June 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] MEDIA: Fix non-ISA_DMA_API link failure of sound code
by Ralf Baechle 24 Jun '11

24 Jun '11

A build with ISA && ISA_DMA && !ISA_DMA_API results in: CC sound/isa/es18xx.o sound/isa/es18xx.c: In function ‘snd_es18xx_playback1_prepare’: sound/isa/es18xx.c:501:9: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/es18xx.c: In function ‘snd_es18xx_playback_pointer’: sound/isa/es18xx.c:818:3: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[2]: *** [sound/isa/es18xx.o] Error 1 CC sound/isa/sscape.o sound/isa/sscape.c: In function ‘upload_dma_data’: sound/isa/sscape.c:481:3: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[2]: *** [sound/isa/sscape.o] Error 1 CC sound/isa/ad1816a/ad1816a_lib.o sound/isa/ad1816a/ad1816a_lib.c: In function ‘snd_ad1816a_playback_prepare’: sound/isa/ad1816a/ad1816a_lib.c:244:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/ad1816a/ad1816a_lib.c: In function ‘snd_ad1816a_playback_pointer’: sound/isa/ad1816a/ad1816a_lib.c:302:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] sound/isa/ad1816a/ad1816a_lib.c: In function ‘snd_ad1816a_free’: sound/isa/ad1816a/ad1816a_lib.c:544:3: error: implicit declaration of function ‘snd_dma_disable’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/ad1816a/ad1816a_lib.o] Error 1 make[3]: Target `__build' not remade because of errors. make[2]: *** [sound/isa/ad1816a] Error 2 CC sound/isa/es1688/es1688_lib.o sound/isa/es1688/es1688_lib.c: In function ‘snd_es1688_playback_prepare’: sound/isa/es1688/es1688_lib.c:417:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/es1688/es1688_lib.c: In function ‘snd_es1688_playback_pointer’: sound/isa/es1688/es1688_lib.c:509:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/es1688/es1688_lib.o] Error 1 make[3]: Target `__build' not remade because of errors. make[2]: *** [sound/isa/es1688] Error 2 CC sound/isa/gus/gus_dma.o sound/isa/gus/gus_dma.c: In function ‘snd_gf1_dma_program’: sound/isa/gus/gus_dma.c:79:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/gus/gus_dma.c: In function ‘snd_gf1_dma_done’: sound/isa/gus/gus_dma.c:177:3: error: implicit declaration of function ‘snd_dma_disable’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/gus/gus_dma.o] Error 1 CC sound/isa/gus/gus_pcm.o sound/isa/gus/gus_pcm.c: In function ‘snd_gf1_pcm_capture_prepare’: sound/isa/gus/gus_pcm.c:591:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/gus/gus_pcm.c: In function ‘snd_gf1_pcm_capture_pointer’: sound/isa/gus/gus_pcm.c:619:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/gus/gus_pcm.o] Error 1 make[3]: Target `__build' not remade because of errors. make[2]: *** [sound/isa/gus] Error 2 CC sound/isa/sb/sb16_csp.o sound/isa/sb/sb16_csp.c: In function ‘snd_sb_csp_ioctl’: sound/isa/sb/sb16_csp.c:228:227: error: case label does not reduce to an integer constant make[3]: *** [sound/isa/sb/sb16_csp.o] Error 1 CC sound/isa/sb/sb16_main.o sound/isa/sb/sb16_main.c: In function ‘snd_sb16_playback_prepare’: sound/isa/sb/sb16_main.c:276:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/sb/sb16_main.c: In function ‘snd_sb16_playback_pointer’: sound/isa/sb/sb16_main.c:456:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/sb/sb16_main.o] Error 1 CC sound/isa/sb/sb8_main.o sound/isa/sb/sb8_main.c: In function ‘snd_sb8_playback_prepare’: sound/isa/sb/sb8_main.c:172:3: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/sb/sb8_main.c: In function ‘snd_sb8_playback_pointer’: sound/isa/sb/sb8_main.c:425:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/sb/sb8_main.o] Error 1 make[3]: Target `__build' not remade because of errors. make[2]: *** [sound/isa/sb] Error 2 CC sound/isa/wss/wss_lib.o sound/isa/wss/wss_lib.c: In function ‘snd_wss_playback_prepare’: sound/isa/wss/wss_lib.c:1025:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/wss/wss_lib.c: In function ‘snd_wss_playback_pointer’: