mic bias range Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/soc.h | 20 +++++++++++++++ sound/soc/soc-jack.c | 67 ++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 87 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4b6c0a8..97f2a3a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -234,6 +234,7 @@ struct snd_soc_codec; struct snd_soc_codec_driver; struct soc_enum; struct snd_soc_jack; +struct snd_soc_jack_zone; struct snd_soc_jack_pin; struct snd_soc_cache_ops; #include <sound/soc-dapm.h> @@ -307,6 +308,9 @@ void snd_soc_jack_notifier_register(struct snd_soc_jack *jack, struct notifier_block *nb); void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack, struct notifier_block *nb); +int snd_soc_jack_add_zones(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_zone *zones); +int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage); #ifdef CONFIG_GPIOLIB int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); @@ -406,6 +410,21 @@ struct snd_soc_jack_pin { bool invert; }; +/* struct snd_soc_jack_zone - Describes voltage zones of jack detection + * + * @adc_value_start: start range of voltage value in adc channel + * when jack is pluggedin + * @adc_value_end: end range of voltage value in adc channel + * when jack is pluggedin + * @jack_type: type of jack that is expected for this voltage + */ +struct snd_soc_jack_zone { + unsigned int adc_value_start; + unsigned int adc_value_end; + unsigned int jack_type; + struct list_head list; +}; + /** * struct snd_soc_jack_gpio - Describes a gpio pin for jack detection * @@ -435,6 +454,7 @@ struct snd_soc_jack { struct list_head pins; int status; struct blocking_notifier_head notifier; + struct list_head jack_zones; }; /* SoC PCM stream information */ diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ac5a5bc..66bd68f 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -37,6 +37,7 @@ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, { jack->codec = codec; INIT_LIST_HEAD(&jack->pins); + INIT_LIST_HEAD(&jack->jack_zones); BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); return snd_jack_new(codec->card->snd_card, id, type, &jack->jack); @@ -111,6 +112,72 @@ out: } EXPORT_SYMBOL_GPL(snd_soc_jack_report); +static int snd_soc_validate_jack_type(int type) +{ + switch (type) { + case SND_JACK_HEADPHONE: + case SND_JACK_MICROPHONE: + case SND_JACK_HEADSET: + case SND_JACK_LINEOUT: + case SND_JACK_MECHANICAL: + case SND_JACK_VIDEOOUT: + case SND_JACK_AVOUT: + break; + default: + return -EINVAL; + + }; + return 0; +} + +/** + * snd_soc_jack_add_zones - Associate voltage zones with jack + * + * @jack: ASoC jack + * @count: Number of zones + * @zone: Array of zones + * + * After this function has been called the zones specified in the + * array will be associated with the jack. + */ +int snd_soc_jack_add_zones(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_zone *zones) +{ + int i; + + for (i = 0; i < count; i++) { + if (snd_soc_validate_jack_type(zones[i].jack_type)) { + printk(KERN_ERR "Unknown jack type given %d\n", i); + return -EINVAL; + } + INIT_LIST_HEAD(&zones[i].list); + list_add(&(zones[i].list), &jack->jack_zones); + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_add_zones); + +/** + * snd_soc_jack_get_type - Based on the mic bias value, this function returns + * the type of jack from the zones delcared in the jack type + * + * @micbias_voltage: mic bias voltage at adc channel when jack is plugged in + * + * Based on the mic bias value passed, this function helps identify + * the type of jack from the already delcared jack zones + **/ +int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage) +{ + struct snd_soc_jack_zone *zone; + + list_for_each_entry(zone, &jack->jack_zones, list) { + if (micbias_voltage >= zone->adc_value_start && + micbias_voltage < zone->adc_value_end) + return zone->jack_type; + } + return -ENODEV; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_get_type); /** * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack * -- 1.7.3.4

3 2

[alsa-devel] NVIDIA HDMI surround sound is broken starting with kernel 2.6.36
by VDR User 04 Feb '11

04 Feb '11

The behavior described below started with 2.6.36 and follows through up to the most recent stable kernel, 2.6.37. The last properly working driver is in kernel 2.6.35.9. I should also mention that while the tests below use the static_hdmi_pcm flag recently introduced to alsa git, I also tested with the in-kernel drivers for each stable kernel. The particular device used for testing is an Nvidia GT220 video card connected directly to my receiver via HDMI. Can someone please look into this? Tested: cold boot, kernel 2.6.37, alsa snapshot jan.25,2010, static_hdmi_pcm=1, receiver ON cold boot, kernel 2.6.37, alsa snapshot jan.25,2010, static_hdmi_pcm=1, receiver OFF cold boot, kernel 2.6.37, alsa snapshot jan.25,2010, static_hdmi_pcm=0, receiver ON cold boot, kernel 2.6.37, alsa snapshot jan.25,2010, static_hdmi_pcm=0, receiver OFF In all cases only the front-left and front-right speakers worked. LFE, rear-left, rear-right, and center were missing. ELD correctly reported channels corresponding to receiver ON/OFF. (receiver on=7.1, receiver off=tv stereo). Also, even though the eld file is removed, the eld info is _not_ updated when stopping alsa & unloading drivers, then reloading. A reboot is required to get the eld info to update. This is why I cold boot the pc for each test. With kernel 2.6.35.9 drivers, front-left & front-right worked with receiver OFF as expected. I then turn the receiver on and all channels worked. I didn't bother to stop alsa, reboot, or even look at eld info since it works. Thanks for any & all help, Derek

2 3

[alsa-devel] [PATCH - hdspmixer 0/4] Beautification patches
by Adrian Knoth 03 Feb '11

03 Feb '11

Hi! Here's a line of code beautification patches for hdspmixer. I broke them up into smaller pieces to ease the review, though one may find four patches for no added functionality a little bit excessive. If so, please feel free simply commit them as a single commit. Cheers Adrian Knoth (4): hdspmixer: retab and reindent code. hdspmixer: Improve readability. hdspmixer: Beautification; Lower nesting depth hdspmixer: Break overlong line and don't plenken. -- 1.7.2.3

1 4

[alsa-devel] QasMixer 0.7.0 released (former Alsamixer-Qt4)
by Sebastian H. 03 Feb '11

03 Feb '11

Hello ALSA users A new version 0.7.0 of QasMixer (formerly Alsamixer-Qt4) is now available at http://xwmw.org/qasmixer or http://sourceforge.net/projects/qasmixer/ QasMixer is a fusion of Alsamixer-Qt4 and AlsaInfo-Qt4. Both will be discontinued because it's easier for me to support just one larger application instead of two smaller ones. Most notably for end users will be the improved desktop integration with single instance mode, a system tray icon, a qasmixer.desktop file for application menus and automatic desktop size detection which should play well with netbooks. Regards, Sebastian Holtermann

2 1

[alsa-devel] [PATCH v2] asoc: davinci: da830/omap-l137: correct cpu_dai_name
by Vaibhav Bedia 03 Feb '11

03 Feb '11

McASP1 is used on the DA830/OMAP-L137 platform for the codec. This is different from the DA850/OMAP-L138 platform which uses McASP0. This is fixed by adding a new snd_soc_dai_link struct. Signed-off-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> --- v2: Fixed the subject as pointed out by Sergei. sound/soc/davinci/davinci-evm.c | 18 +++++++++++++++--- 1 files changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index b36f0b3..fe79842 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -218,7 +218,19 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .ops = &evm_spdif_ops, }, }; -static struct snd_soc_dai_link da8xx_evm_dai = { + +static struct snd_soc_dai_link da830_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name = "davinci-mcasp.1", + .codec_dai_name = "tlv320aic3x-hifi", + .codec_name = "tlv320aic3x-codec.1-0018", + .platform_name = "davinci-pcm-audio", + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +static struct snd_soc_dai_link da850_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", @@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { static struct snd_soc_card da830_snd_soc_card = { .name = "DA830/OMAP-L137 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da830_evm_dai, .num_links = 1, }; static struct snd_soc_card da850_snd_soc_card = { .name = "DA850/OMAP-L138 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da850_evm_dai, .num_links = 1, }; -- 1.6.2.4

3 2

[alsa-devel] [PATCH 1/2] ASoC: Create an AIF1ADCDAT signal widget to match AIF2
by Dimitris Papastamos 03 Feb '11

03 Feb '11

From: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Due to the different routing for AIF1 and AIF2 we weren't using a single widget to represent the ADCDAT signal. For consistency add one. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 14 ++++++++++---- 1 files changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0ca81d3..f53b5a9 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1287,9 +1287,9 @@ SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 9, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, 0, WM8994_POWER_MANAGEMENT_4, 8, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev, @@ -1298,9 +1298,9 @@ SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0, WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), -SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 11, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL, 0, WM8994_POWER_MANAGEMENT_4, 10, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev, @@ -1345,6 +1345,7 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux), @@ -1546,6 +1547,11 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" }, { "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" }, + { "AIF1ADCDAT", NULL, "AIF1ADC1L" }, + { "AIF1ADCDAT", NULL, "AIF1ADC1R" }, + { "AIF1ADCDAT", NULL, "AIF1ADC2L" }, + { "AIF1ADCDAT", NULL, "AIF1ADC2R" }, + { "AIF2ADCDAT", NULL, "AIF2ADC Mux" }, /* AIF3 output */ -- 1.7.3.5

2 2

[alsa-devel] [PATCH] ASoC: Sort ALC5623 in Kconfig and Makefile
by Mark Brown 03 Feb '11

03 Feb '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Makefile | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 06894da..46ffb26 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -92,6 +92,7 @@ obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o @@ -103,7 +104,6 @@ obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o -obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o -- 1.7.2.3

2 1

[alsa-devel] UCM list/set/get
by pl bossart 03 Feb '11

03 Feb '11

I've been trying to figure out what exactly needs to be filled in UCM configuration files, and so far it's been a pretty frustrating experience with alsa-utils/alsaucm :-( The interactive mode in alsa-utils/alsaucm is broken, I fixed a couple of segfaults and bad output (patches attached), but so far the only thing I can do is list the verbs...It looks like alsa-lib/include/use-case.h and alsa-utils/alsaucm are not aligned, whenever I query with predefined identifiers alsaucm seg faults or provides an error message that doesn't help much. This happens with the latest code. [ume@plb alsaucm]$ ./alsaucm listcards 0: AudioPC this is my AudioPCI card 1: SDP4430 this is my SDP4430 card 2: TestHDA A test HDA card [ume@plb alsaucm]$ ./alsaucm -c AudioPCI list _verbs 0: HiFi Play and record HiFi quality Music. 1: Voice Full duplex low-power voice call So far so good. Now the fun begins [ume@plb alsaucm]$ ./alsaucm -c AudioPCI list _devices ./alsaucm: error failed to get list _devices: No such file or directory [ume@plb alsaucm]$ ./alsaucm -c AudioPCI set _verb HiFi ALSA lib main.c:1250:(set_verb_user) error: failed to initialize new use case: HiFi ./alsaucm: error failed to set _verb=HiFi: No such device [ume@plb alsaucm]$ ./alsaucm -c AudioPCI get _verb ./alsaucm: error failed to get _verb: No such file or directory Likewise if the .conf file contains a cset in the default values, then the parser fails (I took the example from src/ucm/parser.c). Commenting out the cset is the only solution. What exactly is this index supposed to mean? SectionDefaults [ cdev "hw:AudioPCI" cset "name='Master Playback Switch',index=2 1,1" ] [ume@plb alsaucm]$ ./alsaucm -c AudioPCI list _verbs ALSA lib main.c:302:(import_master_config) Unable to execute default sequence ALSA lib main.c:637:(snd_use_case_mgr_open) error: failed to import AudioPCI use case configuration -19 ./alsaucm: error failed to open sound card AudioPCI: No such device Ideally I'd like to write an interface between PulseAudio and UCM, there are existing modules I can reuse to set voice or hifi modes, depending on what the client roles are. But if I can't make the examples work I don't think this is going to go anywhere. -Pierre

5 10

[alsa-devel] [PATCH 0/3] ASoC: add support for jack detection in mid-x86 and sn95031 codec
by Harsha, Priya 03 Feb '11

03 Feb '11

Hi Mark, This patch set adds support for jack detection in mid-x6 machine and TI sn95031 codec drivers. Also adds a helper function in soc-jack.c for converting mic bias values to jack types. The comments from previous submission are addressed. [PATCH 1/3] ASoC: soc_jack - add function to add voltage zones and determine jack type This patch adds a helper functions that adds the adc voltage zones and determines the jack type given the mic bias value. When a jack is created, the voltage zones can to be added to it. Later when a jack is inserted, the micbias voltage can be measured and used to find the jack type [PATCH 2/3] ASoC: mid-x86 machine driver - add jack detect support This patch adds the jack detection in mid-86 machine driver [PATCH 3/3] ASoC: sn95031 codec - adding jack detection/reporting This patch adds the jack detection logic in sn95031 codec. This also adds code for reading the ADC to figure out the mic bias and therby report jack type detected -Harsha

2 1

[alsa-devel] [PATCH v6 0/3] input: Add support for ECI (multimedia) accessories
by tapio.vihuri＠nokia.com 03 Feb '11

