January 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: add driver for max9768 amplifier
by Wolfram Sang 19 Jan '12

19 Jan '12

Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> --- include/sound/max9768.h | 23 ++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9768.c | 244 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 273 insertions(+), 0 deletions(-) create mode 100644 include/sound/max9768.h create mode 100644 sound/soc/codecs/max9768.c diff --git a/include/sound/max9768.h b/include/sound/max9768.h new file mode 100644 index 0000000..f8f1639 --- /dev/null +++ b/include/sound/max9768.h @@ -0,0 +1,23 @@ +/* + * Platform data for MAX9768 + * Copyright (C) 2011, 2012 by Wolfram Sang, Pengutronix e.K. + * same licence as the driver + */ + +#ifndef __SOUND_MAX9768_PDATA_H__ +#define __SOUND_MAX9768_PDATA_H__ + +/** + * struct max9768_pdata - optional platform specific MAX9768 configuration + * @shdn_gpio: GPIO to SHDN pin. If not valid, pin must be hardwired HIGH + * @mute_gpio: GPIO to MUTE pin. If not valid, control for mute won't be added + * @flags: configuration flags, e.g. set classic PWM mode. + */ +struct max9768_pdata { + int shdn_gpio; + int mute_gpio; + unsigned flags; +#define MAX9768_FLAG_CLASSIC_PWM (1 << 0) +}; + +#endif /* __SOUND_MAX9768_PDATA_H__*/ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7c205e7..a861014 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DFBMCS320 select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C + select SND_SOC_MAX9768 if I2C select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C @@ -425,6 +426,9 @@ config SND_SOC_WM9713 config SND_SOC_LM4857 tristate +config SND_SOC_MAX9768 + tristate + config SND_SOC_MAX9877 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index de80781..cf3555b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -25,6 +25,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o +snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o @@ -129,6 +130,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o +obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c new file mode 100644 index 0000000..bd57b45 --- /dev/null +++ b/sound/soc/codecs/max9768.c @@ -0,0 +1,244 @@ +/* + * MAX9768 AMP driver + * + * Copyright (C) 2011, 2012 by Wolfram Sang, Pengutronix e.K. + * based on lm4857.c by Lars-Peter Clausen et al. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; version 2 of the License. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/max9768.h> + +/* The register offsets in the cache array */ +#define MAX9768_VOL 0 +#define MAX9768_CTRL 3 + +/* Commands */ +#define MAX9768_CTRL_PWM 0x15 +#define MAX9768_CTRL_FILTERLESS 0x16 + +struct max9768 { + struct i2c_client *client; + int mute_gpio; + int shdn_gpio; + u32 flags; +}; + +static const uint8_t max9768_default_regs[] = { + 0x00, 0x00, 0x00, MAX9768_CTRL_FILTERLESS, +}; + +static int max9768_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + uint8_t data; + int ret; + + ret = snd_soc_cache_write(codec, reg, value); + if (ret < 0) + return ret; + + data = (reg << 6) | value; + ret = i2c_smbus_write_byte(codec->control_data, data); + if (ret) + dev_err(codec->dev, "Failed to write register: %d\n", ret); + + return ret; +} + +static unsigned int max9768_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + unsigned int val; + int ret; + + ret = snd_soc_cache_read(codec, reg, &val); + + return ret ?: val; +} + + +static int max9768_get_gpio(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); + int val = gpio_get_value_cansleep(max9768->mute_gpio); + + ucontrol->value.integer.value[0] = !val; + + return 0; +} + +static int max9768_set_gpio(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); + + gpio_set_value_cansleep(max9768->mute_gpio, !ucontrol->value.integer.value[0]); + + return 0; +} + +static const unsigned int volume_tlv[] = { + TLV_DB_RANGE_HEAD(43), + 0, 0, TLV_DB_SCALE_ITEM(-16150, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-9280, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-9030, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(-8680, 0, 0), + 4, 4, TLV_DB_SCALE_ITEM(-8430, 0, 0), + 5, 5, TLV_DB_SCALE_ITEM(-8080, 0, 0), + 6, 6, TLV_DB_SCALE_ITEM(-7830, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(-7470, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(-7220, 0, 0), + 9, 9, TLV_DB_SCALE_ITEM(-6870, 0, 0), + 10, 10, TLV_DB_SCALE_ITEM(-6620, 0, 0), + 11, 11, TLV_DB_SCALE_ITEM(-6270, 0, 0), + 12, 12, TLV_DB_SCALE_ITEM(-6020, 0, 0), + 13, 13, TLV_DB_SCALE_ITEM(-5670, 0, 0), + 14, 14, TLV_DB_SCALE_ITEM(-5420, 0, 0), + 15, 17, TLV_DB_SCALE_ITEM(-5060, 250, 0), + 18, 18, TLV_DB_SCALE_ITEM(-4370, 0, 0), + 19, 19, TLV_DB_SCALE_ITEM(-4210, 0, 0), + 20, 20, TLV_DB_SCALE_ITEM(-3960, 0, 0), + 21, 21, TLV_DB_SCALE_ITEM(-3760, 0, 0), + 22, 22, TLV_DB_SCALE_ITEM(-3600, 0, 0), + 23, 23, TLV_DB_SCALE_ITEM(-3340, 0, 0), + 24, 24, TLV_DB_SCALE_ITEM(-3150, 0, 0), + 25, 25, TLV_DB_SCALE_ITEM(-2980, 0, 0), + 26, 26, TLV_DB_SCALE_ITEM(-2720, 0, 0), + 27, 27, TLV_DB_SCALE_ITEM(-2520, 0, 0), + 28, 30, TLV_DB_SCALE_ITEM(-2350, 190, 0), + 31, 31, TLV_DB_SCALE_ITEM(-1750, 0, 0), + 32, 34, TLV_DB_SCALE_ITEM(-1640, 100, 0), + 35, 37, TLV_DB_SCALE_ITEM(-1310, 110, 0), + 38, 39, TLV_DB_SCALE_ITEM(-990, 100, 0), + 40, 40, TLV_DB_SCALE_ITEM(-710, 0, 0), + 41, 41, TLV_DB_SCALE_ITEM(-600, 0, 0), + 42, 42, TLV_DB_SCALE_ITEM(-500, 0, 0), + 43, 43, TLV_DB_SCALE_ITEM(-340, 0, 0), + 44, 44, TLV_DB_SCALE_ITEM(-190, 0, 0), + 45, 45, TLV_DB_SCALE_ITEM(-50, 0, 0), + 46, 46, TLV_DB_SCALE_ITEM(50, 0, 0), + 47, 50, TLV_DB_SCALE_ITEM(120, 40, 0), + 51, 57, TLV_DB_SCALE_ITEM(290, 50, 0), + 58, 58, TLV_DB_SCALE_ITEM(650, 0, 0), + 59, 62, TLV_DB_SCALE_ITEM(700, 60, 0), + 63, 63, TLV_DB_SCALE_ITEM(950, 0, 0), +}; + +static const struct snd_kcontrol_new max9768_volume[] = { + SOC_SINGLE_TLV("Playback Volume", MAX9768_VOL, 0, 63, 0, volume_tlv), +}; + +static const struct snd_kcontrol_new max9768_mute[] = { + SOC_SINGLE_BOOL_EXT("Mute Switch", 0, max9768_get_gpio, max9768_set_gpio), +}; + +static int max9768_probe(struct snd_soc_codec *codec) +{ + struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = max9768->client; + + if (max9768->flags & MAX9768_FLAG_CLASSIC_PWM) { + ret = snd_soc_write(codec, MAX9768_CTRL, MAX9768_CTRL_PWM); + if (ret) + return ret; + } + + ret = snd_soc_add_controls(codec, max9768_volume, + ARRAY_SIZE(max9768_volume)); + if (ret) + return ret; + + if (max9768->mute_gpio >= 0) { + ret = snd_soc_add_controls(codec, max9768_mute, + ARRAY_SIZE(max9768_mute)); + if (ret) + return ret; + } + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max9768 = { + .write = max9768_write, + .read = max9768_read, + .probe = max9768_probe, + .reg_cache_size = ARRAY_SIZE(max9768_default_regs), + .reg_word_size = sizeof(uint8_t), + .reg_cache_default = max9768_default_regs, +}; + +static int __devinit max9768_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct max9768 *max9768; + struct max9768_pdata *pdata = client->dev.platform_data; + int err; + + max9768 = devm_kzalloc(&client->dev, sizeof(*max9768), GFP_KERNEL); + if (!max9768) + return -ENOMEM; + + if (pdata) { + /* Mute on powerup to avoid clicks */ + err = gpio_request_one(pdata->mute_gpio, GPIOF_INIT_HIGH, "MAX9768 Mute"); + max9768->mute_gpio = err ?: pdata->mute_gpio; + + /* Activate chip, enabling I2C */ + err = gpio_request_one(pdata->shdn_gpio, GPIOF_INIT_HIGH, "MAX9768 Shutdown"); + max9768->shdn_gpio = err ?: pdata->shdn_gpio; + + max9768->flags = pdata->flags; + } else { + max9768->shdn_gpio = -EINVAL; + max9768->mute_gpio = -EINVAL; + } + + max9768->client = client; + i2c_set_clientdata(client, max9768); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_max9768, NULL, 0); +} + +static int __devexit max9768_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max9768_i2c_id[] = { + { "max9768", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9768_i2c_id); + +static struct i2c_driver max9768_i2c_driver = { + .driver = { + .name = "max9768", + .owner = THIS_MODULE, + }, + .probe = max9768_i2c_probe, + .remove = __devexit_p(max9768_i2c_remove), + .id_table = max9768_i2c_id, +}; +module_i2c_driver(max9768_i2c_driver); + +MODULE_AUTHOR("Wolfram Sang <w.sang(a)pengutronix.de>"); +MODULE_DESCRIPTION("ASoC MAX9768 amplifier driver"); +MODULE_LICENSE("GPL v2"); -- 1.7.8.3