sound/isa/wss/wss_lib.c:1160:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] sound/isa/wss/wss_lib.c: In function ‘snd_wss_free’: sound/isa/wss/wss_lib.c:1695:3: error: implicit declaration of function ‘snd_dma_disable’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/wss/wss_lib.o] Error 1 The root cause for this is hidden in this Kconfig warning: warning: (RADIO_MIROPCM20) selects SND_ISA which has unmet direct dependencies (SOUND && !M68K && SND && ISA && ISA_DMA_API) Adding a dependency on ISA_DMA_API to RADIO_MIROPCM20 fixes these issues. Signed-off-by: Ralf Baechle <ralf(a)linux-mips.org> drivers/media/radio/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/drivers/media/radio/Kconfig b/drivers/media/radio/Kconfig index e4c97fd..0aeed28 100644 --- a/drivers/media/radio/Kconfig +++ b/drivers/media/radio/Kconfig @@ -168,7 +168,7 @@ config RADIO_MAXIRADIO config RADIO_MIROPCM20 tristate "miroSOUND PCM20 radio" - depends on ISA && VIDEO_V4L2 && SND + depends on ISA && ISA_DMA_API && VIDEO_V4L2 && SND select SND_ISA select SND_MIRO ---help---

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Re: [alsa-devel] [PATCH] SOUND: Fix non-ISA_DMA_API build failure
by Ralf Baechle 24 Jun '11

24 Jun '11

On Fri, Jun 24, 2011 at 10:26:13AM +0200, Takashi Iwai wrote: > Hrm... I still don't understand why ES18XX or others were selected at > the first place. Isn't it covered by the conditional in > sound/isa/Kconfig like below? > > ================================================================ > menuconfig SND_ISA > bool "ISA sound devices" > depends on ISA && ISA_DMA_API > ... > if SND_ISA > ... > config SND_ES18XX > tristate "Generic ESS ES18xx driver" > ... > endif # SND_ISA > ================================================================ > > Isn't SND_ISA=n in your case although ISA_DMA_API=n? The answer is hidden in this Kconfig warning: warning: (RADIO_MIROPCM20) selects SND_ISA which has unmet direct dependencies (SOUND && !M68K && SND && ISA && ISA_DMA_API) This is due to the following in drivers/media/radio/Kconfig: config RADIO_MIROPCM20 tristate "miroSOUND PCM20 radio" depends on ISA && VIDEO_V4L2 && SND select SND_ISA select SND_MIRO So SND_ISA gets forced on even though the dependency on ISA_DMA_API is not fulfilled. That's solved by adding the dependency on ISA_DMA_API to RADIO_MIROPCM20. > Also, adlib driver is really only for ISA, so I see no big reason to > allow this built for non-ISA. With the patch applied: [...] menuconfig SND_ISA bool "ISA sound devices" depends on ISA [...] if SND_ISA config SND_ADLIB tristate "AdLib FM card" select SND_OPL3_LIB [...] So the Adlib driver will still only be built with ISA enabled. The only thing that makes the Adlib driver different from all the others in the ifdef SND_ISA ... endif bracket is that it does not directly or indirectly use the ISA DMA API and that's in the end the reason why sound/isa/Kconfig needs to be changed. I originally approach this a different way but now that I'm explaining the details I notice that it probably makes sense to split this patch into two: o The drivers/media/radio/Kconfig part should be applied for 3.0 and maybe -stable. o The sound/isa/Kconfig part is basically only fixing the dependency for the Adlib driver allowing it to be built on non-ISA_DMA_API system and is material for the next release after 3.0. If you agree I'm going to repost the patch with aproprite log messages. Ralf

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[alsa-devel] SND_SOC_CUSTOM and how to use it
by Raffaele Recalcati 24 Jun '11

24 Jun '11