03 Feb '11

From: Tapio Vihuri <tapio.vihuri(a)nokia.com> Thank you for comments. ECI input driver changes: - Add debuging config option with adequate "help" in Kconfig - Clean out debug messages - Clean out error messages - Force all buttons up event when unplugging accessory ECI controller driver changes: - Remove unused debuging config option Created against linux-2.6.38-rc2 at: 70d1f36 Merge branch 'for-linus' of git://git.infradead.org/ubi-2.6 Please review. --- v5 message --- Thank you for comments. ECI input driver changes: - Use sparse_keymap_setup() properly, remove unneeded input set_bits ECI controller driver changes: - Added "button" debugfs entry for input driver buttons handling testing --- v4 message --- Thank you for comments. ECI input driver changes: - Use driver name in debugfs - reject accessory stating more than maximum 31 buttons - clean out couple comments ECI controller driver changes: - name change, from ecibus to eci_xxx_ctrl - location change, from drivers/ecibus to drivers/input/misc - major rework from platform driver to i2c-driver - removed mrst platform dependency Checkpatch warn about Kconfig, but I think it is false positive, and I will report it to the maintainer. Platform code changes: - medfield_ prefix changed to mrst_ - added ECI controller needed power control - remover ECI controller platform driver stuff - added i2c stuff for ECI controller driver Created against linux-2.6.37-rc8 at: a25cecc Merge branch 'next' of git://git.monstr.eu/linux-2.6-microblaze --- v3 message --- Thank you for comments. ECI input driver changes: - Makefile and Kconfig sorted alphabetically - using sparse keymap library, much cleaner solution - correct string aligment with 80 column - debugfs initialization fix - sysfs cable entry using just 0/1 - using cheaper __set_bit - insane comments fixed - remove unneeded misc device Created against linux-2.6.37-rc8 at: 989d873 Merge master.kernel.org:/home/rmk/linux-2.6-arm --- v2 message --- Thank you for comments. In series two there Kconfig dependency problem fixed, preventing compilation if X86_MRST (Intel Moorestown) or INPUT_MISC is not selected. Also created against linux-2.6.37-rc8 --- v1 message --- Hi all This patch set introduce Multimedia Headset Accessory support for Nokia phones. Technically those are known as ECI (Enhancement Control Interface) If headset has many buttons, like play, vol+, vol- etc. then it is propably ECI accessory. Among several buttons ECI accessories contains memory for storing several parameters. This ECI input driver provides the following features: - reading ECI configuration memory - ECI buttons as input events Drive is constructed as follows: - ECI accessory input driver deals with headset accessory - ECI bus control driver deals the HW transfering data to/from headset - platform data match used HW In the future accessory detection logic will be added using ALSA jack reporting. Created against linux-2.6.37-rc6 Please review. Tapio Vihuri (3): ECI: input: introduce ECI accessory input driver ECI: introducing ECI controller driver ECI: adding platform data for ECI driver arch/x86/platform/mrst/mrst.c | 48 ++ drivers/input/misc/Kconfig | 65 +++ drivers/input/misc/Makefile | 3 +- drivers/input/misc/eci.c | 949 ++++++++++++++++++++++++++++++++++++ drivers/input/misc/eci_at20_ctrl.c | 566 +++++++++++++++++++++ include/linux/input/eci.h | 179 +++++++ 6 files changed, 1809 insertions(+), 1 deletions(-) create mode 100644 drivers/input/misc/eci.c create mode 100644 drivers/input/misc/eci_at20_ctrl.c create mode 100644 include/linux/input/eci.h

3 7

[alsa-devel] [PATCH] ASoC: fix build with PM_SLEEP unset
by Guennadi Liakhovetski 03 Feb '11

03 Feb '11

Fix a wrong preprocessor conditional, that leads to soc-core.c compilation failure with CONFIG_PM set and CONFIG_PM_SLEEP unset. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 205cbd7..10bc3c2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1801,7 +1801,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } card->snd_card->dev = card->dev; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* deferred resume work */ INIT_WORK(&card->deferred_resume_work, soc_resume_deferred); #endif

2 1

[alsa-devel] [PATCH 2.6.38] ASoC: fill in snd_soc_pcm_runtime.card before calling snd_soc_dai_link.init()
by Janusz Krzysztofik 03 Feb '11

03 Feb '11

The .card member of the snd_soc_pcm_runtime structure pointed to by the snd_soc_dai_link.init() argument used to be initialized before the function being called. This has changed, probably unintentionally, after recent refactorings. Since the function implementations are free to make use of this pointer, move its assignment back before the function is called to avoid NULL pointer dereferences. Created and tested on Amstrad Delta againts linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Jarkko, I'm not sure if this change won't break anything in your dailess related code path, please verify. Thanks, Janusz sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- git/sound/soc/soc-core.c.orig 2011-01-31 19:49:29.000000000 +0100 +++ git/sound/soc/soc-core.c 2011-02-02 20:23:10.000000000 +0100 @@ -1449,6 +1449,7 @@ static int soc_post_component_init(struc rtd = &card->rtd_aux[num]; name = aux_dev->name; } + rtd->card = card; /* machine controls, routes and widgets are not prefixed */ temp = codec->name_prefix; @@ -1471,7 +1472,6 @@ static int soc_post_component_init(struc /* register the rtd device */ rtd->codec = codec; - rtd->card = card; rtd->dev.parent = card->dev; rtd->dev.release = rtd_release; rtd->dev.init_name = name;

3 2

[alsa-devel] [PATCH 09/19] wl1273: mfd_cell is now implicitly available to drivers
by Andres Salomon 03 Feb '11

03 Feb '11

No need to explicitly set the cell's platform_data/data_size. In this case, move the various platform_data pointers to driver_data. All of the clients which make use of it are also changed. Signed-off-by: Andres Salomon <dilinger(a)queued.net> --- drivers/media/radio/radio-wl1273.c | 2 +- drivers/mfd/wl1273-core.c | 6 ++---- sound/soc/codecs/wl1273.c | 2 +- 3 files changed, 4 insertions(+), 6 deletions(-) diff --git a/drivers/media/radio/radio-wl1273.c b/drivers/media/radio/radio-wl1273.c index 7ecc8e6..ebb6eb5 100644 --- a/drivers/media/radio/radio-wl1273.c +++ b/drivers/media/radio/radio-wl1273.c @@ -2138,7 +2138,7 @@ static int wl1273_fm_radio_remove(struct platform_device *pdev) static int __devinit wl1273_fm_radio_probe(struct platform_device *pdev) { - struct wl1273_core **core = pdev->dev.platform_data; + struct wl1273_core **core = platform_get_drvdata(pdev); struct wl1273_device *radio; struct v4l2_ctrl *ctrl; int r = 0; diff --git a/drivers/mfd/wl1273-core.c b/drivers/mfd/wl1273-core.c index d2ecc24..61ec252 100644 --- a/drivers/mfd/wl1273-core.c +++ b/drivers/mfd/wl1273-core.c @@ -79,8 +79,7 @@ static int __devinit wl1273_core_probe(struct i2c_client *client, cell = &core->cells[children]; cell->name = "wl1273_fm_radio"; - cell->platform_data = &core; - cell->data_size = sizeof(core); + cell->driver_data = &core; children++; if (pdata->children & WL1273_CODEC_CHILD) { @@ -88,8 +87,7 @@ static int __devinit wl1273_core_probe(struct i2c_client *client, dev_dbg(&client->dev, "%s: Have codec.\n", __func__); cell->name = "wl1273-codec"; - cell->platform_data = &core; - cell->data_size = sizeof(core); + cell->driver_data = &core; children++; } diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 861b28f..0af2c2d 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -436,7 +436,7 @@ EXPORT_SYMBOL_GPL(wl1273_get_format); static int wl1273_probe(struct snd_soc_codec *codec) { - struct wl1273_core **core = codec->dev->platform_data; + struct wl1273_core **core = dev_get_drvdata(codec->dev); struct wl1273_priv *wl1273; int r; -- 1.7.2.3

1 0

[alsa-devel] WL1273 FM Radio driver...
by Matti J. Aaltonen 03 Feb '11

03 Feb '11

Hello I have been trying to get the WL1273 FM radio driver into the kernel for some time. It has been kind of difficult, one of the reasons is that I didn't realize I should have tried to involve all relevant maintainers to the discussion form the beginning (AsoC, Media and MFD). At Mark's suggestion I'm trying to reopen the discussion now. The driver consists of an MFD core and two child drivers (the audio codec and the V4L2 driver). And the question is mainly about the role of the MFD driver: the original design had the IO functions in the core. Currently the core is practically empty mainly because Mauro very strongly wanted to have “everything” in the V4L2 driver. I liked the original design because it didn't have the bug that the current MFD has: the codec can be initialized before the V4L2 part sets the IO function pointers. Having in principle equally capable interface drivers is symmetrical and esthetically pleasing:-) Also somebody could easily leave out the existing interfaces and create a completely new one based for example to sysfs or something... Having the IO in the core would also conveniently hide the physical communication layer and make it easy to change I2C to UART, which is also supported by the chip. Thanks, Matti

4 5

[alsa-devel] [PATCH] ASoC: soc-core: Support debugfs entries larger than PAGE_SIZE bytes
by Dimitris Papastamos 02 Feb '11

02 Feb '11

For some codecs with large register maps, it was not possible to dump all registers via the codec_reg file but only up to PAGE_SIZE bytes. This patch fixes this problem. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 110 +++++++++++++++++++++++++++++++++----------------- 1 files changed, 73 insertions(+), 37 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 205cbd7..83626f9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -87,15 +87,56 @@ static int min_bytes_needed(unsigned long val) return c; } +/* fill buf which is 'len' bytes with a formatted + * string of the form 'reg: value\n' */ +static int format_register_str(struct snd_soc_codec *codec, + unsigned int reg, char *buf, size_t len) +{ + int wordsize = codec->driver->reg_word_size * 2; + int regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + int ret; + char tmpbuf[len + 1]; + char regbuf[regsize + 1]; + + /* since tmpbuf is allocated on the stack, warn the callers if they + * try to abuse this function */ + WARN_ON(len > 63); + + /* +2 for ': ' and + 1 for '\n' */ + if (wordsize + regsize + 2 + 1 != len) + return -EINVAL; + + ret = snd_soc_read(codec , reg); + if (ret < 0) { + memset(regbuf, 'X', regsize); + regbuf[regsize] = '\0'; + } else { + snprintf(regbuf, regsize + 1, "%.*x", regsize, ret); + } + + /* prepare the buffer */ + snprintf(tmpbuf, len + 1, "%.*x: %s\n", wordsize, reg, regbuf); + /* copy it back to the caller without the '\0' */ + memcpy(buf, tmpbuf, len); + + return 0; +} + /* codec register dump */ -static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) +static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf, + size_t count, loff_t pos) { - int ret, i, step = 1, count = 0; + int i, step = 1; int wordsize, regsize; + int len; + size_t total = 0; + loff_t p = 0; wordsize = codec->driver->reg_word_size * 2; regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + len = wordsize + regsize + 2 + 1; + if (!codec->driver->reg_cache_size) return 0; @@ -105,51 +146,34 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) for (i = 0; i < codec->driver->reg_cache_size; i += step) { if (codec->readable_register && !codec->readable_register(codec, i)) continue; - - count += sprintf(buf + count, "%.*x: ", regsize, i); - if (count >= PAGE_SIZE - 1) - break; - if (codec->driver->display_register) { count += codec->driver->display_register(codec, buf + count, PAGE_SIZE - count, i); } else { - /* If the read fails it's almost certainly due to - * the register being volatile and the device being - * powered off. - */ - ret = snd_soc_read(codec, i); - if (ret >= 0) - count += snprintf(buf + count, - PAGE_SIZE - count, - "%.*x", wordsize, ret); - else - count += snprintf(buf + count, - PAGE_SIZE - count, - "<no data: %d>", ret); + /* only support larger than PAGE_SIZE bytes debugfs + * entries for the default case */ + if (p >= pos) { + if (total + len >= count - 1) + break; + format_register_str(codec, i, buf + total, len); + total += len; + } + p += len; } - - if (count >= PAGE_SIZE - 1) - break; - - count += snprintf(buf + count, PAGE_SIZE - count, "\n"); - if (count >= PAGE_SIZE - 1) - break; } - /* Truncate count; min() would cause a warning */ - if (count >= PAGE_SIZE) - count = PAGE_SIZE - 1; + total = min(total, count - 1); - return count; + return total; } + static ssize_t codec_reg_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_soc_pcm_runtime *rtd = container_of(dev, struct snd_soc_pcm_runtime, dev); - return soc_codec_reg_show(rtd->codec, buf); + return soc_codec_reg_show(rtd->codec, buf, PAGE_SIZE, 0); } static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); @@ -188,16 +212,28 @@ static int codec_reg_open_file(struct inode *inode, struct file *file) } static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, - size_t count, loff_t *ppos) + size_t count, loff_t *ppos) { ssize_t ret; struct snd_soc_codec *codec = file->private_data; - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + char *buf; + + if (*ppos < 0 || !count) + return -EINVAL; + + buf = kmalloc(count, GFP_KERNEL); if (!buf) return -ENOMEM; - ret = soc_codec_reg_show(codec, buf); - if (ret >= 0) - ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + + ret = soc_codec_reg_show(codec, buf, count, *ppos); + if (ret >= 0) { + if (copy_to_user(user_buf, buf, ret)) { + kfree(buf); + return -EFAULT; + } + *ppos += ret; + } + kfree(buf); return ret; } -- 1.7.3.5

3 2

[alsa-devel] [PATCH] ASoC: PXA: formatting
by Sven Neumann 02 Feb '11

02 Feb '11

Signed-off-by: Sven Neumann <s.neumann(a)raumfeld.com> --- sound/soc/pxa/raumfeld.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 0fd60f4..4a4b73e 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -229,19 +229,19 @@ static struct snd_soc_dai_link raumfeld_dai[] = { { .name = "ak4104", .stream_name = "Playback", - .cpu_dai_name = "pxa-ssp-dai.1", - .codec_dai_name = "ak4104-hifi", - .platform_name = "pxa-pcm-audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "ak4104-hifi", + .platform_name = "pxa-pcm-audio", .ops = &raumfeld_ak4104_ops, - .codec_name = "ak4104-codec.0", + .codec_name = "ak4104-codec.0", }, { .name = "CS4270", .stream_name = "CS4270", - .cpu_dai_name = "pxa-ssp-dai.0", - .platform_name = "pxa-pcm-audio", - .codec_dai_name = "cs4270-hifi", - .codec_name = "cs4270-codec.0-0048", + .cpu_dai_name = "pxa-ssp-dai.0", + .platform_name = "pxa-pcm-audio", + .codec_dai_name = "cs4270-hifi", + .codec_name = "cs4270-codec.0-0048", .ops = &raumfeld_cs4270_ops, },}; -- 1.7.1