3 6

Re: [alsa-devel] [PATCH 00/10] patch set about the MXS-DMA
by Shawn Guo 19 Jan '12

19 Jan '12

On Thu, Jan 19, 2012 at 02:15:57PM +0800, Huang Shijie wrote: > This patch set does two things about the MXS-DMA: > [1] patch 1 ~ patch 5: move the mxs dma header to a more common place. > Beside the mx23/mx28, the APBH-DMA is also used by MX50 and MX6Q. > So move the mxs dma header file to a more common place, and rename > it to mxs-dma.h > For the sake of bitsec, these 5 patches need to be one. > [2] patch 6 ~ patch 10: rewrite the last parameter of mxs_dma_prep_slave_sg(). For the sake of bitsec, at least patches #7 and #8 need to be one patch. That said, if I apply the series and then check out the commit at patch #7, you need to all mxs-dma client drivers, mxs-mmc, gpmi do not break. > In the new GPMI version, some hardware behavior changes. > The WAIT4END bit should be set in the middle one of the ECC READ page DMA chain. > The DMA chain should be set like this: > > +-----+ +-----+ +-----+ > | cmd | ------------> | cmd | ------------------> | cmd | > +-----+ +-----+ +-----+ > ^ ^ > | | > | | > set WAIT4END here too set WAIT4END here > > Please read the comment in patch "MXS-DMA : add more flags for MXS-DMA". > > Huang Shijie (10): > MXS-DMA : move the mxs-dma.h to a more common place > MXS-DMA : change the header > MXS-MMC : change the DMA header file > MTD/GPMI : change the DMA header file > ASoc : change the DMA header file > MXS-DMA : add more flags for MXS-DMA > MXS-DMA : change the last parameter of mxs_dma_prep_slave_sg() > MXS-MMC : use the new DMA flags > MTD/GPMI : add a new field `gpmi_version` > MTD/GPMI : change the code for new DMA interface > Please try to make these subject prefix consistent with the existing ones in the tree. You can get them from 'git log' output. For example "mxs-dma: ..." is better than "MXS-DMA : ...". -- Regards, Shawn

2 1

Re: [alsa-devel] [PATCH 06/10] MXS-DMA : add more flags for MXS-DMA
by Russell King - ARM Linux 19 Jan '12

19 Jan '12

On Thu, Jan 19, 2012 at 02:16:03PM +0800, Huang Shijie wrote: > [1] Background : > The GPMI does ECC read page operation with a DMA chain consist of three DMA > Command Structures. The middle one of the chain is used to enable the BCH, > and read out the NAND page. > > The WAIT4END(wait for command end) is a comunication signal between > the GPMI and MXS-DMA. > > [2] The current DMA code sets the WAIT4END bit at the last one, such as: > > +-----+ +-----+ +-----+ > | cmd | ------------> | cmd | ------------------> | cmd | > +-----+ +-----+ +-----+ > ^ > | > | > set WAIT4END here > > This chain works fine in the mx23/mx28. > > [3] But in the new GPMI version (used in MX50/MX60), the WAIT4END bit should > be set not only at the last DMA Command Structure, > but also at the middle one, such as: > > +-----+ +-----+ +-----+ > | cmd | ------------> | cmd | ------------------> | cmd | > +-----+ +-----+ +-----+ > ^ ^ > | | > | | > set WAIT4END here too set WAIT4END here > > If we do not set WAIT4END, the BCH maybe stall in "ECC reading page" state. > In the next ECC write page operation, a DMA-timeout occurs. > This has been catched in the MX6Q board. > > In order to fix the bug, we should let the driver to > set the proper DMA flags in the DMA command structrues. > > So add the new flags for MXS-DMA. > The driver can use these flags to control the DMA in a more flexible way. > > Signed-off-by: Huang Shijie <b32955(a)freescale.com> > --- > include/linux/mxs-dma.h | 26 ++++++++++++++++++++++++++ > 1 files changed, 26 insertions(+), 0 deletions(-) > > diff --git a/include/linux/mxs-dma.h b/include/linux/mxs-dma.h > index 203d7c4..3ef73b8 100644 > --- a/include/linux/mxs-dma.h > +++ b/include/linux/mxs-dma.h > @@ -11,6 +11,32 @@ > > #include <linux/dmaengine.h> > > +/* > + * The drivers use these flags for ->device_prep_slave_sg() : > + * [1] If there is only one DMA command in the DMA chain, the code should be: > + * ...... > + * ->device_prep_slave_sg(MXS_DMA_F_WAIT4END); > + * ...... > + * [2] If there are two DMA commands in the DMA chain, the code should be > + * ...... > + * ->device_prep_slave_sg(0); > + * ...... > + * ->device_prep_slave_sg(MXS_DMA_F_LASTONE); > + * ...... > + * [3] If there are more than two DMA commands in the DMA chain, the code > + * should be: > + * ...... > + * ->device_prep_slave_sg(0); // First > + * ...... > + * ->device_prep_slave_sg(MXS_DMA_F_APPEND [| MXS_DMA_F_WAIT4END]); > + * ...... > + * ->device_prep_slave_sg(MXS_DMA_F_LASTONE); // Last > + */ > +#define MXS_DMA_F_APPEND (1 << 0) > +#define MXS_DMA_F_WAIT4END (1 << 1) > + > +#define MXS_DMA_F_LASTONE (MXS_DMA_F_APPEND | MXS_DMA_F_WAIT4END) Err, the 'flags' argument to device_prep_slave_sg() is supposed to be from the set of enum dma_ctrl_flags. What this means is that your MXS_DMA_F_APPEND is equivalent to DMA_PREP_INTERRUPT, and MXS_DMA_F_WAIT4END is equivalent to DMA_CTRL_ACK. What this does is make your drivers completely dependent on your DMA engine implementation. That's not a good idea when devices get reused in different SoCs. If you need to supply extra flags which aren't in the dma_ctrl_flags, at least make sure that they're using different bits. For bonus points, also have your driver _check_ the DMA engine it's connected to before it passes these flags.