Hi everibody, I'm integrating the zl38005 Zarling hw codec in ASoC. I'll use analog audio path to send and receive audio from cpu analog port (there are also other ways to interface hw, but that one is implemented in my board). I'll use spi cpu port to read and write registers, but with a little complex protocol. I have this spi commands in a char driver and I suppose I could leave the protocol there also for making easier the parameteres tuning from user-space and also I could call that commands from snd_soc_codec_set_cache_io function. Any comment? Here you are spi protocol that I need to integrate (and that I'll make better befor publishing...) --------------------- drivers/char/zl38005.c ----------------------- /* * Copyright (C) 2011 Bticino S.p.A. * * This program is free software you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option)any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not,write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * * Zarlink zl38005 (Acoustic Echo Canceller) * Some not public information, delivered by Zarlink, * has been used to implement the following code. * */ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/device.h> #include <linux/sysfs.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <linux/kthread.h> #include <linux/fs.h> #include <linux/list.h> #include <linux/smp_lock.h> #include <linux/cdev.h> #include <linux/uaccess.h> #include "zl38005.h" MODULE_DESCRIPTION("zl38005 codec driver"); MODULE_AUTHOR("Raffaele Recalcati <raffaele.recalcati(a)bticino.it>"); MODULE_LICENSE("GPL"); #ifdef DEBUG #define deb(args...) printk(KERN_DEBUG "%s: " __func__, ##args); #else #define deb(args...) #endif /* ZL38005 driver private data */ struct zl38005 { dev_t dev; struct spi_device *spi; struct list_head device_entry; }; /* For registeration of charatcer device */ static struct cdev c_dev; int module_usage_count; struct zl38005 zl38005_data; static DEFINE_MUTEX(zl38005_list_lock); static LIST_HEAD(zl38005_list); struct zl38005_ioctl_par { unsigned int addr; unsigned int data; }; /* ioctl() calls that are permitted to the /dev/zl38005 interface. */ #define ZL_MAGIC 'G' #define ZL_RD_REG _IOR(ZL_MAGIC, 0x09, struct zl38005_ioctl_par) #define ZL_WR_REG _IOR(ZL_MAGIC, 0x0a, struct zl38005_ioctl_par) static struct zl38005_ioctl_par zl38005_ioctl_par; static int zl38005_rd_ctrl_reg(struct device *dev, u8 *val) { int err; u8 cmd; u8 rx_buf[3]; struct zl38005 *zl38005 = dev_get_drvdata(dev); cmd = (CMD_VALID | CMD_READ | CMD_LEN_16_BIT | CMD_TYPE_CTRL); deb("%s TX cmd=%X\n", cmd); err = spi_write_then_read(zl38005->spi, &cmd, sizeof cmd, rx_buf, 2); if (err < 0) { dev_err(dev, "spi_sync failed with %d\n", err); return err; } *val = rx_buf[1]; return 0; } static int zl38005_wr_ctrl_reg(struct device *dev, u8 val, u8 write) { struct spi_message msg; u8 buf[2]; struct spi_transfer xfer = { .len = 2, .tx_buf = buf, }; int err; u8 cmd = (CMD_VALID | CMD_WRITE | CMD_LEN_8_BIT | CMD_TYPE_CTRL); struct zl38005 *zl38005 = dev_get_drvdata(dev); spi_message_init(&msg); buf[0] = cmd; buf[1] = val | (write == 1 ? CTRL_OP_WR : CTRL_OP_RD) | CTRL_WIDTH_16; deb("buf[0]=%02X\n", buf[0]); deb("buf[1]=%02X\n", buf[1]); spi_message_add_tail(&xfer, &msg); err = spi_sync(zl38005->spi, &msg); if (err < 0) { dev_err(dev, "spi_sync failed with %d\n", err); return err; } return 0; } static int zl38005_wr_addr(struct device *dev, u16 val) { struct spi_message msg; u8 buf[3]; struct spi_transfer xfer = { .len = 3, .tx_buf = buf, }; int err; u8 cmd = (CMD_VALID | CMD_WRITE | CMD_LEN_16_BIT | CMD_TYPE_ADDR_WORD); struct zl38005 *zl38005 = dev_get_drvdata(dev); spi_message_init(&msg); buf[0] = cmd; buf[1] = ((val >> 8) & 0xff); buf[2] = (val & 0xff); deb("buf[0]=%02X\n", buf[0]); deb("buf[1]=%02X\n", buf[1]); deb("buf[2]=%02X\n", buf[2]); spi_message_add_tail(&xfer, &msg); err = spi_sync(zl38005->spi, &msg); if (err < 0) { dev_err(dev, "spi_sync failed with %d\n", err); return err; } return 0; } static int zl38005_wr_data(struct device *dev, u16 val) { struct spi_message msg; u8 buf[5]; struct spi_transfer xfer = { .len = 5, .tx_buf = buf, }; int err; u8 cmd = (CMD_VALID | CMD_WRITE | CMD_LEN_32_BIT | CMD_TYPE_WR_DATA); struct zl38005 *zl38005 = dev_get_drvdata(dev); spi_message_init(&msg); buf[0] = cmd; buf[1] = 0; buf[2] = 0; buf[3] = ((val >> 8) & 0xff); buf[4] = (val & 