3 3

[alsa-devel] [PATCH] EDB93xx: Add support for CS4271 CODEC on EDB93xx boards
by Alexander Sverdlin 02 Feb '11

02 Feb '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Add support for CS4271 SPI-connected CODEC on EDB93xx boards. Major rework based on code provided by H Hartley Sweeten <hsweeten(a)visionengravers.com>. Tested on EDB9302, others (EDB9301, EDB9302A, EDB9307A, EDB0315A) should work. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- arch/arm/mach-ep93xx/edb93xx.c | 117 ++++++++++++++++++++++++++++++++++++++++ 1 files changed, 117 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-ep93xx/edb93xx.c b/arch/arm/mach-ep93xx/edb93xx.c index 4b04316..f20be96 100644 --- a/arch/arm/mach-ep93xx/edb93xx.c +++ b/arch/arm/mach-ep93xx/edb93xx.c @@ -30,12 +30,22 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/i2c-gpio.h> +#include <linux/spi/spi.h> +#include <linux/delay.h> +#include <mach/ep93xx_spi.h> #include <mach/hardware.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> +#include <sound/cs4271.h> + +#define edb93xx_has_audio() (machine_is_edb9301() || \ + machine_is_edb9302() || \ + machine_is_edb9302a() || \ + machine_is_edb9307a() || \ + machine_is_edb9315a()) static void __init edb93xx_register_flash(void) { @@ -93,6 +103,111 @@ static void __init edb93xx_register_i2c(void) /************************************************************************* + * EDB93xx SPI peripheral handling + *************************************************************************/ +static int edb93xx_cs4271_hw_setup(struct spi_device *spi) +{ + int gpio_nreset; + int err; + + if (machine_is_edb9301() || machine_is_edb9302()) { + gpio_nreset = EP93XX_GPIO_LINE_EGPIO1; + } else if (machine_is_edb9302a() || machine_is_edb9307a()) { + ep93xx_devcfg_set_bits(EP93XX_SYSCON_DEVCFG_HONIDE); + gpio_nreset = EP93XX_GPIO_LINE_DD2; + } else if (machine_is_edb9315a()) { + gpio_nreset = EP93XX_GPIO_LINE_EGPIO14; + } else { + return -EINVAL; + } + + err = gpio_request(gpio_nreset, spi->modalias); + if (err) + return err; + err = gpio_request(EP93XX_GPIO_LINE_EGPIO6, spi->modalias); + if (err) + return err; + + gpio_direction_output(EP93XX_GPIO_LINE_EGPIO6, 1); + + /* Reset codec */ + gpio_direction_output(gpio_nreset, 0); + udelay(1); + gpio_set_value(gpio_nreset, 1); + /* Give the codec time to wake up */ + udelay(1); + + return 0; +} + +static void edb93xx_cs4271_hw_cleanup(struct spi_device *spi) +{ + int gpio_nreset; + + if (machine_is_edb9301() || machine_is_edb9302()) + gpio_nreset = EP93XX_GPIO_LINE_EGPIO1; + else if (machine_is_edb9302a() || machine_is_edb9307a()) + gpio_nreset = EP93XX_GPIO_LINE_DD2; + else if (machine_is_edb9315a()) + gpio_nreset = EP93XX_GPIO_LINE_EGPIO14; + else + return; + + gpio_set_value(gpio_nreset, 0); + gpio_free(gpio_nreset); + + gpio_set_value(EP93XX_GPIO_LINE_EGPIO6, 1); + gpio_free(EP93XX_GPIO_LINE_EGPIO6); +} + +static void edb93xx_cs4271_hw_cs_control(struct spi_device *spi, int value) +{ + gpio_set_value(EP93XX_GPIO_LINE_EGPIO6, value); +} + +static struct ep93xx_spi_chip_ops edb93xx_cs4271_hw = { + .setup = edb93xx_cs4271_hw_setup, + .cleanup = edb93xx_cs4271_hw_cleanup, + .cs_control = edb93xx_cs4271_hw_cs_control, +}; + +static struct spi_board_info edb93xx_spi_board_info[] __initdata = { + { + .modalias = "cs4271", + .controller_data = &edb93xx_cs4271_hw, + .max_speed_hz = 6000000, + .bus_num = 0, + .chip_select = 0, + .mode = SPI_MODE_3, + }, +}; + +static struct ep93xx_spi_info edb93xx_spi_info = { + .num_chipselect = ARRAY_SIZE(edb93xx_spi_board_info), +}; + +static void __init edb93xx_register_spi(void) +{ + if (edb93xx_has_audio()) { + ep93xx_register_spi(&edb93xx_spi_info, + edb93xx_spi_board_info, + ARRAY_SIZE(edb93xx_spi_board_info)); + } +} + + +/************************************************************************* + * EDB93xx I2S + *************************************************************************/ +static void __init edb93xx_register_i2s(void) +{ + if (edb93xx_has_audio()) { + ep93xx_register_i2s(); + } +} + + +/************************************************************************* * EDB93xx pwm *************************************************************************/ static void __init edb93xx_register_pwm(void) @@ -117,6 +232,8 @@ static void __init edb93xx_init_machine(void) edb93xx_register_flash(); ep93xx_register_eth(&edb93xx_eth_data, 1); edb93xx_register_i2c(); + edb93xx_register_spi(); + edb93xx_register_i2s(); edb93xx_register_pwm(); }

4 17

[alsa-devel] snd-usb-audio dying on jackd xruns
by Rods Bobavich 02 Feb '11

02 Feb '11

So I start jackd and it runs fine for a while. Then ALSA crashes and triggers an xrun. Jackd continues to do its merry thing - minus the part about spitting info out to my soundcard... At first I thought this was a Jackd issue but when I looked deeper it seems to be ALSA. Once this happens I cannot restart jackd and resume an audio session. In fact, I cannot even rmmod snd-usb-audio. Sometimes I cannot lsusb. So it looks like something is getting seriously hosed! It seems to be specific to my machine. I've tried multiple usb intefaces. All work fine with other computers. I've tried multiple version of Ubuntu with various versions of ALSA on this machine. All exhibit the same behavior. I have 3 other machines that run these interfaces just fine... So it seems to be a hardware compatibility issue not a software glitch in one version. I've tried many boot flags but to no avail. I'm at a loss on how to troubleshoot. The machine is M4N75TD motherboard with a Phenom II 1055T and the latest bios flash. Please advise, ...Rods ;-)

2 2

[alsa-devel] Expired security certificate for https://bugtrack.alsa-project.org
by George Morgan 02 Feb '11

02 Feb '11

Any chance that the security certificate can be updated for https://bugtrack.alsa-project.org ? The certificate expired on 7/19/2009 6:18 AM. George Morgan

4 3

[alsa-devel] question about ctl.pulse ?
by Raymond Yau 02 Feb '11

02 Feb '11

Refer to http://git.alsa-project.org/?p=alsa-plugins.git;a=blob;f=doc/README-pulse;h… The mixer control plugin also accepts the parameters "source" and "sink" for when you need to specify a sink/source combination with different names. If you need to do this with PCM:s then specify two PCM:s with different "device". Does it mean that the application only get playback controls from sink0, sink1 and capture controls from source0, source1, source2 ctl.sink0 { type pulse sink "0" } ctl.source0 { type pulse source "0" } ctl.sink1 { type pulse sink "1" } ctl.source1 { type pulse source "1" } ctl.sink2 { type pulse sink "2" } ctl.source2 { type pulse source "2" }

2 1

[alsa-devel] [PATCH] asoc: davinci: da830/omap-l138: correct cpu_dai_name
by Vaibhav Bedia 02 Feb '11

02 Feb '11

McASP1 is used on the DA830/OMAP-L137 platform for the codec. This is different from the DA850/OMAP-L138 platform which uses McASP0. This is fixed by adding a new snd_soc_dai_link struct. Signed-off-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> --- sound/soc/davinci/davinci-evm.c | 18 +++++++++++++++--- 1 files changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index b36f0b3..fe79842 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -218,7 +218,19 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .ops = &evm_spdif_ops, }, }; -static struct snd_soc_dai_link da8xx_evm_dai = { + +static struct snd_soc_dai_link da830_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name = "davinci-mcasp.1", + .codec_dai_name = "tlv320aic3x-hifi", + .codec_name = "tlv320aic3x-codec.1-0018", + .platform_name = "davinci-pcm-audio", + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +static struct snd_soc_dai_link da850_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", @@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { static struct snd_soc_card da830_snd_soc_card = { .name = "DA830/OMAP-L137 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da830_evm_dai, .num_links = 1, }; static struct snd_soc_card da850_snd_soc_card = { .name = "DA850/OMAP-L138 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da850_evm_dai, .num_links = 1, }; -- 1.6.2.4

3 2

[alsa-devel] alsa-lib TLVs and valgrind
by Sebastian H. 02 Feb '11

02 Feb '11

Hello list I'm about to release a new version of alsamixer-qt4 (now named qasmixer) and ran the program through valgrind. valgrind --track-origins=yes src/qasmixer &> valgrind.log The log file contains tons of warnings like this ==3432== Conditional jump or move depends on uninitialised value(s) ... ... (qasmixer function calls) ... ==3432== Uninitialised value was created by a heap allocation ... malloc (vg_replace_malloc.c:236) ... init_db_range (simple_none.c:1111) ... get_dB_range (simple_none.c:1156) ... get_dB_range_ops (simple_none.c:1173) ... snd_mixer_selem_get_playback_dB_range (simple.c:298) ... ... (qasmixer function calls) ... The fresh alsa-lib-10-24.1 was used for this, compiled with debug symbols enabled. Drivers are from Debian/squeeze. This issue doesn't seem critical for the application but it totally floods the valgrind.log file. Best, Sebastian Holtermann

2 4

Re: [alsa-devel] need input on USB Audio class device.
by Clemens Ladisch 02 Feb '11

02 Feb '11

Sakthi Subramanian wrote: > i have opened a capture device, and the captured data i have written to a > file. > That file is of RAW format. > I have opened the raw file in audacity, with 8 channel it shows only 2 > channels having the captured data. Raw-format files do not have any information about the format, so Audacity has to guess. Just write a .wav file with the proper headers. Regards, Clemens

1 0

[alsa-devel] [PATCH] ASoC: soc-core: Remove display_register() callback
by Dimitris Papastamos 02 Feb '11

02 Feb '11

Remove the display_register() callback as there are no users of it in the upstream kernel and makes it difficult to unify the format of the codec_reg file. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 -- sound/soc/soc-core.c | 31 +++++++++++++------------------ 2 files changed, 13 insertions(+), 20 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4b6c0a8..2e6463f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -536,8 +536,6 @@ struct snd_soc_codec_driver { /* codec IO */ unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); - int (*display_register)(struct snd_soc_codec *, char *, - size_t, unsigned int); int (*volatile_register)(struct snd_soc_codec *, unsigned int); int (*readable_register)(struct snd_soc_codec *, unsigned int); short reg_cache_size; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 205cbd7..4c061c3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -110,24 +110,19 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) if (count >= PAGE_SIZE - 1) break; - if (codec->driver->display_register) { - count += codec->driver->display_register(codec, buf + count, - PAGE_SIZE - count, i); - } else { - /* If the read fails it's almost certainly due to - * the register being volatile and the device being - * powered off. - */ - ret = snd_soc_read(codec, i); - if (ret >= 0) - count += snprintf(buf + count, - PAGE_SIZE - count, - "%.*x", wordsize, ret); - else - count += snprintf(buf + count, - PAGE_SIZE - count, - "<no data: %d>", ret); - } + /* If the read fails it's almost certainly due to + * the register being volatile and the device being + * powered off. + */ + ret = snd_soc_read(codec, i); + if (ret >= 0) + count += snprintf(buf + count, + PAGE_SIZE - count, + "%.*x", wordsize, ret); + else + count += snprintf(buf + count, + PAGE_SIZE - count, + "<no data: %d>", ret); if (count >= PAGE_SIZE - 1) break; -- 1.7.3.5

3 4

[alsa-devel] [PATCH] ASoC: soc-core: Support debugfs entries larger than PAGE_SIZE bytes
by Dimitris Papastamos 02 Feb '11