1 0

[alsa-devel] [PATCH] ASoC: Convert blackfin directory to module_platform_driver
by Axel Lin 19 Jan '12

19 Jan '12

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 12 +----------- sound/soc/blackfin/bf5xx-ac97.c | 13 +------------ sound/soc/blackfin/bf5xx-i2s-pcm.c | 12 +----------- sound/soc/blackfin/bf5xx-i2s.c | 13 +------------ sound/soc/blackfin/bf5xx-tdm-pcm.c | 12 +----------- sound/soc/blackfin/bf5xx-tdm.c | 12 +----------- sound/soc/blackfin/bfin-eval-adau1373.c | 12 +----------- sound/soc/blackfin/bfin-eval-adau1701.c | 12 +----------- sound/soc/blackfin/bfin-eval-adav80x.c | 12 +----------- 9 files changed, 9 insertions(+), 101 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 56815c1..fcff583 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -475,17 +475,7 @@ static struct platform_driver bf5xx_pcm_driver = { .remove = __devexit_p(bf5xx_soc_platform_remove), }; -static int __init snd_bf5xx_pcm_init(void) -{ - return platform_driver_register(&bf5xx_pcm_driver); -} -module_init(snd_bf5xx_pcm_init); - -static void __exit snd_bf5xx_pcm_exit(void) -{ - platform_driver_unregister(&bf5xx_pcm_driver); -} -module_exit(snd_bf5xx_pcm_exit); +module_platform_driver(bf5xx_pcm_driver); MODULE_AUTHOR("Cliff Cai"); MODULE_DESCRIPTION("ADI Blackfin AC97 PCM DMA module"); diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 6d21625..f4e9dc4 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -375,18 +375,7 @@ static struct platform_driver asoc_bfin_ac97_driver = { .remove = __devexit_p(asoc_bfin_ac97_remove), }; -static int __init bfin_ac97_init(void) -{ - return platform_driver_register(&asoc_bfin_ac97_driver); -} -module_init(bfin_ac97_init); - -static void __exit bfin_ac97_exit(void) -{ - platform_driver_unregister(&asoc_bfin_ac97_driver); -} -module_exit(bfin_ac97_exit); - +module_platform_driver(asoc_bfin_ac97_driver); MODULE_AUTHOR("Roy Huang"); MODULE_DESCRIPTION("AC97 driver for ADI Blackfin"); diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 7565e15..6ec3d41 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -314,17 +314,7 @@ static struct platform_driver bfin_i2s_pcm_driver = { .remove = __devexit_p(bfin_i2s_soc_platform_remove), }; -static int __init snd_bfin_i2s_pcm_init(void) -{ - return platform_driver_register(&bfin_i2s_pcm_driver); -} -module_init(snd_bfin_i2s_pcm_init); - -static void __exit snd_bfin_i2s_pcm_exit(void) -{ - platform_driver_unregister(&bfin_i2s_pcm_driver); -} -module_exit(snd_bfin_i2s_pcm_exit); +module_platform_driver(bfin_i2s_pcm_driver); MODULE_AUTHOR("Cliff Cai"); MODULE_DESCRIPTION("ADI Blackfin I2S PCM DMA module"); diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 00cc3e0..a400633 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -288,18 +288,7 @@ static struct platform_driver bfin_i2s_driver = { }, }; -static int __init bfin_i2s_init(void) -{ - return platform_driver_register(&bfin_i2s_driver); -} - -static void __exit bfin_i2s_exit(void) -{ - platform_driver_unregister(&bfin_i2s_driver); -} - -module_init(bfin_i2s_init); -module_exit(bfin_i2s_exit); +module_platform_driver(bfin_i2s_driver); /* Module information */ MODULE_AUTHOR("Cliff Cai"); diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index c95cc03..4406f9a 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -339,17 +339,7 @@ static struct platform_driver bfin_tdm_driver = { .remove = __devexit_p(bf5xx_soc_platform_remove), }; -static int __init snd_bfin_tdm_init(void) -{ - return platform_driver_register(&bfin_tdm_driver); -} -module_init(snd_bfin_tdm_init); - -static void __exit snd_bfin_tdm_exit(void) -{ - platform_driver_unregister(&bfin_tdm_driver); -} -module_exit(snd_bfin_tdm_exit); +module_platform_driver(bfin_tdm_driver); MODULE_AUTHOR("Barry Song"); MODULE_DESCRIPTION("ADI Blackfin TDM PCM DMA module"); diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index a822d1e..bf65e0e 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -314,17 +314,7 @@ static struct platform_driver bfin_tdm_driver = { }, }; -static int __init bfin_tdm_init(void) -{ - return platform_driver_register(&bfin_tdm_driver); -} -module_init(bfin_tdm_init); - -static void __exit bfin_tdm_exit(void) -{ - platform_driver_unregister(&bfin_tdm_driver); -} -module_exit(bfin_tdm_exit); +module_platform_driver(bfin_tdm_driver); /* Module information */ MODULE_AUTHOR("Barry Song"); diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c index 8df2a3b..85ed39a 100644 --- a/sound/soc/blackfin/bfin-eval-adau1373.c +++ b/sound/soc/blackfin/bfin-eval-adau1373.c @@ -184,17 +184,7 @@ static struct platform_driver bfin_eval_adau1373_driver = { .remove = __devexit_p(bfin_eval_adau1373_remove), }; -static int __init bfin_eval_adau1373_init(void) -{ - return platform_driver_register(&bfin_eval_adau1373_driver); -} -module_init(bfin_eval_adau1373_init); - -static void __exit bfin_eval_adau1373_exit(void) -{ - platform_driver_unregister(&bfin_eval_adau1373_driver); -} -module_exit(bfin_eval_adau1373_exit); +module_platform_driver(bfin_eval_adau1373_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("ALSA SoC bfin adau1373 driver"); diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c index e5550ac..1a88fe9 100644 --- a/sound/soc/blackfin/bfin-eval-adau1701.c +++ b/sound/soc/blackfin/bfin-eval-adau1701.c @@ -121,17 +121,7 @@ static struct platform_driver bfin_eval_adau1701_driver = { .remove = __devexit_p(bfin_eval_adau1701_remove), }; -static int __init bfin_eval_adau1701_init(void) -{ - return platform_driver_register(&bfin_eval_adau1701_driver); -} -module_init(bfin_eval_adau1701_init); - -static void __exit bfin_eval_adau1701_exit(void) -{ - platform_driver_unregister(&bfin_eval_adau1701_driver); -} -module_exit(bfin_eval_adau1701_exit); +module_platform_driver(bfin_eval_adau1701_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("ALSA SoC bfin ADAU1701 driver"); diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c index 897cfa6..0bc995f 100644 --- a/sound/soc/blackfin/bfin-eval-adav80x.c +++ b/sound/soc/blackfin/bfin-eval-adav80x.c @@ -157,17 +157,7 @@ static struct platform_driver bfin_eval_adav80x_driver = { .id_table = bfin_eval_adav80x_ids, }; -static int __init bfin_eval_adav80x_init(void) -{ - return platform_driver_register(&bfin_eval_adav80x_driver); -} -module_init(bfin_eval_adav80x_init); - -static void __exit bfin_eval_adav80x_exit(void) -{ - platform_driver_unregister(&bfin_eval_adav80x_driver); -} -module_exit(bfin_eval_adav80x_exit); +module_platform_driver(bfin_eval_adav80x_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("ALSA SoC bfin adav80x driver"); -- 1.7.5.4