0xff); deb("buf[0]=%02X\n", buf[0]); deb("buf[1]=%02X\n", buf[1]); deb("buf[2]=%02X\n", buf[2]); deb("buf[3]=%02X\n", buf[3]); deb("buf[4]=%02X\n", buf[4]); spi_message_add_tail(&xfer, &msg); err = spi_sync(zl38005->spi, &msg); if (err < 0) { dev_err(dev, "spi_sync failed with %d\n", err); return err; } return 0; } static int zl38005_rd_data(struct device *dev, char *pval) { int err; u8 cmd = (CMD_VALID | CMD_READ | CMD_LEN_32_BIT | CMD_TYPE_RD_DATA); u8 rx_buf[5]; struct zl38005 *zl38005 = dev_get_drvdata(dev); deb("tx cmd=%X\n", cmd); err = spi_write_then_read(zl38005->spi, &cmd, sizeof cmd, rx_buf, sizeof rx_buf); if (err < 0) { dev_err(dev, "spi_write_then_read failed with %d\n", err); return err; } deb("rx_buf[0]=%X\n", rx_buf[0]); deb("rx_buf[1]=%X\n", rx_buf[1]); deb("rx_buf[2]=%X\n", rx_buf[2]); deb("rx_buf[3]=%X\n", rx_buf[3]); deb("rx_buf[4]=%X\n", rx_buf[4]); memcpy(pval, &rx_buf[3], 2); return 0; } static int zl38005_wr_reg(struct device *dev, u16 addr, u16 val) { int t = 0; u8 ctrl_val = CTRL_PENDING; u8 v; /* See if there is a pending operation */ while (t++ < COUNT) { if (!(zl38005_rd_ctrl_reg(dev, &ctrl_val))) { v = ctrl_val & CTRL_PENDING; if (v == 0) break; } schedule(); } if (t == COUNT) { printk(KERN_ERR "zl38005: TIMEOUT for pending operations\n"); return -1; } /* Write Address */ if (zl38005_wr_addr(dev, addr)) return -ENODEV; /* Write Data */ if (zl38005_wr_data(dev, val)) return -ENODEV; /* Writing Ctrl reg */ if (zl38005_wr_ctrl_reg(dev, CTRL_PENDING, 1)) return -ENODEV; return 0; } static int zl38005_rd_reg(struct device *dev, u16 addr, u8 *pval) { int t = 0; u8 ctrl_val = CTRL_PENDING; u8 v; /* See if there is a pending operation */ while (t++ < COUNT) { if (!(zl38005_rd_ctrl_reg(dev, &ctrl_val))) { v = ctrl_val & CTRL_PENDING; if (v == 0) break; } schedule(); } if (t == COUNT) return -1; /* Write Address */ if (zl38005_wr_addr(dev, addr)) return -ENODEV; /* Writing Ctrl reg */ if (zl38005_wr_ctrl_reg(dev, CTRL_PENDING, 0)) return -ENODEV; /* See if there is a pending operation */ while (t++ < COUNT) { if (!(zl38005_rd_ctrl_reg(dev, &ctrl_val))) { v = ctrl_val & CTRL_PENDING; if (v == 0) break; } schedule(); } if (t == COUNT) return -1; /* Read Data */ if (zl38005_rd_data(dev, pval)) return -ENODEV; return 0; } static int zl38005_spi_port_test(struct device *dev) { int t = 0; u8 buf[2]; zl38005_wr_reg(dev, REG_0403, REG_0403_ACK); while (t++ < COUNT) { zl38005_wr_reg(dev, REG_0402, TX_START); if (!(zl38005_rd_reg(dev, REG_0403, buf)) && (buf[0] == RX_START_OK)) break; } if (t < COUNT) return 0; else return -1; } static int zl38005_open(struct inode *inode, struct file *filp) { if (module_usage_count) { printk(KERN_ERR "zl38005 device alrady opened\n"); return -EBUSY; } module_usage_count++; return 0; } static int zl38005_close(struct inode *inode, struct file *filp) { if (module_usage_count) { module_usage_count--; } return 0; } static int zl38005_ioctl(struct inode *inode, struct file *file, unsigned int cmd, unsigned long arg) { struct zl38005 *zl38005 = &zl38005_data; int ret; u8 buf[2]; switch (cmd) { case ZL_WR_REG: if (copy_from_user(&zl38005_ioctl_par, (struct zl38005_ioctl_par *)arg, sizeof(struct zl38005_ioctl_par))) return -EFAULT; deb("ioctl wr ADDR=%X\n", zl38005_ioctl_par.addr); deb("ioctl wr DATA=%X\n", zl38005_ioctl_par.data); ret = zl38005_wr_reg(&zl38005->spi->dev, zl38005_ioctl_par.addr, zl38005_ioctl_par.data); break; case ZL_RD_REG: if (copy_from_user(&zl38005_ioctl_par, (struct zl38005_ioctl_par *)arg, sizeof(struct zl38005_ioctl_par))) { return -EFAULT; } deb("ioctl wr ADDR=%X\n", zl38005_ioctl_par.addr); ret = zl38005_rd_reg(&zl38005->spi->dev, (u16)zl38005_ioctl_par.addr, buf); zl38005_ioctl_par.data = (buf[0] << 8) | buf[1]; if (!ret) { deb("ioctl rd DATA=%X\n", zl38005_ioctl_par.data); if (copy_to_user((struct zl38005_ioctl_par *)arg, &zl38005_ioctl_par, sizeof(struct zl38005_ioctl_par))) return -EFAULT; } else return -EAGAIN; break; default: printk(KERN_DEBUG "ioctl: Invalid Command Value"); ret = -EINVAL; } return ret; } static const struct file_operations