02 Feb '11

For some codecs with large register maps, it was not possible to dump all registers via the codec_reg file but only up to PAGE_SIZE bytes. This patch fixes this problem. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 114 +++++++++++++++++++++++++++++++++---------------- 1 files changed, 77 insertions(+), 37 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 205cbd7..a7710d2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -87,15 +87,55 @@ static int min_bytes_needed(unsigned long val) return c; } +/* fill buf which is 'len' bytes with a formatted + * string of the form 'reg: value\n' */ +static int format_register_str(struct snd_soc_codec *codec, + unsigned int regidx, char *buf, size_t len) +{ + int wordsize = codec->driver->reg_word_size * 2; + int regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + int ret; + char tmpbuf[len + 1]; + + /* since tmpbuf is allocated on the stack, warn the callers if they + * try to abuse this function */ + WARN_ON(len > 63); + + /* +2 for ': ' and + 1 for '\n' */ + if (wordsize + regsize + 2 + 1 != len) + return -EINVAL; + + /* we don't currently handle failed reads */ + ret = snd_soc_read(codec , regidx); + if (ret < 0) { + dev_err(codec->dev, "Failed to read from %#x\n", regidx); + return ret; + } + + /* prepare the buffer */ + snprintf(tmpbuf, len + 1, "%.*x: %.*x\n", wordsize, + regidx, regsize, ret); + /* copy it back to the caller without the '\0' */ + memcpy(buf, tmpbuf, len); + + return 0; +} + /* codec register dump */ -static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) +static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf, + size_t count, loff_t pos) { - int ret, i, step = 1, count = 0; + int i, step = 1; int wordsize, regsize; + int len; + size_t total = 0; + loff_t p = 0; wordsize = codec->driver->reg_word_size * 2; regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + len = wordsize + regsize + 2 + 1; + if (!codec->driver->reg_cache_size) return 0; @@ -105,51 +145,34 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) for (i = 0; i < codec->driver->reg_cache_size; i += step) { if (codec->readable_register && !codec->readable_register(codec, i)) continue; - - count += sprintf(buf + count, "%.*x: ", regsize, i); - if (count >= PAGE_SIZE - 1) - break; - if (codec->driver->display_register) { count += codec->driver->display_register(codec, buf + count, PAGE_SIZE - count, i); } else { - /* If the read fails it's almost certainly due to - * the register being volatile and the device being - * powered off. - */ - ret = snd_soc_read(codec, i); - if (ret >= 0) - count += snprintf(buf + count, - PAGE_SIZE - count, - "%.*x", wordsize, ret); - else - count += snprintf(buf + count, - PAGE_SIZE - count, - "<no data: %d>", ret); + /* only support larger than PAGE_SIZE bytes debugfs + * entries for the default case */ + if (p >= pos) { + if (total + len >= count - 1) + break; + format_register_str(codec, i, buf + total, len); + total += len; + } + p += len; } - - if (count >= PAGE_SIZE - 1) - break; - - count += snprintf(buf + count, PAGE_SIZE - count, "\n"); - if (count >= PAGE_SIZE - 1) - break; } - /* Truncate count; min() would cause a warning */ - if (count >= PAGE_SIZE) - count = PAGE_SIZE - 1; + total = min(total, count - 1); - return count; + return total; } + static ssize_t codec_reg_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_soc_pcm_runtime *rtd = container_of(dev, struct snd_soc_pcm_runtime, dev); - return soc_codec_reg_show(rtd->codec, buf); + return soc_codec_reg_show(rtd->codec, buf, PAGE_SIZE, 0); } static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); @@ -188,16 +211,33 @@ static int codec_reg_open_file(struct inode *inode, struct file *file) } static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, - size_t count, loff_t *ppos) + size_t count, loff_t *ppos) { ssize_t ret; struct snd_soc_codec *codec = file->private_data; - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + char *buf; + + if (*ppos < 0 || !count) + return -EINVAL; + + buf = kmalloc(count, GFP_KERNEL); if (!buf) return -ENOMEM; - ret = soc_codec_reg_show(codec, buf); - if (ret >= 0) - ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + + ret = soc_codec_reg_show(codec, buf, count, *ppos); + if (ret >= 0) { + if (!access_ok(VERIFY_WRITE, user_buf, ret)) { + ret = -EFAULT; + goto out; + } + if (copy_to_user(user_buf, buf, ret)) { + ret = -EFAULT; + goto out; + } + *ppos += ret; + } + +out: kfree(buf); return ret; } -- 1.7.3.5

2 3

[alsa-devel] [PATCH] ASoC: Amstrad Delta: fix NULL pointer dereference
by Janusz Krzysztofik 02 Feb '11

02 Feb '11

It looks like the .card member of the snd_soc_pcm_runtime structure pointed to by the snd_soc_dai_link.init() argument is no longer initialized before the function being called. As the Amstrad Delta sound card driver was making use of it for locating its snd_soc_card structure, this resulted in actual or potential NULL pointer dereferences. Since the structure in question is local to the driver, its address can be referred directly instead of trying to derive it from the function argument. Fix it. Created and tested against linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Please consider queuing this patch for the current rc cycle. sound/soc/omap/ams-delta.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) --- git/sound/soc/omap/ams-delta.c.orig 2011-01-31 19:49:29.000000000 +0100 +++ git/sound/soc/omap/ams-delta.c 2011-02-01 17:54:22.000000000 +0100 @@ -492,12 +492,14 @@ static void ams_delta_shutdown(struct sn * Card initialization */ +static struct snd_soc_card ams_delta_audio_card; + static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_card *card = rtd->card; + struct snd_soc_card *card = &ams_delta_audio_card; int ret; /* Codec is ready, now add/activate board specific controls */

2 2

Re: [alsa-devel] problem with mms stream - [AO_ALSA] Trying to reset soundcard
by Raymond Yau 02 Feb '11

02 Feb '11

>Приветствую, Reimar > On Sun, Jan 24, 2010 at 02:36:50PM +0300, Goga777 wrote: > > [AO_ALSA] Write error: Broken pipe4428928.0 (-24.-8) 96.4% 6% > > [AO_ALSA] Trying to reset soundcard. > > Increasing filtered audio buffer size from 65536 to 675361% 9% > > Increasing filtered audio buffer size from 67536 to 675442% 12% > > [AO_ALSA] Write error: Broken pipe4428928.0 (-24.-8) 24.0% 6% > > [AO_ALSA] Trying to reset soundcard. > > I am not sure this is the same issue, but I had this part as well. res = snd_pcm_writei(alsa_handler, data, num_frames); ... if (res < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_WriteError, snd_strerror(res)); mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_TryingToResetSoundcard); if ((res = snd_pcm_prepare(alsa_handler)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(res)); return 0; break; } } The "pulse" plugin return -EPIPE "Broken pipe" at four conditions after mplayer call snd_pcm_writei() 1) io->state == SND_PCM_STATE_XRUN 2) pcm->underrun 3) update_ptr(pcm) return err mplayer call snd_pcm_prepare() to recover from "Broken Pipe". but why mplayer increasing filtered audio buffer ? static snd_pcm_sframes_t pulse_pointer(snd_pcm_ioplug_t * io) { ... if (io->state == SND_PCM_STATE_XRUN) return -EPIPE; if (pcm->underrun) { ret = -EPIPE; goto finish; } if (io->state != SND_PCM_STATE_RUNNING) return 0; pa_threaded_mainloop_lock(pcm->p->mainloop); ret = check_stream(pcm); if (ret < 0) goto finish; if (pcm->underrun) { ret = -EPIPE; goto finish; } ret = update_ptr(pcm); if (ret < 0) { ret = -EPIPE; goto finish; } if (pcm->underrun) ret = -EPIPE; else ret = snd_pcm_bytes_to_frames(io->pcm, pcm->ptr); finish:

1 0

Re: [alsa-devel] [PATCH] ASoC: EDB93xx machine sound driver with CS4271
by Alexander Sverdlin 02 Feb '11

02 Feb '11

Dear Todor, > Zdrastvuj Aleks, > I'd like to test the sound module. Which is the latest patch versions > and > coresponding kernel version? In which official kernel release do you > plan to > merge it? As I can conclude from list, you should see it in 2.6.39 and possibly -next already. 75% of the driver has been merged already. All you have to do to start testing is to add patch that I've just sent to the list ([PATCH] EDB93xx: Add support for CS4271 CODEC on EDB93xx boards). Best regards, Alexander.

1 0

[alsa-devel] snd_soc_jack_add_gpios vs. SOC_DAPM_PIN_SWITCH
by Stephen Warren 02 Feb '11

02 Feb '11

I've set up headphone hack detection in my machine driver, using snd_soc_jack_new, snd_soc_jack_add_pins, and snd_soc_jack_add_gpios. I've also created a control to allow the user to enable/disable the headphone jack: static const struct snd_kcontrol_new harmony_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), }; On their own, both of these work fine. However, if I use the control to disable the headphones, then unplug/replug the headphones, the jack detect seems to override the control, and I hear audio from the headphones. Is this something that's known not yet implemented or was never intended to be used this way, or should I look further to find the issue? FYI, sound/soc/samsung/smartq_wm8987.c has this same code structure, and I imagine will have the same issue. Thanks. -- nvpublic

2 3

[alsa-devel] ASoC updates for 2.6.39.
by Mark Brown 01 Feb '11

01 Feb '11

The following changes since commit a211701eb1e4a41bbee22d61f35dba8e55925db4: ASoC: sst_platform porting sst dsp driver interface as per latest in Greg's staging tree (2011-01-19 11:28:11 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Alexander Sverdlin (3): ASoC: CS4271 codec support ASoC: EDB93xx machine sound driver with CS4271 ASoC: cs4271.c: improve error handling Dimitris Papastamos (5): ASoC: WM8995: Fix incorrect use of snd_soc_update_bits() ASoC: soc-cache: Introduce the cache_bypass option ASoC: soc-cache: Apply the cache_bypass option ASoC: soc-cache: Add trace event for snd_soc_cache_sync() ASoC: soc-core: Ensure codec_reg has fixed length fields Dmitry Eremin-Solenikov (1): ASoC: correct link specifications for corgi, poodle and spitz H Hartley Sweeten (1): ASoC: neo1973_wm8753 audio support does not require scoop Harsha Priya (3): ASoC: sn95031: add capture support ASoC: sst_platform: add support for capture stream on headset dai ASoC: mid-x86: Add support for capture in machine driver Janusz Krzysztofik (2): ASoC: Amstrad Delta: fix const related build error ASoC: CX20442: fix NULL pointer dereference Jarkko Nikula (4): ASoC: Fix module refcount for auxiliary devices ASoC: omap: rx51: Add stereo output support to audio jack ASoC: soc-core: Increment codec and platform driver refcounts before probing ASoC: omap: rx51: Add earphone support Kuninori Morimoto (7): ASoC: ak4642: add SND_SOC_DAIFMT_FORMAT support ASoC: sh: fsi: Add fsi_get_priv_frm_dai function ASoC: sh: fsi: Add snd_soc_dai_set_fmt support ASoC: sh: fsi-hdmi: Add FSI port and HDMI selection ASoC: sh: fsi: move chan_num from fsi_stream to fsi_priv ASoC: sh: fsi: free from NULL pointer of struct sh_fsi_platform_info ASoC: sh: fsi: modify selection method of I2S/PCM/SPDIF format Lars-Peter Clausen (2): ASoC: Fix codec device id format used by some dai_links ASoC: Samsung: Fix outdated cpu_dai_name for s3c24xx i2s Manjunathappa, Prakash (1): ASoC: DaVinci: fix kernel panic due to uninitialized platform_data Mark Brown (25): ASoC: Explicitly say if we're powering up or down ASoC: Add support for sequencing within ASoC: Provide per widget type callback when executing DAPM sequences Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Staticise twl6040_hs_jack_report() ASoC: Handle low measured DC offsets for wm_hubs devices Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Fix type for snd_soc_volatile_register() ASoC: Remove controls from sequenced PGA arguments Merge branch 'for-2.6.38' into for-2.6.39 Merge branch 'tegra-arch' into for-2.6.39 ASoC: Staticise non-exported symbols in cs4271 ASoC: Use card rather than soc-audio device to card PM functions ASoC: Replace pdev with card in machine driver probe and remove ASoC: Export card PM callbacks for use in direct registered cards ASoC: Make cache status available via debugfs ASoC: Add subsequence information to seq_notify callbacks Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Add card driver data Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Accept any logical value for WM8962 GPIO set() ASoC: Accept any logical value WM8903 GPIO set() Merge branches 'for-2.6.38' and 'tegra-arch' into for-2.6.39 ASoC: Use snd_pcm_format_width() in snd_soc_params_to_frame_size() Merge branch 'for-2.6.39' of git://git.kernel.org/.../lrg/asoc-2.6 into for-2.6.39 Qiao Zhou (1): ASoC: WM8994: fix wrong value in tristate function Rajashekhara, Sudhakar (1): ASoC: da8xx/omap-l1xx: match codec_name with i2c ids Stephen Warren (12): ASoC: wm8903: Expose GPIOs through gpiolib ARM: tegra: Add Harmony sound platform data type ASoC: Fix mask/val_mask confusion snd_soc_dapm_put_volsw() ASoC: Move card list initialization to snd_soc_register_card ASoC: Tegra: Harmony: Don't use soc-audio platform device ASoC: Tegra: Harmony: Support the internal speaker ASoC: Tegra: Harmony: Fix indentation issue. ASoC: Tegra: Harmony: Use dev_err not pr_err ASoC: Tegra: utils: Don't use global variables ASoC: Tegra: I2S: Use dev_err not pr_err ASoC: Tegra: Harmony: Remove redundant !! ARM: tegra: Add to struct harmony_audio_platform_data arch/arm/mach-shmobile/board-ag5evm.c | 10 - arch/arm/mach-shmobile/board-ap4evb.c | 13 +- arch/arm/mach-shmobile/board-mackerel.c | 13 +- arch/arm/mach-tegra/include/mach/harmony_audio.h | 22 + arch/sh/boards/mach-ecovec24/setup.c | 6 +- arch/sh/boards/mach-se/7724/setup.c | 6 +- include/sound/cs4271.h | 25 + include/sound/sh_fsi.h | 76 +-- include/sound/soc-dapm.h | 16 + include/sound/soc.h | 41 +- include/sound/wm8903.h | 20 +- include/trace/events/asoc.h | 25 + sound/soc/atmel/snd-soc-afeb9260.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 2 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4642.c | 24 + sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/cs4271.c | 659 ++++++++++++++++++++++ sound/soc/codecs/cx20442.c | 3 + sound/soc/codecs/sn95031.c | 281 +++++++++ sound/soc/codecs/twl6040.c | 4 +- sound/soc/codecs/wm8903.c | 127 ++++- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm_hubs.c | 15 +- sound/soc/davinci/davinci-evm.c | 2 +- sound/soc/ep93xx/Kconfig | 9 + sound/soc/ep93xx/Makefile | 2 + sound/soc/ep93xx/edb93xx.c | 142 +++++ sound/soc/fsl/mpc8610_hpcd.c | 6 +- sound/soc/fsl/p1022_ds.c | 6 +- sound/soc/mid-x86/mfld_machine.c | 3 + sound/soc/mid-x86/sst_platform.c | 6 + sound/soc/omap/Kconfig | 1 + sound/soc/omap/ams-delta.c | 2 - sound/soc/omap/rx51.c | 93 +++- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/raumfeld.c | 4 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/tosa.c | 4 +- sound/soc/pxa/zylonite.c | 9 +- sound/soc/samsung/neo1973_gta02_wm8753.c | 6 +- sound/soc/samsung/neo1973_wm8753.c | 7 +- sound/soc/samsung/s3c24xx_simtec_hermes.c | 4 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 4 +- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- sound/soc/sh/fsi-ak4642.c | 13 +- sound/soc/sh/fsi-da7210.c | 13 +- sound/soc/sh/fsi-hdmi.c | 77 +++- sound/soc/sh/fsi.c | 203 ++++--- sound/soc/soc-cache.c | 50 ++- sound/soc/soc-core.c | 114 +++-- sound/soc/soc-dapm.c | 71 ++- sound/soc/soc-utils.c | 23 +- sound/soc/tegra/harmony.c | 170 +++++-- sound/soc/tegra/tegra_asoc_utils.c | 91 ++-- sound/soc/tegra/tegra_asoc_utils.h | 20 +- sound/soc/tegra/tegra_i2s.c | 2 +- 61 files changed, 2136 insertions(+), 437 deletions(-) create mode 100644 arch/arm/mach-tegra/include/mach/harmony_audio.h create mode 100644 include/sound/cs4271.h create mode 100644 sound/soc/codecs/cs4271.c create mode 100644 sound/soc/ep93xx/edb93xx.c