3 2

[alsa-devel] [PATCH 1/2] ASoC: Make WM5100 interrupt path use regmap directly
by Mark Brown 18 Jan '12

18 Jan '12

This will allow us to move the interrupt allocation out of the ASoC part of the driver and simplifies the locking by removing any reliance in the bulk of the interrupt path on the big CODEC lock. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 187 ++++++++++++++++++++++++++------------------- 1 files changed, 107 insertions(+), 80 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index b22b425..02a759d 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -51,6 +51,7 @@ struct wm5100_fll { /* codec private data */ struct wm5100_priv { + struct device *dev; struct regmap *regmap; struct snd_soc_codec *codec; @@ -891,48 +892,48 @@ static int wm5100_dbvdd_ev(struct snd_soc_dapm_widget *w, } } -static void wm5100_log_status3(struct snd_soc_codec *codec, int val) +static void wm5100_log_status3(struct wm5100_priv *wm5100, int val) { if (val & WM5100_SPK_SHUTDOWN_WARN_EINT) - dev_crit(codec->dev, "Speaker shutdown warning\n"); + dev_crit(wm5100->dev, "Speaker shutdown warning\n"); if (val & WM5100_SPK_SHUTDOWN_EINT) - dev_crit(codec->dev, "Speaker shutdown\n"); + dev_crit(wm5100->dev, "Speaker shutdown\n"); if (val & WM5100_CLKGEN_ERR_EINT) - dev_crit(codec->dev, "SYSCLK underclocked\n"); + dev_crit(wm5100->dev, "SYSCLK underclocked\n"); if (val & WM5100_CLKGEN_ERR_ASYNC_EINT) - dev_crit(codec->dev, "ASYNCCLK underclocked\n"); + dev_crit(wm5100->dev, "ASYNCCLK underclocked\n"); } -static void wm5100_log_status4(struct snd_soc_codec *codec, int val) +static void wm5100_log_status4(struct wm5100_priv *wm5100, int val) { if (val & WM5100_AIF3_ERR_EINT) - dev_err(codec->dev, "AIF3 configuration error\n"); + dev_err(wm5100->dev, "AIF3 configuration error\n"); if (val & WM5100_AIF2_ERR_EINT) - dev_err(codec->dev, "AIF2 configuration error\n"); + dev_err(wm5100->dev, "AIF2 configuration error\n"); if (val & WM5100_AIF1_ERR_EINT) - dev_err(codec->dev, "AIF1 configuration error\n"); + dev_err(wm5100->dev, "AIF1 configuration error\n"); if (val & WM5100_CTRLIF_ERR_EINT) - dev_err(codec->dev, "Control interface error\n"); + dev_err(wm5100->dev, "Control interface error\n"); if (val & WM5100_ISRC2_UNDERCLOCKED_EINT) - dev_err(codec->dev, "ISRC2 underclocked\n"); + dev_err(wm5100->dev, "ISRC2 underclocked\n"); if (val & WM5100_ISRC1_UNDERCLOCKED_EINT) - dev_err(codec->dev, "ISRC1 underclocked\n"); + dev_err(wm5100->dev, "ISRC1 underclocked\n"); if (val & WM5100_FX_UNDERCLOCKED_EINT) - dev_err(codec->dev, "FX underclocked\n"); + dev_err(wm5100->dev, "FX underclocked\n"); if (val & WM5100_AIF3_UNDERCLOCKED_EINT) - dev_err(codec->dev, "AIF3 underclocked\n"); + dev_err(wm5100->dev, "AIF3 underclocked\n"); if (val & WM5100_AIF2_UNDERCLOCKED_EINT) - dev_err(codec->dev, "AIF2 underclocked\n"); + dev_err(wm5100->dev, "AIF2 underclocked\n"); if (val & WM5100_AIF1_UNDERCLOCKED_EINT) - dev_err(codec->dev, "AIF1 underclocked\n"); + dev_err(wm5100->dev, "AIF1 underclocked\n"); if (val & WM5100_ASRC_UNDERCLOCKED_EINT) - dev_err(codec->dev, "ASRC underclocked\n"); + dev_err(wm5100->dev, "ASRC underclocked\n"); if (val & WM5100_DAC_UNDERCLOCKED_EINT) - dev_err(codec->dev, "DAC underclocked\n"); + dev_err(wm5100->dev, "DAC underclocked\n"); if (val & WM5100_ADC_UNDERCLOCKED_EINT) - dev_err(codec->dev, "ADC underclocked\n"); + dev_err(wm5100->dev, "ADC underclocked\n"); if (val & WM5100_MIXER_UNDERCLOCKED_EINT) - dev_err(codec->dev, "Mixer underclocked\n"); + dev_err(wm5100->dev, "Mixer underclocked\n"); } static int wm5100_post_ev(struct snd_soc_dapm_widget *w, @@ -940,16 +941,17 @@ static int wm5100_post_ev(struct snd_soc_dapm_widget *w, int event) { struct snd_soc_codec *codec = w->codec; + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); int ret; ret = snd_soc_read(codec, WM5100_INTERRUPT_RAW_STATUS_3); ret &= WM5100_SPK_SHUTDOWN_WARN_STS | WM5100_SPK_SHUTDOWN_STS | WM5100_CLKGEN_ERR_STS | WM5100_CLKGEN_ERR_ASYNC_STS; - wm5100_log_status3(codec, ret); + wm5100_log_status3(wm5100, ret); ret = snd_soc_read(codec, WM5100_INTERRUPT_RAW_STATUS_4); - wm5100_log_status4(codec, ret); + wm5100_log_status4(wm5100, ret); return 0; } @@ -2200,55 +2202,59 @@ static int wm5100_dig_vu[] = { WM5100_DAC_DIGITAL_VOLUME_6R, }; -static void wm5100_set_detect_mode(struct snd_soc_codec *codec, int the_mode) +static void wm5100_set_detect_mode(struct wm5100_priv *wm5100, int the_mode) { - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); struct wm5100_jack_mode *mode = &wm5100->pdata.jack_modes[the_mode]; BUG_ON(the_mode >= ARRAY_SIZE(wm5100->pdata.jack_modes)); gpio_set_value_cansleep(wm5100->pdata.hp_pol, mode->hp_pol); - snd_soc_update_bits(codec, WM5100_ACCESSORY_DETECT_MODE_1, - WM5100_ACCDET_BIAS_SRC_MASK | - WM5100_ACCDET_SRC, - (mode->bias << WM5100_ACCDET_BIAS_SRC_SHIFT) | - mode->micd_src << WM5100_ACCDET_SRC_SHIFT); - snd_soc_update_bits(codec, WM5100_MISC_CONTROL, - WM5100_HPCOM_SRC, - mode->micd_src << WM5100_HPCOM_SRC_SHIFT); + regmap_update_bits(wm5100->regmap, WM5100_ACCESSORY_DETECT_MODE_1, + WM5100_ACCDET_BIAS_SRC_MASK | + WM5100_ACCDET_SRC, + (mode->bias << WM5100_ACCDET_BIAS_SRC_SHIFT) | + mode->micd_src << WM5100_ACCDET_SRC_SHIFT); + regmap_update_bits(wm5100->regmap, WM5100_MISC_CONTROL, + WM5100_HPCOM_SRC, + mode->micd_src << WM5100_HPCOM_SRC_SHIFT); wm5100->jack_mode = the_mode; - dev_dbg(codec->dev, "Set microphone polarity to %d\n", + dev_dbg(wm5100->dev, "Set microphone polarity to %d\n", wm5100->jack_mode); } -static void wm5100_micd_irq(struct snd_soc_codec *codec) +static void wm5100_micd_irq(struct wm5100_priv *wm5100) { - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); - int val; + unsigned int val; + int ret; - val = snd_soc_read(codec, WM5100_MIC_DETECT_3); + ret = regmap_read(wm5100->regmap, WM5100_MIC_DETECT_3, &val); + if (ret != 0) { + dev_err(wm5100->dev, "Failed to read micropone status: %d\n", + ret); + return; + } - dev_dbg(codec->dev, "Microphone event: %x\n", val); + dev_dbg(wm5100->dev, "Microphone event: %x\n", val); if (!(val & WM5100_ACCDET_VALID)) { - dev_warn(codec->dev, "Microphone detection state invalid\n"); + dev_warn(wm5100->dev, "Microphone detection state invalid\n"); return; } /* No accessory, reset everything and report removal */ if (!