zl38005_fops = { .owner = THIS_MODULE, .open = zl38005_open, .release = zl38005_close, .ioctl = zl38005_ioctl, }; /*-------------------------------------------------------------------------*/ static struct class *zl38005_class; static int zl38005_spi_probe(struct spi_device *spi) { struct zl38005 *zl38005 = &zl38005_data; int err; dev_dbg(&spi->dev, "probing zl38005 spi device\n"); spi->bits_per_word = 8; spi->mode = SPI_MODE_0; err = spi_setup(spi); if (err < 0) return err; /* Initialize the driver data */ zl38005->spi = spi; spi_set_drvdata(spi, zl38005); #if 0 /* Trying to see if zl38005 is live */ if (zl38005_spi_port_test(&spi->dev)) { dev_err(&spi->dev, "ZL38005 spi port test NOT passed\n"); } dev_info(&spi->dev, "ZL38005 spi port test passed\n"); #endif return 0; } static int zl38005_spi_remove(struct spi_device *spi) { device_destroy(zl38005_class, zl38005_data.dev); class_destroy(zl38005_class); kfree(spi_get_drvdata(spi)); return 0; } static struct spi_driver zl38005_spi = { .driver = { .name = "zl38005", .owner = THIS_MODULE, }, .probe = zl38005_spi_probe, .remove = zl38005_spi_remove, }; static int __init zl38005_init(void) { struct zl38005 *zl38005 = &zl38005_data; int result; result = alloc_chrdev_region(&zl38005->dev, 0, 1, "zl38005"); if (result < 0) { printk(KERN_ERR "\nzl38005: Module intialization failed.\ could not register character device"); return -ENODEV; } /* Initialize of character device */ cdev_init(&c_dev, &zl38005_fops); c_dev.owner = THIS_MODULE; c_dev.ops = &zl38005_fops; /* addding character device */ result = cdev_add(&c_dev, zl38005->dev, 1); if (result) { printk("NOTICE \nzl38005:Error %d adding zl38005\ ..error no:", result); unregister_chrdev_region(zl38005->dev, 1); } /* registration of character device */ register_chrdev(MAJOR(zl38005->dev), DRIVER_NAME, &zl38005_fops); zl38005_class = class_create(THIS_MODULE, "zl38005"); if (!zl38005_class) { cdev_del(&c_dev); unregister_chrdev(MAJOR(zl38005->dev), DRIVER_NAME); } device_create(zl38005_class, NULL, zl38005->dev, NULL, "zl38005"); result = spi_register_driver(&zl38005_spi); if (result < 0) { class_destroy(zl38005_class); unregister_chrdev(MAJOR(zl38005->dev), zl38005_spi.driver.name); } return result; } module_init(zl38005_init); static void __exit zl38005_exit(void) { spi_unregister_driver(&zl38005_spi); } module_exit(zl38005_exit); /* Module information */ MODULE_AUTHOR("Raffaele Recalcati <raffaele.recalcati(a)bticino.it>"); MODULE_DESCRIPTION("ZL38005 acoustic echo canceller"); MODULE_LICENSE("GPLv2"); MODULE_VERSION(DRIVER_VERSION); --------------------- drivers/char/zl38005.h ----------------------- #ifndef __ZL38005_H #define __ZL38005_H #define DRIVER_NAME "zl38005" #define CMD_VALID (2<<6) #define CMD_READ (0<<5) #define CMD_WRITE (1<<5) #define CMD_LEN_8_BIT (0<<3) #define CMD_LEN_16_BIT (1<<3) #define CMD_LEN_24_BIT (2<<3) #define CMD_LEN_32_BIT (3<<3) #define CMD_TYPE_CTRL (1<<0) #define CMD_TYPE_ADDR_WORD (2<<0) #define CMD_TYPE_RD_DATA (3<<0) #define CMD_TYPE_WR_DATA (4<<0) #define CTRL_OP_RD (0<<0) #define CTRL_OP_WR (1<<0) #define CTRL_WIDTH_8 (0<<1) #define CTRL_WIDTH_16 (1<<1) #define CTRL_WIDTH_24 (2<<1) #define CTRL_WIDTH_32 (3<<1) #define CTRL_PENDING (1<<3) #define CTRL_MEMORY_SEL (1<<8) #define REG_0402 0x0402 #define REG_0403 0x0403 #define TX_START 0x7e01 #define RX_START_OK 0x5e01 #define REG_0403_ACK 0 // x1 #define COUNT 10 #define ZL38005_RX_BUF_SIZE 4 #define ZL38005_TX_BUF_SIZE 4 #endif /* __ZL38005_H */ -- www.opensurf.it

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[alsa-devel] ALSA SoC on Android problems
by Marco Sinigaglia 24 Jun '11

24 Jun '11