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[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 01 Feb '11

01 Feb '11

The following changes since commit a3adfa00e8089aa72826c6ba04bcb18cfceaf0a9: ASoC: correct link specifications for corgi, poodle and spitz (2011-01-25 15:18:42 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Nothing exciting here except for Stephen's mask/val_mask fix which fixes a fairly wide breakage. Janusz Krzysztofik (2): ASoC: Amstrad Delta: fix const related build error ASoC: CX20442: fix NULL pointer dereference Jarkko Nikula (1): ASoC: Fix module refcount for auxiliary devices Manjunathappa, Prakash (1): ASoC: DaVinci: fix kernel panic due to uninitialized platform_data Stephen Warren (1): ASoC: Fix mask/val_mask confusion snd_soc_dapm_put_volsw() sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/cx20442.c | 3 +++ sound/soc/omap/ams-delta.c | 2 -- sound/soc/soc-core.c | 3 --- sound/soc/soc-dapm.c | 6 +++--- 5 files changed, 7 insertions(+), 9 deletions(-)

2 1

[alsa-devel] Digital IO and SPDIF format
by Torsten Schenk 01 Feb '11

01 Feb '11

Hello, I'm currenty trying to make digital IO working for snd-usb-6fire. I figured out what the card does but I don't know how things (especially input) should be handled in alsa. Unfortunately it is relatively hard to find documentation on that topic so that I decided to ask the pros. The card can receive the whole spdif subframe with all bits (BMW + at least channel status bit and parity bit, but I guess also the other two). It also does receive two spdif-frames per frame (making it possible to receive f.ex. 96kHz digital audio if 6fire's sample rate is set to 48kHz) This also leads to the fact that the 6fire does not drop spdif frames if playback device's sample rate is not very much higher than 6fire's. If the playback device's clock is a bit quicker (up to twice the speed) than the 6fire's one, the 6fire simply captures a frame (when neccesary) that contains two spdif frames. My question is (concerning capturing): How should I handle: - if 6fire's clock and playback device's clock differ, should I use playback device's clock (no interpolation of samples but capturing tool might have invalid sample rate information) or the 6fire's clock (samples need to be interpolated somehow)? - if the 6fire's clock should be used, are small clock drifts acceptable without interpolation? - in case of interpolation: how? - should I receive a whole spdif block (usually 192 spdif frames) and use the information I got from the block (f.ex. left/right channel bits) before copying it to alsa or should I directly copy every frame to alsa? Concerning (also output): Am I correct in assuming that SNDRV_PCM_FORMAT_IEC958_SUBFRAME_XX samples are formatted like in the specification (4 bit preamble, 24 bit audio data, VUCP flags)? If you know of any tool that is able to handle this format, please let me know so that I have a basis to test the whole stuff. Thanks, Torsten

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[alsa-devel] [PATCH] ASoC: soc-core: Ensure codec_reg has fixed length fields
by Dimitris Papastamos 01 Feb '11

01 Feb '11

Make the format of the codec_reg file more easily parsable. Remove the header field which gives the codec name. These changes are important when it comes to extend the debugfs codec_reg file to dump more than PAGE_SIZE bytes to make it easier to calculate offsets within the file. We still need to handle the case when the snd_soc_read() call fails and <no data: %d> is outputted. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 27 ++++++++++++++++++++++++--- 1 files changed, 24 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3feddd9..205cbd7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -69,10 +69,32 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); +/* returns the minimum number of bytes needed to represent + * a particular given value */ +static int min_bytes_needed(unsigned long val) +{ + int c = 0; + int i; + + for (i = (sizeof val * 8) - 1; i >= 0; --i, ++c) + if (val & (1UL << i)) + break; + c = (sizeof val * 8) - c; + if (!c || (c % 8)) + c = (c + 8) / 8; + else + c /= 8; + return c; +} + /* codec register dump */ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) { int ret, i, step = 1, count = 0; + int wordsize, regsize; + + wordsize = codec->driver->reg_word_size * 2; + regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; if (!codec->driver->reg_cache_size) return 0; @@ -80,12 +102,11 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) if (codec->driver->reg_cache_step) step = codec->driver->reg_cache_step; - count += sprintf(buf, "%s registers\n", codec->name); for (i = 0; i < codec->driver->reg_cache_size; i += step) { if (codec->readable_register && !codec->readable_register(codec, i)) continue; - count += sprintf(buf + count, "%2x: ", i); + count += sprintf(buf + count, "%.*x: ", regsize, i); if (count >= PAGE_SIZE - 1) break; @@ -101,7 +122,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) if (ret >= 0) count += snprintf(buf + count, PAGE_SIZE - count, - "%4x", ret); + "%.*x", wordsize, ret); else count += snprintf(buf + count, PAGE_SIZE - count, -- 1.7.3.5

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[alsa-devel] [PATCH] ASoC: CX20442: fix NULL pointer dereference
by Janusz Krzysztofik 01 Feb '11

01 Feb '11

The CX20442 codec driver never provided the snd_soc_codec_driver's .reg_cache_default member. With the latest ASoC framework changes, it seems to be referred unconditionally, resulting in a NULL pointer dereference if missing. Provide it. Created and tested on Amstrad Delta against linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- NOTE: I'm not sure if the value choosen is Amstrad Delta specific or not. sound/soc/codecs/cx20442.c | 3 +++ 1 file changed, 3 insertions(+) --- git/sound/soc/codecs/cx20442.c.orig 2011-01-31 20:09:18.000000000 +0100 +++ git/sound/soc/codecs/cx20442.c 2011-01-31 20:16:20.000000000 +0100 @@ -367,9 +367,12 @@ static int cx20442_codec_remove(struct s return 0; } +static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC; + static struct snd_soc_codec_driver cx20442_codec_dev = { .probe = cx20442_codec_probe, .remove = cx20442_codec_remove, + .reg_cache_default = &cx20442_reg, .reg_cache_size = 1, .reg_word_size = sizeof(u8), .read = cx20442_read_reg_cache,

4 6

[alsa-devel] [PATCH] ASoC: Use snd_pcm_format_width() in snd_soc_params_to_frame_size()
by Mark Brown 01 Feb '11

01 Feb '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-utils.c | 23 +++-------------------- 1 files changed, 3 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 1d07b93..3f45e6a 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -28,26 +28,9 @@ int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params) { int sample_size; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - case SNDRV_PCM_FORMAT_S16_BE: - sample_size = 16; - break; - case SNDRV_PCM_FORMAT_S20_3LE: - case SNDRV_PCM_FORMAT_S20_3BE: - sample_size = 20; - break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S24_BE: - sample_size = 24; - break; - case SNDRV_PCM_FORMAT_S32_LE: - case SNDRV_PCM_FORMAT_S32_BE: - sample_size = 32; - break; - default: - return -ENOTSUPP; - } + sample_size = snd_pcm_format_width(params_format(params)); + if (sample_size < 0) + return sample_size; return snd_soc_calc_frame_size(sample_size, params_channels(params), 1); -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ARM: tegra: Add to struct harmony_audio_platform_data
by Stephen Warren 01 Feb '11

01 Feb '11

Add fields to describe all the audio-related GPIOs on Harmony, except for the codec's GPIO IRQ, which will be passed in its i2c_board_info. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Mark, can you please apply this to your tegra-arch branch, and merge that. Thanks! arch/arm/mach-tegra/include/mach/harmony_audio.h | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-tegra/include/mach/harmony_audio.h b/arch/arm/mach-tegra/include/mach/harmony_audio.h index 5c46391..af08650 100644 --- a/arch/arm/mach-tegra/include/mach/harmony_audio.h +++ b/arch/arm/mach-tegra/include/mach/harmony_audio.h @@ -16,4 +16,7 @@ struct harmony_audio_platform_data { int gpio_spkr_en; + int gpio_hp_det; + int gpio_int_mic_en; + int gpio_ext_mic_en; }; -- 1.7.1

3 2

[alsa-devel] [PATCH] ASoC: Amstrad Delta: fix const related build error
by Janusz Krzysztofik 01 Feb '11

01 Feb '11

The Amstrad Delta ASoC driver used to override the digital_mute() callback, expected to be not provided by the on-board CX20442 CODEC driver, with its own implementation. While this is still posssible when substituting the whole empty snd_soc_dai_driver.ops member (the CX20442 case), replacing snd_soc_dai_ops.digital_mute only is no longer correct after the snd_soc_dai_driver.ops member has been constified, and results in build error. Drop this actually not used code path in hope the CX20442 driver never provides its own snd_soc_dai_ops structure. Created and tested against linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Please consider queuing this patch for the current rc cycle. sound/soc/omap/ams-delta.c | 2 -- 1 file changed, 2 deletions(-) --- git/sound/soc/omap/ams-delta.c.orig 2011-01-31 19:49:29.000000000 +0100 +++ git/sound/soc/omap/ams-delta.c 2011-01-31 23:24:45.000000000 +0100 @@ -507,8 +507,6 @@ static int ams_delta_cx20442_init(struct /* Set up digital mute if not provided by the codec */ if (!codec_dai->driver->ops) { codec_dai->driver->ops = &ams_delta_dai_ops; - } else if (!codec_dai->driver->ops->digital_mute) { - codec_dai->driver->ops->digital_mute = ams_delta_digital_mute; } else { ams_delta_ops.startup = ams_delta_startup; ams_delta_ops.shutdown = ams_delta_shutdown;

3 3

[alsa-devel] [PATCH] ASoC: Tegra: Harmony: Remove redundant !!
by Stephen Warren 01 Feb '11

01 Feb '11

gpio_set_value* should accept logic values not just 0 or 1. The WM8903 GPIO driver has been fixed to work this way, so remove the redundant !! previously required when it didn't accept values >1. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/harmony.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index d1faa63..11e2cb8 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -132,7 +132,7 @@ static int harmony_event_int_spk(struct snd_soc_dapm_widget *w, struct harmony_audio_platform_data *pdata = harmony->pdata; gpio_set_value_cansleep(pdata->gpio_spkr_en, - !!SND_SOC_DAPM_EVENT_ON(event)); + SND_SOC_DAPM_EVENT_ON(event)); return 0; } -- 1.7.1

3 2

[alsa-devel] [PATCH 1/2] ASoC: omap: rx51: Add stereo output support to audio jack
by Jarkko Nikula 01 Feb '11

01 Feb '11

Audio jack in Nokia RX-51/N900 is driven by TPA6130 headphone amplifier. This patch adds support for it and stereo output can be active when "Jack Function" == "TV-OUT" || "Headphone". As the TPA6130 can output very high volume levels the output is limited with snd_soc_limit_volume. Limiting value is found from Maemo kernel sources. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Just repost with Mark's ack added. Previous thread here: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-January/035312.ht… --- sound/soc/omap/Kconfig | 1 + sound/soc/omap/rx51.c | 43 +++++++++++++++++++++++++++++++++++++++---- 2 files changed, 40 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a088db6..b592298 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -24,6 +24,7 @@ config SND_OMAP_SOC_RX51 select OMAP_MCBSP select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X + select SND_SOC_TPA6130A2 help Say Y if you want to add support for SoC audio on Nokia RX-51 hardware. This is also known as Nokia N900 product. diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 09fb0df..251afbe 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -31,6 +31,7 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <plat/mcbsp.h> +#include "../codecs/tpa6130a2.h" #include <asm/mach-types.h> @@ -47,7 +48,8 @@ enum { RX51_JACK_DISABLED, - RX51_JACK_TVOUT, /* tv-out */ + RX51_JACK_TVOUT, /* tv-out with stereo output */ + RX51_JACK_HP, /* headphone: stereo output, no mic */ }; static int rx51_spk_func; @@ -57,6 +59,15 @@ static int rx51_jack_func; static void rx51_ext_control(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; + int hp = 0, tvout = 0; + + switch (rx51_jack_func) { + case RX51_JACK_TVOUT: + tvout = 1; + case RX51_JACK_HP: + hp = 1; + break; + } if (rx51_spk_func) snd_soc_dapm_enable_pin(dapm, "Ext Spk"); @@ -66,9 +77,12 @@ static void rx51_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_enable_pin(dapm, "DMic"); else snd_soc_dapm_disable_pin(dapm, "DMic"); + if (hp) + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - gpio_set_value(RX51_TVOUT_SEL_GPIO, - rx51_jack_func == RX51_JACK_TVOUT); + gpio_set_value(RX51_TVOUT_SEL_GPIO, tvout); snd_soc_dapm_sync(dapm); } @@ -153,6 +167,19 @@ static int rx51_spk_event(struct snd_soc_dapm_widget *w, return 0; } +static int rx51_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = w->dapm->codec; + + if (SND_SOC_DAPM_EVENT_ON(event)) + tpa6130a2_stereo_enable(codec, 1); + else + tpa6130a2_stereo_enable(codec, 0); + + return 0; +} + static int rx51_get_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -212,11 +239,14 @@ static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = { static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event), SND_SOC_DAPM_MIC("DMic", NULL), + SND_SOC_DAPM_HP("Headphone Jack", rx51_hp_event), }; static const struct snd_soc_dapm_route audio_map[] = { {"Ext Spk", NULL, "HPLOUT"}, {"Ext Spk", NULL, "HPROUT"}, + {"Headphone Jack", NULL, "LLOUT"}, + {"Headphone Jack", NULL, "RLOUT"}, {"DMic Rate 64", NULL, "Mic Bias 2V"}, {"Mic Bias 2V", NULL, "DMic"}, @@ -224,7 +254,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static const char *spk_function[] = {"Off", "On"}; static const char *input_function[] = {"ADC", "Digital Mic"}; -static const char *jack_function[] = {"Off", "TV-OUT"}; +static const char *jack_function[] = {"Off", "TV-OUT", "Headphone"}; static const struct soc_enum rx51_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), @@ -265,6 +295,11 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) /* Set up RX-51 specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + err = tpa6130a2_add_controls(codec); + if (err < 0) + return err; + snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); + snd_soc_dapm_sync(dapm); /* AV jack detection */ -- 1.7.2.3