(val & WM5100_ACCDET_STS)) { - dev_dbg(codec->dev, "Jack removal detected\n"); + dev_dbg(wm5100->dev, "Jack removal detected\n"); wm5100->jack_mic = false; wm5100->jack_detecting = true; snd_soc_jack_report(wm5100->jack, 0, SND_JACK_LINEOUT | SND_JACK_HEADSET | SND_JACK_BTN_0); - snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, - WM5100_ACCDET_RATE_MASK, - WM5100_ACCDET_RATE_MASK); + regmap_update_bits(wm5100->regmap, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + WM5100_ACCDET_RATE_MASK); return; } @@ -2258,7 +2264,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) */ if (val & 0x400) { if (wm5100->jack_detecting) { - dev_dbg(codec->dev, "Microphone detected\n"); + dev_dbg(wm5100->dev, "Microphone detected\n"); wm5100->jack_mic = true; snd_soc_jack_report(wm5100->jack, SND_JACK_HEADSET, @@ -2266,11 +2272,11 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) /* Increase poll rate to give better responsiveness * for buttons */ - snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, - WM5100_ACCDET_RATE_MASK, - 5 << WM5100_ACCDET_RATE_SHIFT); + regmap_update_bits(wm5100->regmap, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + 5 << WM5100_ACCDET_RATE_SHIFT); } else { - dev_dbg(codec->dev, "Mic button up\n"); + dev_dbg(wm5100->dev, "Mic button up\n"); snd_soc_jack_report(wm5100->jack, 0, SND_JACK_BTN_0); } @@ -2283,7 +2289,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) * plain headphones. */ if (wm5100->jack_detecting && (val & 0x3f8)) { - wm5100_set_detect_mode(codec, !wm5100->jack_mode); + wm5100_set_detect_mode(wm5100, !wm5100->jack_mode); return; } @@ -2293,20 +2299,20 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) */ if (val & 0x3fc) { if (wm5100->jack_mic) { - dev_dbg(codec->dev, "Mic button detected\n"); + dev_dbg(wm5100->dev, "Mic button detected\n"); snd_soc_jack_report(wm5100->jack, SND_JACK_BTN_0, SND_JACK_BTN_0); } else if (wm5100->jack_detecting) { - dev_dbg(codec->dev, "Headphone detected\n"); + dev_dbg(wm5100->dev, "Headphone detected\n"); snd_soc_jack_report(wm5100->jack, SND_JACK_HEADPHONE, SND_JACK_HEADPHONE); /* Increase the detection rate a bit for * responsiveness. */ - snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, - WM5100_ACCDET_RATE_MASK, - 7 << WM5100_ACCDET_RATE_SHIFT); + regmap_update_bits(wm5100->regmap, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + 7 << WM5100_ACCDET_RATE_SHIFT); } } } @@ -2319,7 +2325,7 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) wm5100->jack = jack; wm5100->jack_detecting = true; - wm5100_set_detect_mode(codec, 0); + wm5100_set_detect_mode(wm5100, 0); /* Slowest detection rate, gives debounce for initial * detection */ @@ -2358,52 +2364,70 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) static irqreturn_t wm5100_irq(int irq, void *data) { - struct snd_soc_codec *codec = data; - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct wm5100_priv *wm5100 = data; irqreturn_t status = IRQ_NONE; - int irq_val; + unsigned int irq_val, mask_val; + int ret; - irq_val = snd_soc_read(codec, WM5100_INTERRUPT_STATUS_3); - if (irq_val < 0) { - dev_err(codec->dev, "Failed to read IRQ status 3: %d\n", - irq_val); + ret = regmap_read(wm5100->regmap, WM5100_INTERRUPT_STATUS_3, &irq_val); + if (ret < 0) { + dev_err(wm5100->dev, "Failed to read IRQ status 3: %d\n", + ret); irq_val = 0; } - irq_val &= ~snd_soc_read(codec, WM5100_INTERRUPT_STATUS_3_MASK); - snd_soc_write(codec, WM5100_INTERRUPT_STATUS_3, irq_val); + ret = regmap_read(wm5100->regmap, WM5100_INTERRUPT_STATUS_3_MASK, + &mask_val); + if (ret < 0) { + dev_err(wm5100->dev, "Failed to read IRQ mask 3: %d\n", + ret); + mask_val = 0xffff; + } + + irq_val &= ~mask_val; + + regmap_write(wm5100->regmap, WM5100_INTERRUPT_STATUS_3, irq_val); if (irq_val) status = IRQ_HANDLED; - wm5100_log_status3(codec, irq_val); + wm5100_log_status3(wm5100, irq_val); if (irq_val & WM5100_FLL1_LOCK_EINT) { - dev_dbg(codec->dev, "FLL1 locked\n"); + dev_dbg(wm5100->dev, "FLL1 locked\n"); complete(&wm5100->fll[0].lock); } if (irq_val & WM5100_FLL2_LOCK_EINT) { - dev_dbg(codec->dev, "FLL2 locked\n"); + dev_dbg(wm5100->dev, "FLL2 locked\n"); complete(&wm5100->fll[1].lock); } if (irq_val & WM5100_ACCDET_EINT) - wm5100_micd_irq(codec); + wm5100_micd_irq(wm5100); - irq_val = snd_soc_read(codec, WM5100_INTERRUPT_STATUS_4); - if (irq_val < 0) { - dev_err(codec->dev, "Failed to read IRQ status 4: %d\n", - irq_val); + ret = regmap_read(wm5100->regmap, WM5100_INTERRUPT_STATUS_4, &irq_val); + if (ret < 0) { + dev_err(wm5100->dev, "Failed to read IRQ status 4: %d\n", + ret); irq_val = 0; } - irq_val &= ~snd_soc_read(codec, WM5100_INTERRUPT_STATUS_4_MASK); + + ret = regmap_read(wm5100->regmap, WM5100_INTERRUPT_STATUS_4_MASK, + &mask_val); + if (ret < 0) { + dev_err(wm5100->dev, "Failed to read IRQ mask 4: %d\n", + ret); + mask_val = 0xffff; + } + + irq_val &= ~mask_val; if (irq_val) status = IRQ_HANDLED; - snd_soc_write(codec, WM5100_INTERRUPT_STATUS_4, irq_val); + regmap_write(wm5100->regmap, WM5100_INTERRUPT_STATUS_4, irq_val); - wm5100_log_status4(codec, irq_val); + wm5100_log_status4(wm5100, irq_val); return status; } @@ -2612,11 +2636,12 @@ static int wm5100_probe(struct snd_soc_codec *codec) if (irq_flags & (IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING)) ret = request_threaded_irq(i2c->irq, NULL, - wm5100_edge_irq, - irq_flags, "wm5100", codec); + wm5100_edge_irq, irq_flags, + "wm5100", wm5100); else ret = request_threaded_irq(i2c->irq, NULL, wm5100_irq, - irq_flags, "wm5100", codec); + irq_flags, "wm5100", + wm5100); if (ret != 0) { dev_err(codec->dev, "Failed to request IRQ %d: %d\n", @@ -2679,7 +2704,7 @@ static int wm5100_probe(struct snd_soc_codec *codec) err_gpio: if (i2c->irq) - free_irq(i2c->irq, codec); + free_irq(i2c->irq, wm5100); for (i = 0; i < ARRAY_SIZE(wm5100->adsp1); i++) wm_adsp1_exit(wm5100->adsp1[i]); @@ -2698,7 +2723,7 @@ static int wm5100_remove(struct snd_soc_codec *codec) } if (i2c->irq) - free_irq(i2c->irq, codec); + free_irq(i2c->irq, wm5100); for (i = 0; i < ARRAY_SIZE(wm5100->adsp1); i++) wm_adsp1_exit(wm5100->adsp1[i]); @@ -2758,6 +2783,8 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, if (wm5100 == NULL) return -ENOMEM; + wm5100->dev = &i2c->dev; + wm5100->regmap = regmap_init_i2c(i2c, &wm5100_regmap); if (IS_ERR(wm5100->regmap)) { ret = PTR_ERR(wm5100->regmap); -- 1.7.7.3