Hi, I am trying to enable the WM9714 attached to a S3C6410 through the AC97 Bus. I am using the Android Gingerbread running the kernel linux-2.6.35. I am not sure if this is the right mailing list, please redirect me if it is not. I can see, on the kernel source, that there is the WM9713 codec but not the WM9714. Is this a problem? I am quite sure that the kernel does not find/setup the WM9714. Assuming that is the kernel that has to configure it, do I have to setup something? I can see that, on linux-2.6.35/sound/soc/codecs/wm9713.c, the wm9713_soc_probe function is not call. Nevertheless, the smdk_wm9713.c does run the smdk_init without errors. The kernel log is following: No device for DAI AD73311 Asahi Kasei AK4104 ALSA SoC Codec Driver No device for DAI AK4535 Cirrus Logic CS4270 ALSA SoC Codec Driver No device for DAI PCM3008 HiFi No device for DAI SSM2602 No device for DAI tlv320aic23 No device for DAI UDA134X No device for DAI WM8510 HiFi No device for DAI WM8728 No device for DAI WM8971 No device for DAI WM8990 ADC/DAC Primary ALSA device list: No soundcards found. Indeed, the user space is complaining: E/ALSALib ( 53): external/alsa-lib/src/control/control.c:902:(snd_ctl_open_noupdate) Invalid CTL AndroidOut W/AudioHardwareALSA( 53): Unable to attach mixer to device AndroidOut: No such file or directory E/AudioHardwareALSA( 53): Unable to attach mixer to device default: No such file or directory E/ALSALib ( 53): external/alsa-lib/src/control/control.c:902:(snd_ctl_open_noupdate) Invalid CTL AndroidIn Kernel config : # CONFIG_SOUND_OSS_CORE is not set CONFIG_SND=y CONFIG_SND_TIMER=y CONFIG_SND_PCM=y CONFIG_SND_JACK=y CONFIG_SND_SEQUENCER=y CONFIG_SND_SEQ_DUMMY=y # CONFIG_SND_MIXER_OSS is not set # CONFIG_SND_PCM_OSS is not set # CONFIG_SND_SEQUENCER_OSS is not set CONFIG_SND_DYNAMIC_MINORS=y CONFIG_SND_SUPPORT_OLD_API=y CONFIG_SND_VERBOSE_PROCFS=y CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y # CONFIG_SND_DEBUG_VERBOSE is not set # CONFIG_SND_PCM_XRUN_DEBUG is not set CONFIG_SND_VMASTER=y # CONFIG_SND_RAWMIDI_SEQ is not set # CONFIG_SND_OPL3_LIB_SEQ is not set # CONFIG_SND_OPL4_LIB_SEQ is not set # CONFIG_SND_SBAWE_SEQ is not set # CONFIG_SND_EMU10K1_SEQ is not set CONFIG_SND_AC97_CODEC=y CONFIG_SND_DRIVERS=y # CONFIG_SND_DUMMY is not set # CONFIG_SND_VIRMIDI is not set # CONFIG_SND_MTPAV is not set # CONFIG_SND_SERIAL_U16550 is not set # CONFIG_SND_MPU401 is not set # CONFIG_SND_AC97_POWER_SAVE is not set CONFIG_SND_ARM=y # CONFIG_SND_SPI is not set CONFIG_SND_SOC=y CONFIG_SND_SOC_AC97_BUS=y CONFIG_SND_S3C24XX_SOC=y CONFIG_SND_S3C_I2SV2_SOC=y CONFIG_SND_S3C64XX_SOC_I2S_V4=y CONFIG_SND_S3C_SOC_AC97=y CONFIG_SND_S3C64XX_SOC_WM8580=y CONFIG_SND_SOC_SMDK_WM9713=y CONFIG_SND_SOC_I2C_AND_SPI=y CONFIG_SND_SOC_ALL_CODECS=y CONFIG_SND_SOC_WM_HUBS=y CONFIG_SND_SOC_AC97_CODEC=y CONFIG_SND_SOC_AD1836=y CONFIG_SND_SOC_AD193X=y CONFIG_SND_SOC_AD1980=y CONFIG_SND_SOC_AD73311=y CONFIG_SND_SOC_ADS117X=y CONFIG_SND_SOC_AK4104=y CONFIG_SND_SOC_AK4535=y CONFIG_SND_SOC_AK4642=y CONFIG_SND_SOC_AK4671=y CONFIG_SND_SOC_CS4270=y CONFIG_SND_SOC_DA7210=y CONFIG_SND_SOC_L3=y CONFIG_SND_SOC_PCM3008=y CONFIG_SND_SOC_SPDIF=y CONFIG_SND_SOC_SSM2602=y CONFIG_SND_SOC_STAC9766=y CONFIG_SND_SOC_TLV320AIC23=y CONFIG_SND_SOC_TLV320AIC26=y CONFIG_SND_SOC_TLV320AIC3X=y CONFIG_SND_SOC_TLV320DAC33=y CONFIG_SND_SOC_UDA134X=y CONFIG_SND_SOC_UDA1380=y CONFIG_SND_SOC_WM8510=y CONFIG_SND_SOC_WM8523=y CONFIG_SND_SOC_WM8580=y CONFIG_SND_SOC_WM8711=y CONFIG_SND_SOC_WM8727=y CONFIG_SND_SOC_WM8728=y CONFIG_SND_SOC_WM8731=y CONFIG_SND_SOC_WM8750=y CONFIG_SND_SOC_WM8753=y CONFIG_SND_SOC_WM8776=y CONFIG_SND_SOC_WM8900=y CONFIG_SND_SOC_WM8903=y CONFIG_SND_SOC_WM8904=y CONFIG_SND_SOC_WM8940=y CONFIG_SND_SOC_WM8955=y CONFIG_SND_SOC_WM8960=y CONFIG_SND_SOC_WM8961=y CONFIG_SND_SOC_WM8971=y CONFIG_SND_SOC_WM8974=y CONFIG_SND_SOC_WM8978=y CONFIG_SND_SOC_WM8988=y CONFIG_SND_SOC_WM8990=y CONFIG_SND_SOC_WM8993=y CONFIG_SND_SOC_WM9081=y CONFIG_SND_SOC_WM9705=y CONFIG_SND_SOC_WM9712=y CONFIG_SND_SOC_WM9713=y CONFIG_SND_SOC_MAX9877=y CONFIG_SND_SOC_TPA6130A2=y CONFIG_SND_SOC_WM2000=y CONFIG_SND_SOC_WM9090=y # CONFIG_SOUND_PRIME is not set CONFIG_AC97_BUS=y CONFIG_HID_SUPPORT=y CONFIG_HID=y # CONFIG_HIDRAW is not set # CONFIG_HID_PID is not set Cheers Marco Member of the CSR plc group of companies. CSR plc registered in England and Wales, registered number 4187346, registered office Churchill House, Cambridge Business Park, Cowley Road, Cambridge, CB4 0WZ, United Kingdom More information can be found at www.csr.com. Follow CSR on Twitter at http://twitter.com/CSR_PLC and read our blog at www.csr.com/blog