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[alsa-devel] [PATCH 1/3] ASoC: soc_jack - add function to determine jack type
by Harsha Priya 01 Feb '11

01 Feb '11

This patch adds a helper function that determines the jack type given the mic bias value. For any codec that would give the mic bias value on a jack insertion, this function would return the type of jack based on mic bias range Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/soc.h | 15 +++++++++++++++ sound/soc/soc-jack.c | 18 ++++++++++++++++++ 2 files changed, 33 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ecdaef..afe98eb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -246,6 +246,19 @@ typedef int (*hw_write_t)(void *,const char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; +enum soc_mic_bias_zones { + /* mic bias volutage range for Headphones*/ + SND_SOC_JACK_HP_START = 0, + SND_SOC_JACK_HP_END = 400, + /* mic bias volutage range for American Headset*/ + SND_SOC_JACK_AM_HS_START = 400, + SND_SOC_JACK_AM_HS_END = 650, + /* mic bias volutage range for Headset*/ + SND_SOC_JACK_HS_START = 650, + SND_SOC_JACK_HS_END = 2000, + SND_SOC_UNDEFINED, +}; + enum snd_soc_control_type { SND_SOC_CUSTOM, SND_SOC_I2C, @@ -307,6 +320,8 @@ void snd_soc_jack_notifier_register(struct snd_soc_jack *jack, struct notifier_block *nb); void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack, struct notifier_block *nb); +int snd_soc_jack_get_type(int mic_bias); + #ifdef CONFIG_GPIOLIB int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ac5a5bc..462965a 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -186,6 +186,24 @@ void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack, } EXPORT_SYMBOL_GPL(snd_soc_jack_notifier_unregister); +/** + * snd_soc_jack_get_type - Based on the mic bias value, this function returns + * the type of jack + * + * @mic_bias: mic bias value detected + * + * Based on the mic bias value passed, this function helps identify + * the type of jack as either SND_JACK_HEADSET or SND_JACK_HEADPHONE. + */ +int snd_soc_jack_get_type(int mic_bias) +{ + if (mic_bias >= SND_SOC_JACK_HP_START && mic_bias < SND_SOC_JACK_HS_END) + return SND_JACK_HEADSET; + else + return SND_JACK_HEADPHONE; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_get_type); + #ifdef CONFIG_GPIOLIB /* gpio detect */ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) -- 1.7.3.4

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[alsa-devel] [PATCH] ASoC: Support 8 bit samples in soc-util
by Mark Brown 01 Feb '11

01 Feb '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-utils.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 1d07b93..e31d843 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -29,6 +29,10 @@ int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params) int sample_size; switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + case SNDRV_PCM_FORMAT_U8: + sample_size = 8; + break; case SNDRV_PCM_FORMAT_S16_LE: case SNDRV_PCM_FORMAT_S16_BE: sample_size = 16; -- 1.7.2.3

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[alsa-devel] No soundcards found with 2.6.37 (and 2.6.38-rc2)
by Sven Neumann 01 Feb '11

01 Feb '11

Hi, we are using an ARM PXA300 based device with SoC devices. The code is completely upstream, our platform is defined in sound/soc/pxa/raumfeld.c I've recently tried to update our kernel to 2.6.37, but it looks like there have been some changes in the Linux kernel that cause our sound hardware not to be recognized or initialised correctly any longer. This is the output when booting Linux 2.6.36.3: [ 26.909242] Asahi Kasei AK4104 ALSA SoC Codec Driver [ 26.915118] No device for DAI ak4104 [ 26.918680] ak4104 spi0.0: SPI device initialized [ 26.923870] Cirrus Logic CS4270 ALSA SoC Codec Driver [ 26.929471] cs4270 0-0048: found device at i2c address 48 [ 26.934935] cs4270 0-0048: hardware revision 3 [ 26.941557] No device for DAI pxa2xx-ssp1 [ 26.945554] No device for DAI pxa2xx-ssp2 [ 26.949533] No device for DAI pxa2xx-ssp3 [ 26.953590] No device for DAI pxa2xx-ssp4 [ 26.959814] asoc: cs4270 <-> pxa2xx-ssp1 mapping ok [ 26.970007] asoc: ak4104 <-> pxa2xx-ssp2 mapping ok [ 27.080719] ALSA device list: [ 27.083690] #0: Raumfeld analog (CS4270) [ 27.087750] #1: Raumfeld S/PDIF (ak4104) And here are the relevant lines from booting 2.6.37: [ 26.927802] Asahi Kasei AK4104 ALSA SoC Codec Driver [ 26.933269] Cirrus Logic CS4270 ALSA SoC Codec Driver [ 26.938886] cs4270-codec 0-0048: found device at i2c address 48 [ 26.944774] cs4270-codec 0-0048: hardware revision 3 [ 27.058972] ALSA device list: [ 27.061933] No soundcards found. Based on this output I've done a git bisect. Unfortunately there is a larger series of commits that don't even compile, so I have not been able to find out the exact commit that broke it, but I've boiled it down to a number of commits: There are only 'skip'ped commits left to test. The first bad commit could be any of: 5dcba5d6741d4533e0ef696507f93f2a4c738efb cf7af01aa77ec1b17687f5328ce0a598709efd59 54d8d0aeb96e677c129918b391b9e74d48e65294 a6d14342dc46f0ef94cccf323fd9481de84b458f 48bd3472d6512c74257706f7c38ad10a7117bac7 960d0697919aef453273e3a0ccc87daf52d808ac 381ac990db3f4fdfbe91e3a636cab03edf293b9c 38445af3bc0658615fd9a724fbcb67148eac2cd3 13cb61f8c261ca6a218f83f4ee9c3bd5cfc223a5 af3751a0bfe4d461e0567961dc06448f2128e26f 26e277d7151abcef8efa1e6f1f0d1420d4ed17c6 6ba6c9c3414b95c3ed50d529fb0efa744ed45761 10e2f11326e0a263f0336686454be67efb2b56dc ff71334a46844d0ae6ff0055c549790bcd27bb10 87a0632b29410bab5c1783d7eb979c8d942d4209 dd99a4524bd9f82358e9a20a44c5e784baa4e702 1a3c5a491af6756dbba6ee166a9dee72bb414ba8 f5d1e5ed58dc0178e1678db63156772999379d49 6e6f66226f0092a39526f8d6f02ebb447d995be2 6f341d14811550d863ba804ce6ec7757a7145081 eaae183f4b3f25522cd13ebf30162c1cb863b1f1 9d37e8947c613190b7b92c63830e638ffdb08e48 e231cab0a4b5844ae13a9584433ca3b9b204629e f51582fd8d9b1196d58cd94c2b4b759cc1baf57a 3782a528974c9607bde03ec265bbebd2dc300ea1 698cb111f44d81dd3a604ba2a7f3f865b66b9ddf 8ef339df25ed424e7430fd411a52840c6af368c6 9a76f1ff6e299fbb04149fe15aff061351fd0dab c5607d8e7a4c30d2ff62b8eefe3f977d5c71d2fe f0fba2ad1b6b53d5360125c41953b7afcd6deff0 ba2772edbe1f03a695029ca82844615fe41d28fc We cannot bisect more! As far as I can see these patches are all part of the introduction of ASoC Multi-Component Support. Perhaps someone who's been working on this could have a look and give me some hints on how to proceed further? I've already tried 2.6.38-rc2 with same results as with 2.6.37. Thanks, Sven -- Sven Neumann Head of RAUMFELD Software Development Lautsprecher Teufel GmbH | Bülowstr. 66 | 10783 Berlin | Germany Tel: +49 (0)30-300 930 153 | s.neumann(a)raumfeld.com

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[alsa-devel] Questions about Multi-streaming playback with HDA codec ad1988b
by Raymond Yau 01 Feb '11

01 Feb '11

The patch is to provide a new model which front panel headphone of a 6stack-dig desktop can be used for playback a stereo stream independent of the rear panel ad1988b is a HDA codec support 10 channel ( enough DAC for stereo + surround71 ) 1) Is this approach to create a new model correct or just need to implement a switch for activatie the "independent head phone" like those VIA codecs ? struct ad198x_spec { + hda_nid_t *alt_dac_nid; + struct hda_pcm_stream *stream_analog_alt_playback; } +static hda_nid_t ad1988_6stack_alt_dac_nid[1] = { + 0x03 +}; static const char *ad1988_models[AD1988_MODEL_LAST] = { [AD1988_6STACK] = "6stack", [AD1988_6STACK_DIG] = "6stack-dig", + [AD1988_6STACK_DIG_FP] = "6stack-dig-fp", [AD1988_3STACK] = "3stack", [AD1988_3STACK_DIG] = "3stack-dig", [AD1988_LAPTOP] = "laptop", [AD1988_LAPTOP_DIG] = "laptop-dig", [AD1988_AUTO] = "auto", }; enum { AD1988_6STACK, AD1988_6STACK_DIG, + AD1988_6STACK_DIG_FP, AD1988_3STACK, AD1988_3STACK_DIG, AD1988_LAPTOP, AD1988_LAPTOP_DIG, AD1988_AUTO, AD1988_MODEL_LAST, }; 2) do the driver need to define .prepare and .cleanup function or just need to use "hda_pcm_default_prepare() and hda_pcm_default_cleanup() static int ad198x_alt_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; snd_hda_codec_setup_stream(codec, spec->alt_dac_nid[0], stream_tag, 0, format); return 0; } static int ad198x_alt_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; snd_hda_codec_cleanup_stream(codec, spec->alt_dac_nid[0]); return 0; } static struct hda_pcm_stream ad198x_pcm_analog_alt_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, /* NID is set in ad198x_build_pcms */ .ops = { .prepare = ad198x_alt_playback_pcm_prepare, .cleanup = ad198x_alt_playback_pcm_cleanup }, }; 3) Is it correct to use device 2 for analog alternate playback since some HDA codec implement this feature on "hw:0,0,1" and realtek driver create alternate capture on "hw:0,2" , this seem create a lot of problem of the application (e.g. PA server ) ? static int ad198x_build_pcms(struct hda_codec *codec) { if (spec->alt_dac_nid && spec->stream_analog_alt_playback) { codec->num_pcms++; info = spec->pcm_rec + 2; info->name = "AD198x Alternate Analog"; info->pcm_type = HDA_PCM_TYPE_AUDIO; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *spec->stream_analog_alt_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->alt_dac_nid[0]; } } static struct snd_kcontrol_new ad1988_6stack_fp_mixers[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT), HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT), HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT), HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT), HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT), { } /* end */ }; static struct hda_verb ad1988_6stack_fp_init_verbs[] = { /* Front, Surround, CLFE, side DAC; unmute as default */ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Headphone; unmute as default */ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Port-A front headphon path */ {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC:03h */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, /* Port-D line-out path */ {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Port-F surround path */ {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Port-G CLFE path */ {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Port-H side path */ {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Mono out path */ {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */ /* Port-B front mic-in path */ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* Port-C line-in path */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x33, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Port-E mic-in path */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Analog CD Input */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Analog Mix output amp */ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ { } }; static int patch_ad1988(struct hda_codec *codec) { switch (board_config) { case AD1988_6STACK_DIG_FP: spec->multiout.max_channels = 8; spec->multiout.num_dacs = 4; spec->multiout.dac_nids = ad1988_6stack_dac_nids; spec->alt_dac_nid = ad1988_6stack_alt_dac_nid; spec->stream_analog_alt_playback = &ad198x_pcm_analog_alt_playback; spec->input_mux = &ad1988_6stack_capture_source; spec->num_mixers = 1; spec->mixers[0] = ad1988_6stack_fp_mixers; spec->num_init_verbs = 1; spec->init_verbs[0] = ad1988_6stack_fp_init_verbs; spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; spec->dig_in_nid = AD1988_SPDIF_IN; break; 3) What functions are still missing in this model ? a), do this new model need to disable hp_automute() function b) any change in ad198x_power_eapd_xxx() functions ? c) idenifty ad1988a and ad1988b

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[alsa-devel] ALSA 1.0.24 warning when compiling on 32bits machine
by Raymond Yau 01 Feb '11