1 1

[alsa-devel] [PATCH] ASoC: Remove redundant set_bias_level() from WM5100 remove()
by Mark Brown 18 Jan '12

18 Jan '12

The framework should bring the device down before it calls the driver. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index fca7f0a..b22b425 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2693,8 +2693,6 @@ static int wm5100_remove(struct snd_soc_codec *codec) struct i2c_client *i2c = to_i2c_client(codec->dev); int i; - wm5100_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (wm5100->pdata.hp_pol) { gpio_free(wm5100->pdata.hp_pol); } -- 1.7.7.3

1 0

[alsa-devel] [PATCH 1/2] ASoC: tlv320aic32x4: always enable dividers
by Wolfram Sang 18 Jan '12

18 Jan '12

Dividers (such as MDAC) are always needed, independent of the codec being I2S master or slave. Needed on a custom board where the codec has to be slave. Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> --- sound/soc/codecs/tlv320aic32x4.c | 102 +++++++++++++++++--------------------- 1 files changed, 46 insertions(+), 56 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index eb401ef..3806cb6 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -60,7 +60,6 @@ struct aic32x4_rate_divs { struct aic32x4_priv { u32 sysclk; - s32 master; u8 page_no; void *control_data; u32 power_cfg; @@ -369,7 +368,6 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, 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_1; u8 iface_reg_2; u8 iface_reg_3; @@ -384,11 +382,9 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - aic32x4->master = 1; iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER; break; case SND_SOC_DAIFMT_CBS_CFS: - aic32x4->master = 0; break; default: printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); @@ -526,64 +522,58 @@ static int aic32x4_mute(struct snd_soc_dai *dai, int mute) 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); - switch (level) { case SND_SOC_BIAS_ON: - if (aic32x4->master) { - /* Switch on PLL */ - snd_soc_update_bits(codec, AIC32X4_PLLPR, - AIC32X4_PLLEN, AIC32X4_PLLEN); - - /* Switch on NDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_NDAC, - AIC32X4_NDACEN, AIC32X4_NDACEN); - - /* Switch on MDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_MDAC, - AIC32X4_MDACEN, AIC32X4_MDACEN); - - /* Switch on NADC Divider */ - snd_soc_update_bits(codec, AIC32X4_NADC, - AIC32X4_NADCEN, AIC32X4_NADCEN); - - /* Switch on MADC Divider */ - snd_soc_update_bits(codec, AIC32X4_MADC, - AIC32X4_MADCEN, AIC32X4_MADCEN); - - /* Switch on BCLK_N Divider */ - snd_soc_update_bits(codec, AIC32X4_BCLKN, - AIC32X4_BCLKEN, AIC32X4_BCLKEN); - } + /* Switch on PLL */ + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, AIC32X4_PLLEN); + + /* Switch on NDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, AIC32X4_NDACEN); + + /* Switch on MDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, AIC32X4_MDACEN); + + /* Switch on NADC Divider */ + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, AIC32X4_NADCEN); + + /* Switch on MADC Divider */ + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, AIC32X4_MADCEN); + + /* Switch on BCLK_N Divider */ + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, AIC32X4_BCLKEN); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (aic32x4->master) { - /* Switch off PLL */ - snd_soc_update_bits(codec, AIC32X4_PLLPR, - AIC32X4_PLLEN, 0); - - /* Switch off NDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_NDAC, - AIC32X4_NDACEN, 0); - - /* Switch off MDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_MDAC, - AIC32X4_MDACEN, 0); - - /* Switch off NADC Divider */ - snd_soc_update_bits(codec, AIC32X4_NADC, - AIC32X4_NADCEN, 0); - - /* Switch off MADC Divider */ - snd_soc_update_bits(codec, AIC32X4_MADC, - AIC32X4_MADCEN, 0); - - /* Switch off BCLK_N Divider */ - snd_soc_update_bits(codec, AIC32X4_BCLKN, - AIC32X4_BCLKEN, 0); - } + /* Switch off PLL */ + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, 0); + + /* Switch off NDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, 0); + + /* Switch off MDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, 0); + + /* Switch off NADC Divider */ + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, 0); + + /* Switch off MADC Divider */ + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, 0); + + /* Switch off BCLK_N Divider */ + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, 0); break; case SND_SOC_BIAS_OFF: break; -- 1.7.8.3

3 7

[alsa-devel] [PATCH] ASoC: da7210: Add support for PLL and SRM
by Ashish Chavan 18 Jan '12