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[alsa-devel] [PATCH 1/3] ASoC: Add ADAV80x codec driver
by Lars-Peter Clausen 24 Jun '11

24 Jun '11

This patch adds support for the Analog Devices ADAV801 and ADAV803 audio codec. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Sync register on resume * Don't use arrays for the mux enums and controls * Rework PLL handling. Use DAPM for powering the PLL and OSC down and use set_sysclk to select the system clock instead of doing this implicitly in set_pll. --- MAINTAINERS | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adav80x.c | 908 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/adav80x.h | 31 ++ 5 files changed, 946 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/adav80x.c create mode 100644 sound/soc/codecs/adav80x.h diff --git a/MAINTAINERS b/MAINTAINERS index a26e9ec..4f7665c 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -534,6 +534,7 @@ L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://wiki.analog.com/ S: Supported F: sound/soc/codecs/adau* +F: sound/soc/codecs/adav* F: sound/soc/codecs/ad1* F: sound/soc/codecs/ssm* diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2998e65..ff43405 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 + select SND_SOC_ADAV80X select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C @@ -137,6 +138,9 @@ config SND_SOC_ADAU1701 select SIGMA tristate +config SND_SOC_ADAV80X + tristate + config SND_SOC_ADS117X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 51cd3d4..4957431 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -5,6 +5,7 @@ snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-adau1701-objs := adau1701.o +snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o @@ -99,6 +100,7 @@ obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o +obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c new file mode 100644 index 0000000..60f5d8c --- /dev/null +++ b/sound/soc/codecs/adav80x.c @@ -0,0 +1,908 @@ +/* + * ADAV80X Audio Codec driver supporting ADAV801, ADAV803 + * + * Copyright 2011 Analog Devices Inc. + * Author: Yi Li <yi.li(a)analog.com> + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> + +#include "adav80x.h" + +#define ADAV80X_PLAYBACK_CTRL 0x04 +#define ADAV80X_AUX_IN_CTRL 0x05 +#define ADAV80X_REC_CTRL 0x06 +#define ADAV80X_AUX_OUT_CTRL 0x07 +#define ADAV80X_DPATH_CTRL1 0x62 +#define ADAV80X_DPATH_CTRL2 0x63 +#define ADAV80X_DAC_CTRL1 0x64 +#define ADAV80X_DAC_CTRL2 0x65 +#define ADAV80X_DAC_CTRL3 0x66 +#define ADAV80X_DAC_L_VOL 0x68 +#define ADAV80X_DAC_R_VOL 0x69 +#define ADAV80X_PGA_L_VOL 0x6c +#define ADAV80X_PGA_R_VOL 0x6d +#define ADAV80X_ADC_CTRL1 0x6e +#define ADAV80X_ADC_CTRL2 0x6f +#define ADAV80X_ADC_L_VOL 0x70 +#define ADAV80X_ADC_R_VOL 0x71 +#define ADAV80X_PLL_CTRL1 0x74 +#define ADAV80X_PLL_CTRL2 0x75 +#define ADAV80X_ICLK_CTRL1 0x76 +#define ADAV80X_ICLK_CTRL2 0x77 +#define ADAV80X_PLL_CLK_SRC 0x78 +#define ADAV80X_PLL_OUTE 0x7a + +#define ADAV80X_PLL_CLK_SRC_PLL_XIN(pll) 0x00 +#define ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll) (0x40 << (pll)) +#define ADAV80X_PLL_CLK_SRC_PLL_MASK(pll) (0x40 << (pll)) + +#define ADAV80X_ICLK_CTRL1_DAC_SRC(src) ((src) << 5) +#define ADAV80X_ICLK_CTRL1_ADC_SRC(src) ((src) << 2) +#define ADAV80X_ICLK_CTRL1_ICLK2_SRC(src) (src) +#define ADAV80X_ICLK_CTRL2_ICLK1_SRC(src) ((src) << 3) + +#define ADAV80X_PLL_CTRL1_PLLDIV 0x10 +#define ADAV80X_PLL_CTRL1_PLLPD(pll) (0x04 << (pll)) +#define ADAV80X_PLL_CTRL1_XTLPD 0x02 + +#define ADAV80X_PLL_CTRL2_FIELD(pll, x) ((x) << ((pll) * 4)) + +#define ADAV80X_PLL_CTRL2_FS_48(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x00) +#define