01 Feb '11

CC [M] /alsa-driver-1.0.24/acore/pcm_native.o /alsa-driver-1.0.24/acore/pcm_native.c: In function ‘snd_pcm_mmap_status’: /alsa-driver-1.0.24/acore/pcm_native.c:3171: warning: assignment discards qualifiers from pointer target type /alsa-driver-1.0.24/acore/pcm_native.c: In function ‘snd_pcm_mmap_control’: /allsa-driver-1.0.24/acore/pcm_native.c:3255: warning: assignment discards qualifiers from pointer target type /alsa-driver-1.0.24/acore/pcm_native.c: In function ‘snd_pcm_default_mmap’: /allsa-driver-1.0.24/acore/pcm_native.c:3437: warning: assignment discards qualifiers from pointer target type /alsa-driver-1.0.24/acore/pcm_native.c: In function ‘snd_pcm_mmap_data’: /alsa-driver-1.0.24/acore/pcm_native.c:3524: warning: assignment discards qualifiers from pointer target type CC [M] /alsa-driver-1.0.24/pci/emu10k1/emufx.o In file included from /home/raymond/alsa-driver-1.0.24/pci/emu10k1/emufx.c:3: /alsa-driver-1.0.24/pci/emu10k1/../../alsa-kernel/pci/emu10k1/emufx.c: In function ‘__check_high_res_gpr_volume’: /alsa-driver-1.0.24/pci/emu10k1/../../alsa-kernel/pci/emu10k1/emufx.c:55: warning: return from incompatible pointer type CC [M] /alsa-driver-1.0.24/pci/hda/patch_hdmi.o In file included from /home/raymond/alsa-driver-1.0.24/pci/hda/patch_hdmi.c:3: /alsa-driver-1.0.24/pci/hda/../../alsa-kernel/pci/hda/patch_hdmi.c: In function ‘__check_static_hdmi_pcm’: /alsa-driver-1.0.24/pci/hda/../../alsa-kernel/pci/hda/patch_hdmi.c:40: warning: return from incompatible pointer type CC [M] /alsa-driver-1.0.24/usb/usx2y/us122l.o In file included from /allsa-driver-1.0.24/usb/usx2y/us122l.c:2: /alsa-driver-1.0.24/usb/usx2y/../../alsa-kernel/usb/usx2y/us122l.c: In function ‘usb_stream_hwdep_mmap’: /alsa-driver-1.0.24/usb/usx2y/../../alsa-kernel/usb/usx2y/us122l.c:263: warning: assignment discards qualifiers from pointer target type CC [M] /alsa-driver-1.0.24/usb/usx2y/usbusx2y.o CC [M] /alsa-driver-1.0.24/usb/usx2y/usX2Yhwdep.o /alsa-driver-1.0.24/usb/usx2y/usX2Yhwdep.c: In function ‘snd_us428ctls_mmap’: /alsa-driver-1.0.24/usb/usx2y/usX2Yhwdep.c:144: warning: assignment discards qualifiers from pointer target type CC [M] /alsa-driver-1.0.24/usb/usx2y/usx2yhwdeppcm.o /alsa-driver-1.0.24/usb/usx2y/usx2yhwdeppcm.c: In function ‘snd_usX2Y_hwdep_pcm_mmap’: /alsa-driver-1.0.24/usb/usx2y/usx2yhwdeppcm.c:805: warning: assignment discards qualifiers from pointer target type

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[alsa-devel] Problem with VIA VT1708S and git versions of alsa-driver.
by Mark Goldstein 31 Jan '11

31 Jan '11

Hello, Recently I'm having some strange issues with my VIA VT1708S on-board sound card. The description might be a bit long, so I'll give a brief summary first. I have ASUS P5QL/EPU motherboard with VT1708S 8-channel HDA Codec. There are 3 OSs installed on this machine: openSUSE 11.1 (my main system), openSUSE 11.3, Kubuntu 10.04. Audio worked fine until some updates at the beginning of September. Since then I have 2 different issues. 1).On 11.1 - Line In control seems inactive, instead Front Mic control controls the level from Line In jack. Front Mic is constanly disabled. alsaconf does not show hda-intel in the list of detected cards. Latest version it happened with is git20100925. When I compiled stable 1.0.23 sources, everything started working as expected again. I reported this issue on bugtrack: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5133 2) On oS 11.3 I start experiencing problems recording audio either from Mic or from Line. After some experiments I found out that the Capture1 control is somehow broken. Mute switch is reversed. When it is unmuted, there is no audio recorded. When it is muted, recording works. The same happened to at least one more user of openSUSE, see the thread "How to investigate the problem with MIC" on alsa-user mailing list. I've also added my comments to issue https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5130 but now I think it is different issue. Even when I have audio recorded normally when Capture1 is muted, I still have very distorted voice from Mic in Skype. It worked fine before the problem started. More details: On oS 11.1 I had my audio working fine after updating to 1.0.23 versions of alsa driver. I'm using opensuse/multimedia: repository that provides binary rpm of alsa-driver git snapshots. I have only one current kernel 2.6.27.48 on 11.1. On oS 11.3 audio worked right out of the box, since it uses kernels with alsa-driver 1.0.23. I have 2-3 kernels installed at the same time - normal 2.6.34 from official repository and 2.6.35 from kernel build service. I'm not sure when exactly the problems started, but definitely at the beginning of September. I've initially saw that on oS 11.1 I can't hear the audio from my TV card connected to Line In. Then I noticed that Mic does not work in Skype on both 11.1 and 11.3, while it still works on Kubuntu. When I unmuted Front Mic on 11.1 I've suddenly found that it actually enabled Line In. I think now that something went bad with card detecting. Probably on 11.1 with latest git versions my card is not properly detected and some "defaults" are used. I'm attaching here 2 outputs of alsa-info.sh - 11.1_stable captured with alsa-driver compiled from stable 1.0.23 sources. 11.1_git25 captured with driver compiled with git20100925 version from opensuse/multimedia: repository. For me the problem with 11.1 is resolved, but probably it's worth checking why latest version from git behave this way. I'll continue with 11.3 investigation later on. Regards, -- Mark Goldstein

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[alsa-devel] [PATCH] ASoC: Accept any logical value for WM8962 GPIO set()
by Mark Brown 31 Jan '11

31 Jan '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 5c7b730..3b71dd6 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3635,7 +3635,7 @@ static void wm8962_gpio_set(struct gpio_chip *chip, unsigned offset, int value) struct snd_soc_codec *codec = wm8962->codec; snd_soc_update_bits(codec, WM8962_GPIO_BASE + offset, - WM8962_GP2_LVL, value << WM8962_GP2_LVL_SHIFT); + WM8962_GP2_LVL, !!value << WM8962_GP2_LVL_SHIFT); } static int wm8962_gpio_direction_out(struct gpio_chip *chip, -- 1.7.2.3