18 Jan '12

Hi, Current DA7210 driver does support PLL mode fully. It uses fixed value of input master clock and PLL mode is enabled and disabled based on the sampling frequency being used for playback or recording. It also doesn't support Sample Rate Measurement feature of DA7210 hardware. This patch adds full support for PLL and SRM. Basically following three modes of operation are possible for DA7210 hardware, (1) I2S SLAVE mode with PLL bypassed (2) I2S SLAVE mode with PLL enabled (3) I2S Master mode with PLL enabled This patch adds support for all three modes. Also, in case of SLAVE mode with PLL, it supports SRM (Sample Rate Measurement) feature of the chip. This patch has been tested on DA7210 EVB with Samsung SMDK6410 board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 293 ++++++++++++++++++++++++++++++++++++++------- 1 files changed, 250 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index ab38e93..3945d65 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -145,7 +145,12 @@ #define DA7210_DAI_EN (1 << 7) /*PLL_DIV3 bit fields */ +#define DA7210_PLL_DIV_L_MASK (0xF << 0) +#define DA7210_MCLK_RANGE_MASK (0x30 << 0) +#define DA7210_MCLK_RANGE_32768_HZ (0 << 4) #define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4) +#define DA7210_MCLK_RANGE_20_40_MHZ (2 << 4) +#define DA7210_MCLK_RANGE_40_80_MHZ (3 << 4) #define DA7210_PLL_BYP (1 << 6) /* PLL bit fields */ @@ -161,12 +166,16 @@ #define DA7210_PLL_FS_48000 (0xB << 0) #define DA7210_PLL_FS_88200 (0xE << 0) #define DA7210_PLL_FS_96000 (0xF << 0) +#define DA7210_MCLK_DET_EN (0x1 << 5) +#define DA7210_MCLK_SRM_EN (0x1 << 6) #define DA7210_PLL_EN (0x1 << 7) /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) /* CONTROL bit fields */ +#define DA7210_REG_EN (1 << 0) +#define DA7210_BIAS_EN (1 << 2) #define DA7210_NOISE_SUP_EN (1 << 3) /* IN_GAIN bit fields */ @@ -205,6 +214,47 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) +/* PLL Master mode dividers */ +static const u8 da7210_pll_master_div[] = { + 0xE8, 0x6C, 0x2, /* MCLK=12Mhz Fs=44.1Khz */ + 0xF3, 0x12, 0x7, /* MCLK=12Mhz Fs=48Khz */ + 0xDF, 0x28, 0xC, /* MCLK=13Mhz Fs=44.1Khz */ + 0xE8, 0xFD, 0x5, /* MCLK=13Mhz Fs=48Khz */ + 0xDB, 0x0A, 0xD, /* MCLK=13.5Mhz Fs=44.1Khz */ + 0xE4, 0x82, 0x3, /* MCLK=13.5Mhz Fs=48Khz */ + 0xD4, 0x5A, 0x2, /* MCLK=14.4Mhz Fs=44.1Khz */ + 0xDD, 0x3A, 0x0, /* MCLK=14.4Mhz Fs=48Khz */ + 0xBB, 0x43, 0x9, /* MCLK=19.2Mhz Fs=44.1Khz */ + 0xC1, 0xEB, 0x8, /* MCLK=19.2Mhz Fs=48Khz */ + 0xB9, 0x6D, 0xA, /* MCLK=19.68Mhz Fs=44.1Khz */ + 0xBF, 0xEC, 0x0, /* MCLK=19.68Mhz Fs=48Khz */ + 0xB8, 0xFB, 0xB, /* MCLK=19.8Mhz Fs=44.1Khz */ + 0xBF, 0x70, 0x0, /* MCLK=19.8Mhz Fs=48Khz */ +}; + +/* PLL SRM mode dividers */ +static const u8 da7210_pll_srm_div[] = { + 0xED, 0xBF, 0x5, /* MCLK=12Mhz */ + 0xE4, 0x13, 0x0, /* MCLK=13Mhz */ + 0xDF, 0xC6, 0x8, /* MCLK=13.5Mhz */ + 0xD8, 0xCA, 0x1, /* MCLK=14.4Mhz */ + 0xBE, 0x97, 0x9, /* MCLK=19.2Mhz */ + 0xBC, 0xAC, 0xD, /* MCLK=19.68Mhz */ + 0xBC, 0x35, 0xE, /* MCLK=19.8Mhz */ +}; + +/* Constants for MCLK range */ +enum da7210_mclk_range { + MCLK_32768_HZ, + MCLK_10_20_MHZ, + MCLK_20_40_MHZ, + MCLK_40_80_MHZ, +}; + +enum clk_src { + DA7210_CLKSRC_MCLK +}; + #define DA7210_VERSION "0.0.1" /* @@ -627,6 +677,8 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; + unsigned int mclk_rate; + int master; }; /* @@ -673,8 +725,9 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; - u32 fs, bypass; + u32 fs; /* set DAI source to Left and Right ADC */ snd_soc_write(codec, DA7210_DAI_SRC_SEL, @@ -707,43 +760,33 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - bypass = DA7210_PLL_BYP; break; case 11025: fs = DA7210_PLL_FS_11025; - bypass = 0; break; case 12000: fs = DA7210_PLL_FS_12000; - bypass = DA7210_PLL_BYP; break; case 16000: fs = DA7210_PLL_FS_16000; - bypass = DA7210_PLL_BYP; break; case 22050: fs = DA7210_PLL_FS_22050; - bypass = 0; break; case 32000: fs = DA7210_PLL_FS_32000; - bypass = DA7210_PLL_BYP; break; case 44100: fs = DA7210_PLL_FS_44100; - bypass = 0; break; case 48000: fs = DA7210_PLL_FS_48000; - bypass = DA7210_PLL_BYP; break; case 88200: fs = DA7210_PLL_FS_88200; - bypass = 0; break; case 96000: fs = DA7210_PLL_FS_96000; - bypass = DA7210_PLL_BYP; break; default: return -EINVAL; @@ -753,8 +796,15 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); - snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); + if (da7210->mclk_rate) { + /* PLL mode, disable PLL bypass */ + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, 0); + } else { + /* PLL bypass mode, enable PLL bypass */ + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, + DA7210_PLL_BYP); + } /* Enable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, DA7210_SC_MST_EN); @@ -768,6 +818,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; u32 dai_cfg3; @@ -776,9 +827,11 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: + da7210->master = 1; dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: + da7210->master = 0; dai_cfg1 |= DA7210_DAI_MODE_SLAVE; break; default: @@ -830,10 +883,190 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute) #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static int da7210_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 da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA7210_CLKSRC_MCLK: + switch (freq) { + case 12000000: + case 13000000: + case 13500000: + case 14400000: + case 19200000: + case 19680000: + case 19800000: + da7210->mclk_rate = freq; + return 0; + default: + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/** + * da7210_set_dai_clkdiv: Set codec input clock divider + * @param codec_dai : pointer to codec DAI + * @param div_id : DA7210 has only one clk_div, so div_id is always zero + * @param div : Divider value (configured via control reg MCLK_RANGE) + * @return int : Zero for success, negative error code for error + * + */ +static int da7210_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 mclk_range = 0; + + /* + * Select proper MCLK range in PLL DIV3 register. + * Bits[4:5] decides this. + * Bit[5:4] MCLK Range + * 0:0 - 32.768 KHz + * 0:1 - 10 to 20 MHz + * 1:0 - 20 to 40 MHz + * 1:1 - 40 to 80 MHz + */ + switch (div) { + case MCLK_32768_HZ: + mclk_range = DA7210_MCLK_RANGE_32768_HZ; + break; + case MCLK_10_20_MHZ: + mclk_range = DA7210_MCLK_RANGE_10_20_MHZ; + break; + case MCLK_20_40_MHZ: + mclk_range = DA7210_MCLK_RANGE_20_40_MHZ; + break; + case MCLK_40_80_MHZ: + mclk_range = DA7210_MCLK_RANGE_40_80_MHZ; + break; + default: + dev_err(codec_dai->dev, "Unsupported MCLK range %d\n", div); + return -EINVAL; + } + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_MASK, + mclk_range); + return 0; +} + +/** + * da7210_set_dai_pll :Configure the codec PLL + * @param codec_dai : pointer to codec DAI + * @param pll_id : da7210 has only one pll, so pll_id is always zero + * @param fref : MCLK frequency, should be < 20MHz + * @param fout : FsDM value, Refer page 44 & 45 of datasheet + * @return int : Zero for success, negative error code for error + * + * Note: Supported PLL input frequencies are 12MHz, 13MHz, 13.5MHz, 14.4MHz, + * 19.2MHz, 19.6MHz and 19.8MHz + */ +static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + u8 pll_div1, pll_div2, pll_div3; + int pll_div_offset = 0, fsdm_offset = 0, mclk_offset = 0; + + switch (fref) { + case 12000000: + mclk_offset = 0; + break; + case 13000000: + mclk_offset = 1; + break; + case 13500000: + mclk_offset = 2; + break; + case 14400000: + mclk_offset = 3; + break; + case 19200000: + mclk_offset = 4; + break; + case 19680000: + mclk_offset = 5; + break; + case 19800000: + mclk_offset = 6; + break; + default: + dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", + fref); + return -EINVAL; + } + + switch (fout) { + case 2822400: + fsdm_offset = 0; + break; + case 3072000: + fsdm_offset = 1; + break; + default: + dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", + fout); + return -EINVAL; + } + + if (da7210->master) { + /* In PLL master mode, use Master mode PLL dividers */ + pll_div_offset = (mclk_offset * 2 + fsdm_offset) * 3; + pll_div1 = da7210_pll_master_div[pll_div_offset]; + pll_div2 = da7210_pll_master_div[pll_div_offset + 1]; + pll_div3 = da7210_pll_master_div[pll_div_offset + 2]; + + } else { + /* In PLL slave mode, use SRM mode PLL dividers */ + pll_div_offset = (mclk_offset * 3); + pll_div1 = da7210_pll_srm_div[pll_div_offset]; + pll_div2 = da7210_pll_srm_div[pll_div_offset + 1]; + pll_div3 = da7210_pll_srm_div[pll_div_offset + 2]; + } + + /* Disable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); + /* Write PLL dividers */ + snd_soc_write(codec, DA7210_PLL_DIV1, pll_div1); + snd_soc_write(codec, DA7210_PLL_DIV2, pll_div2); + snd_soc_update_bits(codec, DA7210_PLL_DIV3, + DA7210_PLL_DIV_L_MASK, pll_div3); + + if (da7210->master) { + /* In master mode, no need to enable SRM */ + snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, + DA7210_PLL_EN); + } else { + /* In slave mode, enable SRM and PLL */ + snd_soc_update_bits(codec, DA7210_PLL, + (DA7210_PLL_EN | DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN), + (DA7210_PLL_EN | DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN)); + } + /* Enable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, + DA7210_SC_MST_EN); + return 0; +} + /* DAI operations */ static const struct snd_soc_dai_ops da7210_dai_ops = { .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .set_sysclk = da7210_set_dai_sysclk, + .set_clkdiv = da7210_set_dai_clkdiv, + .set_pll = da7210_set_dai_pll, .digital_mute = da7210_mute, }; @@ -872,24 +1105,13 @@ static int da7210_probe(struct snd_soc_codec *codec) return ret; } - /* FIXME - * - * This driver use fixed value here - * And below settings expects MCLK = 12.288MHz - * - * When you select different MCLK, please check... - * DA7210_PLL_DIV1 val - * DA7210_PLL_DIV2 val - * DA7210_PLL_DIV3 val - * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx - */ + da7210->mclk_rate = 0; /* This will be set from set_sysclk() */ + da7210->master = 0; /* This will be set from set_fmt() */ + + /* Enable internal regulator & bias current */ + snd_soc_write(codec, DA7210_CONTROL, DA7210_REG_EN | DA7210_BIAS_EN); - /* - * make sure that DA7210 use bypass mode before start up - */ snd_soc_write(codec, DA7210_STARTUP1, 0); - snd_soc_write(codec, DA7210_PLL_DIV3, - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); /* * ADC settings @@ -967,21 +1189,6 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); - /* - * If 48kHz sound came, it use bypass mode, - * and when it is 44.1kHz, it use PLL. - * - * This time, this driver sets PLL always ON - * and controls bypass/PLL mode by switching - * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. - * see da7210_hw_params - */ - snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ - snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); - snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); - snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); - /* As suggested by Dialog */ snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); -- 1.7.1