2 1

[PATCH] OMAP4: hwmod data: Add McBSP
by Benoit Cousson 31 Jan '11

31 Jan '11

Add mcbsp data including a revision member in hwmod_class in order to provide mcbsp revision information for different omap. Signed-off-by: Benoit Cousson <b-cousson(a)ti.com> Signed-off-by: Kishon Vijay Abraham I <kishon(a)ti.com> Signed-off-by: Charulatha V <charu(a)ti.com> [b-cousson(a)ti.com: Remove the mcbsp4 memory name, re-order properly the structures, update changelog] --- arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 301 +++++++++++++++++++++++++++- arch/arm/plat-omap/include/plat/mcbsp.h | 1 + 2 files changed, 298 insertions(+), 4 deletions(-) diff --git a/arch/arm/mach-omap2/omap_hwmod_44xx_data.c b/arch/arm/mach-omap2/omap_hwmod_44xx_data.c index 4ff4eff..0616fb2 100644 --- a/arch/arm/mach-omap2/omap_hwmod_44xx_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_44xx_data.c @@ -26,6 +26,7 @@ #include <plat/dma.h> #include <plat/mcspi.h> #include <plat/dmtimer.h> +#include <plat/mcbsp.h> #include "omap_hwmod_common_data.h" @@ -534,10 +535,6 @@ static struct omap_hwmod omap44xx_mpu_private_hwmod = { * kbd * mailbox * mcasp - * mcbsp1 - * mcbsp2 - * mcbsp3 - * mcbsp4 * mcpdm * mmc1 * mmc2 @@ -2008,6 +2005,296 @@ static struct omap_hwmod omap44xx_iva_hwmod = { }; /* + * 'mcbsp' class + * multi channel buffered serial port controller + */ + +static struct omap_hwmod_class_sysconfig omap44xx_mcbsp_sysc = { + .sysc_offs = 0x008c, + .sysc_flags = (SYSC_HAS_CLOCKACTIVITY | SYSC_HAS_ENAWAKEUP | + SYSC_HAS_SIDLEMODE | SYSC_HAS_SOFTRESET), + .idlemodes = (SIDLE_FORCE | SIDLE_NO | SIDLE_SMART), + .sysc_fields = &omap_hwmod_sysc_type1, +}; + +static struct omap_hwmod_class omap44xx_mcbsp_hwmod_class = { + .name = "mcbsp", + .sysc = &omap44xx_mcbsp_sysc, + .rev = MCBSP_CONFIG_TYPE4, +}; + +/* mcbsp1 */ +static struct omap_hwmod omap44xx_mcbsp1_hwmod; +static struct omap_hwmod_irq_info omap44xx_mcbsp1_irqs[] = { + { .irq = 17 + OMAP44XX_IRQ_GIC_START }, +}; + +static struct omap_hwmod_dma_info omap44xx_mcbsp1_sdma_reqs[] = { + { .name = "tx", .dma_req = 32 + OMAP44XX_DMA_REQ_START }, + { .name = "rx", .dma_req = 33 + OMAP44XX_DMA_REQ_START }, +}; + +static struct omap_hwmod_addr_space omap44xx_mcbsp1_addrs[] = { + { + .name = "mpu", + .pa_start = 0x40122000, + .pa_end = 0x401220ff, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_abe -> mcbsp1 */ +static struct omap_hwmod_ocp_if omap44xx_l4_abe__mcbsp1 = { + .master = &omap44xx_l4_abe_hwmod, + .slave = &omap44xx_mcbsp1_hwmod, + .clk = "ocp_abe_iclk", + .addr = omap44xx_mcbsp1_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_mcbsp1_addrs), + .user = OCP_USER_MPU, +}; + +static struct omap_hwmod_addr_space omap44xx_mcbsp1_dma_addrs[] = { + { + .name = "dma", + .pa_start = 0x49022000, + .pa_end = 0x490220ff, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_abe -> mcbsp1 (dma) */ +static struct omap_hwmod_ocp_if omap44xx_l4_abe__mcbsp1_dma = { + .master = &omap44xx_l4_abe_hwmod, + .slave = &omap44xx_mcbsp1_hwmod, + .clk = "ocp_abe_iclk", + .addr = omap44xx_mcbsp1_dma_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_mcbsp1_dma_addrs), + .user = OCP_USER_SDMA, +}; + +/* mcbsp1 slave ports */ +static struct omap_hwmod_ocp_if *omap44xx_mcbsp1_slaves[] = { + &omap44xx_l4_abe__mcbsp1, + &omap44xx_l4_abe__mcbsp1_dma, +}; + +static struct omap_hwmod omap44xx_mcbsp1_hwmod = { + .name = "mcbsp1", + .class = &omap44xx_mcbsp_hwmod_class, + .mpu_irqs = omap44xx_mcbsp1_irqs, + .mpu_irqs_cnt = ARRAY_SIZE(omap44xx_mcbsp1_irqs), + .sdma_reqs = omap44xx_mcbsp1_sdma_reqs, + .sdma_reqs_cnt = ARRAY_SIZE(omap44xx_mcbsp1_sdma_reqs), + .main_clk = "mcbsp1_fck", + .prcm = { + .omap4 = { + .clkctrl_reg = OMAP4430_CM1_ABE_MCBSP1_CLKCTRL, + }, + }, + .slaves = omap44xx_mcbsp1_slaves, + .slaves_cnt = ARRAY_SIZE(omap44xx_mcbsp1_slaves), + .omap_chip = OMAP_CHIP_INIT(CHIP_IS_OMAP4430), +}; + +/* mcbsp2 */ +static struct omap_hwmod omap44xx_mcbsp2_hwmod; +static struct omap_hwmod_irq_info omap44xx_mcbsp2_irqs[] = { + { .irq = 22 + OMAP44XX_IRQ_GIC_START }, +}; + +static struct omap_hwmod_dma_info omap44xx_mcbsp2_sdma_reqs[] = { + { .name = "tx", .dma_req = 16 + OMAP44XX_DMA_REQ_START }, + { .name = "rx", .dma_req = 17 + OMAP44XX_DMA_REQ_START }, +}; + +static struct omap_hwmod_addr_space omap44xx_mcbsp2_addrs[] = { + { + .name = "mpu", + .pa_start = 0x40124000, + .pa_end = 0x401240ff, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_abe -> mcbsp2 */ +static struct omap_hwmod_ocp_if omap44xx_l4_abe__mcbsp2 = { + .master = &omap44xx_l4_abe_hwmod, + .slave = &omap44xx_mcbsp2_hwmod, + .clk = "ocp_abe_iclk", + .addr = omap44xx_mcbsp2_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_mcbsp2_addrs), + .user = OCP_USER_MPU, +}; + +static struct omap_hwmod_addr_space omap44xx_mcbsp2_dma_addrs[] = { + { + .name = "dma", + .pa_start = 0x49024000, + .pa_end = 0x490240ff, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_abe -> mcbsp2 (dma) */ +static struct omap_hwmod_ocp_if omap44xx_l4_abe__mcbsp2_dma = { + .master = &omap44xx_l4_abe_hwmod, + .slave = &omap44xx_mcbsp2_hwmod, + .clk = "ocp_abe_iclk", + .addr = omap44xx_mcbsp2_dma_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_mcbsp2_dma_addrs), + .user = OCP_USER_SDMA, +}; + +/* mcbsp2 slave ports */ +static struct omap_hwmod_ocp_if *omap44xx_mcbsp2_slaves[] = { + &omap44xx_l4_abe__mcbsp2, + &omap44xx_l4_abe__mcbsp2_dma, +}; + +static struct omap_hwmod omap44xx_mcbsp2_hwmod = { + .name = "mcbsp2", + .class = &omap44xx_mcbsp_hwmod_class, + .mpu_irqs = omap44xx_mcbsp2_irqs, + .mpu_irqs_cnt = ARRAY_SIZE(omap44xx_mcbsp2_irqs), + .sdma_reqs = omap44xx_mcbsp2_sdma_reqs, + .sdma_reqs_cnt = ARRAY_SIZE(omap44xx_mcbsp2_sdma_reqs), + .main_clk = "mcbsp2_fck", + .prcm = { + .omap4 = { + .clkctrl_reg = OMAP4430_CM1_ABE_MCBSP2_CLKCTRL, + }, + }, + .slaves = omap44xx_mcbsp2_slaves, + .slaves_cnt = ARRAY_SIZE(omap44xx_mcbsp2_slaves), + .omap_chip = OMAP_CHIP_INIT(CHIP_IS_OMAP4430), +}; + +/* mcbsp3 */ +static struct omap_hwmod omap44xx_mcbsp3_hwmod; +static struct omap_hwmod_irq_info omap44xx_mcbsp3_irqs[] = { + { .irq = 23 + OMAP44XX_IRQ_GIC_START }, +}; + +static struct omap_hwmod_dma_info omap44xx_mcbsp3_sdma_reqs[] = { + { .name = "tx", .dma_req = 18 + OMAP44XX_DMA_REQ_START }, + { .name = "rx", .dma_req = 19 + OMAP44XX_DMA_REQ_START }, +}; + +static struct omap_hwmod_addr_space omap44xx_mcbsp3_addrs[] = { + { + .name = "mpu", + .pa_start = 0x40126000, + .pa_end = 0x401260ff, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_abe -> mcbsp3 */ +static struct omap_hwmod_ocp_if omap44xx_l4_abe__mcbsp3 = { + .master = &omap44xx_l4_abe_hwmod, + .slave = &omap44xx_mcbsp3_hwmod, + .clk = "ocp_abe_iclk", + .addr = omap44xx_mcbsp3_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_mcbsp3_addrs), + .user = OCP_USER_MPU, +}; + +static struct omap_hwmod_addr_space omap44xx_mcbsp3_dma_addrs[] = { + { + .name = "dma", + .pa_start = 0x49026000, + .pa_end = 0x490260ff, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_abe -> mcbsp3 (dma) */ +static struct omap_hwmod_ocp_if omap44xx_l4_abe__mcbsp3_dma = { + .master = &omap44xx_l4_abe_hwmod, + .slave = &omap44xx_mcbsp3_hwmod, + .clk = "ocp_abe_iclk", + .addr = omap44xx_mcbsp3_dma_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_mcbsp3_dma_addrs), + .user = OCP_USER_SDMA, +}; + +/* mcbsp3 slave ports */ +static struct omap_hwmod_ocp_if *omap44xx_mcbsp3_slaves[] = { + &omap44xx_l4_abe__mcbsp3, + &omap44xx_l4_abe__mcbsp3_dma, +}; + +static struct omap_hwmod omap44xx_mcbsp3_hwmod = { + .name = "mcbsp3", + .class = &omap44xx_mcbsp_hwmod_class, + .mpu_irqs = omap44xx_mcbsp3_irqs, + .mpu_irqs_cnt = ARRAY_SIZE(omap44xx_mcbsp3_irqs), + .sdma_reqs = omap44xx_mcbsp3_sdma_reqs, + .sdma_reqs_cnt = ARRAY_SIZE(omap44xx_mcbsp3_sdma_reqs), + .main_clk = "mcbsp3_fck", + .prcm = { + .omap4 = { + .clkctrl_reg = OMAP4430_CM1_ABE_MCBSP3_CLKCTRL, + }, + }, + .slaves = omap44xx_mcbsp3_slaves, + .slaves_cnt = ARRAY_SIZE(omap44xx_mcbsp3_slaves), + .omap_chip = OMAP_CHIP_INIT(CHIP_IS_OMAP4430), +}; + +/* mcbsp4 */ +static struct omap_hwmod omap44xx_mcbsp4_hwmod; +static struct omap_hwmod_irq_info omap44xx_mcbsp4_irqs[] = { + { .irq = 16 + OMAP44XX_IRQ_GIC_START }, +}; + +static struct omap_hwmod_dma_info omap44xx_mcbsp4_sdma_reqs[] = { + { .name = "tx", .dma_req = 30 + OMAP44XX_DMA_REQ_START }, + { .name = "rx", .dma_req = 31 + OMAP44XX_DMA_REQ_START }, +}; + +static struct omap_hwmod_addr_space omap44xx_mcbsp4_addrs[] = { + { + .pa_start = 0x48096000, + .pa_end = 0x480960ff, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_per -> mcbsp4 */ +static struct omap_hwmod_ocp_if omap44xx_l4_per__mcbsp4 = { + .master = &omap44xx_l4_per_hwmod, + .slave = &omap44xx_mcbsp4_hwmod, + .clk = "l4_div_ck", + .addr = omap44xx_mcbsp4_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_mcbsp4_addrs), + .user = OCP_USER_MPU | OCP_USER_SDMA, +}; + +/* mcbsp4 slave ports */ +static struct omap_hwmod_ocp_if *omap44xx_mcbsp4_slaves[] = { + &omap44xx_l4_per__mcbsp4, +}; + +static struct omap_hwmod omap44xx_mcbsp4_hwmod = { + .name = "mcbsp4", + .class = &omap44xx_mcbsp_hwmod_class, + .mpu_irqs = omap44xx_mcbsp4_irqs, + .mpu_irqs_cnt = ARRAY_SIZE(omap44xx_mcbsp4_irqs), + .sdma_reqs = omap44xx_mcbsp4_sdma_reqs, + .sdma_reqs_cnt = ARRAY_SIZE(omap44xx_mcbsp4_sdma_reqs), + .main_clk = "mcbsp4_fck", + .prcm = { + .omap4 = { + .clkctrl_reg = OMAP4430_CM_L4PER_MCBSP4_CLKCTRL, + }, + }, + .slaves = omap44xx_mcbsp4_slaves, + .slaves_cnt = ARRAY_SIZE(omap44xx_mcbsp4_slaves), + .omap_chip = OMAP_CHIP_INIT(CHIP_IS_OMAP4430), +}; + +/* * 'mcspi' class * multichannel serial port interface (mcspi) / master/slave synchronous serial * bus @@ -3568,6 +3855,12 @@ static __initdata struct omap_hwmod *omap44xx_hwmods[] = { &omap44xx_iva_seq0_hwmod, &omap44xx_iva_seq1_hwmod, + /* mcbsp class */ + &omap44xx_mcbsp1_hwmod, + &omap44xx_mcbsp2_hwmod, + &omap44xx_mcbsp3_hwmod, + &omap44xx_mcbsp4_hwmod, + /* mcspi class */ &omap44xx_mcspi1_hwmod, &omap44xx_mcspi2_hwmod, diff --git a/arch/arm/plat-omap/include/plat/mcbsp.h b/arch/arm/plat-omap/include/plat/mcbsp.h index 8d8434b..06bd1dc 100644 --- a/arch/arm/plat-omap/include/plat/mcbsp.h +++ b/arch/arm/plat-omap/include/plat/mcbsp.h @@ -39,6 +39,7 @@ static struct platform_device omap_mcbsp##port_nr = { \ #define MCBSP_CONFIG_TYPE2 0x2 #define MCBSP_CONFIG_TYPE3 0x3 +#define MCBSP_CONFIG_TYPE4 0x4 #define OMAP7XX_MCBSP1_BASE 0xfffb1000 #define OMAP7XX_MCBSP2_BASE 0xfffb1800 -- 1.7.0.4

1 0

[alsa-devel] [PATCH] ASoC: Accept any logical value WM8903 GPIO set()
by Mark Brown 31 Jan '11

31 Jan '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 54383c3..27cd8f5 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1904,7 +1904,8 @@ static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) struct snd_soc_codec *codec = wm8903->codec; snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - WM8903_GP1_LVL_MASK, value << WM8903_GP1_LVL_SHIFT); + WM8903_GP1_LVL_MASK, + !!value << WM8903_GP1_LVL_SHIFT); } static struct gpio_chip wm8903_template_chip = { -- 1.7.2.3

2 1

[alsa-devel] [PATCH] soc: neo1973_wm8753 audio support does not require scoop
by H Hartley Sweeten 31 Jan '11

31 Jan '11

This driver does not use any of the functionality provided by the scoop hardware. Remove the unneeded header. Signed-off-by: H Hartley Sweeten <hsweeten(a)visionengravers.com> --- diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index c7a2451..c5b3113 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -24,7 +24,6 @@ #include <sound/tlv.h> #include <asm/mach-types.h> -#include <asm/hardware/scoop.h> #include <mach/regs-clock.h> #include <mach/regs-gpio.h> #include <mach/hardware.h>

3 3

[alsa-devel] Hello, first mail and a quesion.
by Matías Goinheix Freitas 31 Jan '11

31 Jan '11

Hello people, This is my first mail here. Before purchasing the cheap awful USB interface I have now, I was in love with the Saffire 6 USB. Sadly it didn't run on linux. A friend of mine got it (he uses another OS) and so I tested for myself that it's not recognized by Ubuntu 10.04. My question is this: I have access to that interface because I can borrow it anytime I want. I don't have the slightest idea on developing a driver or hacking stuff (I suscribed to this mailing list because I thought it's the one I could have contact with developers even though I'm not, if I'm in the wrong place I will unsuscribe it's not my intention to bother). Is there any way I could help developers interested in creating a driver for this interfase? I don't know if that makes sense, but maybe there's a project going on that I'm unaware of and they could use some info about that interface. I could copy/paste/send them the info they needed, or run the commands they tell me to run and give them the info. Thank you, Matías

1 0

[alsa-devel] [PATCH 1/7] ASoC: Move card list initialization to snd_soc_register_card
by Stephen Warren 31 Jan '11

31 Jan '11

All ASoC cards need snd_soc_initialize_card_lists called. Previously, it was only called for cards backed by a "soc-audio" platform device, via soc_probe(). However, it's also needed for cards backed by other platform devices, and registered directly via snd_soc_register_card(). Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-core.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9dfbb8f..3feddd9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1881,7 +1881,6 @@ static int soc_probe(struct platform_device *pdev) /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; - snd_soc_initialize_card_lists(card); ret = snd_soc_register_card(card); if (ret != 0) { @@ -3122,6 +3121,8 @@ int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; + snd_soc_initialize_card_lists(card); + soc_init_card_debugfs(card); card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) * -- 1.7.1

3 9

[alsa-devel] [PATCH 0/3] ASoC: patch series to add support for capture in mid-x86 and sn95031 codec
by Harsha, Priya 31 Jan '11

31 Jan '11

Hi Mark, This patch set adds support for capture in mid-x6 platform, machine and TI sn95031 codec drivers. [PATCH 1/3] ASoC: sn95031: add capture support This patch adds support for capture codec dais and widgets [PATCH 2/3] ASoC: sst_platform: add support for capture stream on headset dai This patch adds mid-x86 sst platform driver support for capture on headset dai [PATCH 3/3] ASoC: mid-x86: Add support for capture in machine driver This patch configures the capture unused pins -Harsha

3 3

[alsa-devel] need input on USB Audio class device.
by Sakthi Subramanian 31 Jan '11

31 Jan '11

Hi All, i am trying to develop a ALSA record application for USB - Audio device. This USB Audio device has stereo recording and 4 inputs (4 mics). I am trying to capture PCM audio from the 4 mics using the simple record application. Need inputs on the below questions, thanks in advance for the inputs. 1. I have used the ALSA API to set the number of channels as 8. snd_pcm_hw_params_set_channels (capture_handle, hw_params, 8); I have started capturing the data with the above change, and opened the same using Audacity, it says 16 bit PCM Mono channel Little Endian. in my opinion, that must show as 8 channel. Am i calling the right API with parameters? or Please let me know how to get the multi channel PCM capture data from ALSA? This is the output of arecord -l: # arecord -l **** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Audio Device], device 0: USB Audio [USB Audio] Subdevices: 0/1 Subdevice #0: subdevice #0 Once again thanks a lot for ur inputs.. Regards Sakthi SG

2 1

[alsa-devel] [PATCH] MFD: DaVinci: Fix voice codec device name
by Manjunathappa, Prakash 31 Jan '11

31 Jan '11

Fix the device name in DaVinci Voice Codec MFD driver to load davinci-vcif and cq93vc codec client drivers. Signed-off-by: Manjunathappa, Prakash <prakash.pm(a)ti.com> --- This patch applies to Linus's tree at [1]. [1] http://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6.git --- drivers/mfd/davinci_voicecodec.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/mfd/davinci_voicecodec.c b/drivers/mfd/davinci_voicecodec.c index 33c923d..fdd8a1b 100644 --- a/drivers/mfd/davinci_voicecodec.c +++ b/drivers/mfd/davinci_voicecodec.c @@ -118,12 +118,12 @@ static int __init davinci_vc_probe(struct platform_device *pdev) /* Voice codec interface client */ cell = &davinci_vc->cells[DAVINCI_VC_VCIF_CELL]; - cell->name = "davinci_vcif"; + cell->name = "davinci-vcif"; cell->driver_data = davinci_vc; /* Voice codec CQ93VC client */ cell = &davinci_vc->cells[DAVINCI_VC_CQ93VC_CELL]; - cell->name = "cq93vc"; + cell->name = "cq93vc-codec"; cell->driver_data = davinci_vc; ret = mfd_add_devices(&pdev->dev, pdev->id, davinci_vc->cells, -- 1.7.1

3 2