2 3

Re: [alsa-devel] [PATCH 4/4] snd-usb-6fire: add analog input volume control
by Torsten Schenk 18 Jan '12

18 Jan '12

> > static struct __devinitdata snd_kcontrol_new elements[] = { > > @@ -442,6 +497,15 @@ > > .get = usb6fire_control_digital_thru_get, > > .put = usb6fire_control_digital_thru_put > > }, > > + { > > + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > > + .name = "Input 1/2 Capture Volume", > > + .index = 0, > > If this is the only volume element, you don't need to add "1/2", no? The problem is, that there is only an input control for 2 out of 4 analog input channels since channel 3 and 4 use a different ADC and there doesn't seem to be a volume control on that chip. Since this control only changes volume of channel 1 and 2, I wanted to have an indication about that. Thanks, Torsten

2 2

[alsa-devel] [PATCH v4 0/5] ASoC: updates for core support for msbits constraints
by Peter Ujfalusi 18 Jan '12

18 Jan '12

Hi Mark, Changes from v3: - no need to initialize ret to zero in soc_pcm_apply_msb. The function is void. - created against for-3.4 branch (leaving out already applied patches) Intro mail for v1: I would prefer to have the msbits constraints to be applied to only sample sizes bigger than the requested msbits from dais. I'm resending your patch plus the modification which will make the possible. I have also converted the drivers I know have this constraint. If you prefer, you can squash the first two patch. Regards, Peter --- Peter Ujfalusi (5): ASoC: Apply msbits constraint for sample size bigger than the msbits ASoC: omap-dmic: Use core to set the msbits constraint ASoC: omap-mcpdm: Set 24msbits constraint ASoC: tlv320dac33: Use core to set the msbits constraint ASoC: twl4030: Use core to set the msbits constraint sound/soc/codecs/tlv320dac33.c | 3 +-- sound/soc/codecs/twl4030.c | 7 ++++--- sound/soc/omap/omap-dmic.c | 7 +++---- sound/soc/omap/omap-mcpdm.c | 2 ++ sound/soc/soc-pcm.c | 4 +++- 5 files changed, 13 insertions(+), 10 deletions(-) -- 1.7.8.3

2 10