ADAV80X_PLL_CTRL2_FS_32(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x08) +#define ADAV80X_PLL_CTRL2_FS_44(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0c) + +#define ADAV80X_PLL_CTRL2_SEL(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x02) +#define ADAV80X_PLL_CTRL2_DOUB(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x01) +#define ADAV80X_PLL_CTRL2_PLL_MASK(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0f) + +#define ADAV80X_ADC_CTRL1_MODULATOR_MASK 0x80 +#define ADAV80X_ADC_CTRL1_MODULATOR_128FS 0x00 +#define ADAV80X_ADC_CTRL1_MODULATOR_64FS 0x80 + +#define ADAV80X_DAC_CTRL1_PD 0x80 + +#define ADAV80X_DAC_CTRL2_DIV1 0x00 +#define ADAV80X_DAC_CTRL2_DIV1_5 0x10 +#define ADAV80X_DAC_CTRL2_DIV2 0x20 +#define ADAV80X_DAC_CTRL2_DIV3 0x30 +#define ADAV80X_DAC_CTRL2_DIV_MASK 0x30 + +#define ADAV80X_DAC_CTRL2_INTERPOL_256FS 0x00 +#define ADAV80X_DAC_CTRL2_INTERPOL_128FS 0x40 +#define ADAV80X_DAC_CTRL2_INTERPOL_64FS 0x80 +#define ADAV80X_DAC_CTRL2_INTERPOL_MASK 0xc0 + +#define ADAV80X_DAC_CTRL2_DEEMPH_NONE 0x00 +#define ADAV80X_DAC_CTRL2_DEEMPH_44 0x01 +#define ADAV80X_DAC_CTRL2_DEEMPH_32 0x02 +#define ADAV80X_DAC_CTRL2_DEEMPH_48 0x03 +#define ADAV80X_DAC_CTRL2_DEEMPH_MASK 0x01 + +#define ADAV80X_CAPTURE_MODE_MASTER 0x20 +#define ADAV80X_CAPTURE_WORD_LEN24 0x00 +#define ADAV80X_CAPTURE_WORD_LEN20 0x04 +#define ADAV80X_CAPTRUE_WORD_LEN18 0x08 +#define ADAV80X_CAPTURE_WORD_LEN16 0x0c +#define ADAV80X_CAPTURE_WORD_LEN_MASK 0x0c + +#define ADAV80X_CAPTURE_MODE_LEFT_J 0x00 +#define ADAV80X_CAPTURE_MODE_I2S 0x01 +#define ADAV80X_CAPTURE_MODE_RIGHT_J 0x03 +#define ADAV80X_CAPTURE_MODE_MASK 0x03 + +#define ADAV80X_PLAYBACK_MODE_MASTER 0x10 +#define ADAV80X_PLAYBACK_MODE_LEFT_J 0x00 +#define ADAV80X_PLAYBACK_MODE_I2S 0x01 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_24 0x04 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_20 0x05 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_18 0x06 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_16 0x07 +#define ADAV80X_PLAYBACK_MODE_MASK 0x07 + +static u8 adav80x_default_regs[] = { + 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00, + 0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37, + 0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b, + 0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00, + 0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee, + 0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f, + 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x52, 0x00, +}; + +struct adav80x { + enum snd_soc_control_type control_type; + + enum adav80x_clk_src clk_src; + unsigned int sysclk; + enum adav80x_pll_src pll_src; + + unsigned int dai_fmt[2]; + unsigned int rate; + bool deemph; +}; + +static const char *adav80x_mux_text[] = { + "ADC", + "Playback", + "Aux Playback", +}; + +static const unsigned int adav80x_mux_values[] = { + 0, 2, 3, +}; + +#define ADAV80X_MUX_ENUM_DECL(name, reg, shift) \ + SOC_VALUE_ENUM_DOUBLE_DECL(name, reg, shift, 7, \ + ARRAY_SIZE(adav80x_mux_text), adav80x_mux_text, \ + adav80x_mux_values) + +static ADAV80X_MUX_ENUM_DECL(adav80x_aux_capture_enum, ADAV80X_DPATH_CTRL1, 0); +static ADAV80X_MUX_ENUM_DECL(adav80x_capture_enum, ADAV80X_DPATH_CTRL1, 3); +static ADAV80X_MUX_ENUM_DECL(adav80x_dac_enum, ADAV80X_DPATH_CTRL2, 3); + +static const struct snd_kcontrol_new adav80x_aux_capture_mux_ctrl = + SOC_DAPM_VALUE_ENUM("Route", adav80x_aux_capture_enum); +static const struct snd_kcontrol_new adav80x_capture_mux_ctrl = + SOC_DAPM_VALUE_ENUM("Route", adav80x_capture_enum); +static const struct snd_kcontrol_new adav80x_dac_mux_ctrl = + SOC_DAPM_VALUE_ENUM("Route", adav80x_dac_enum); + +#define ADAV80X_MUX(name, ctrl) \ + SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl) + +static const struct snd_soc_dapm_widget adav80x_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", NULL, ADAV80X_DAC_CTRL1, 7, 1), + SND_SOC_DAPM_ADC("ADC", NULL, ADAV80X_ADC_CTRL1, 5, 1), + + SND_SOC_DAPM_PGA("Right PGA", ADAV80X_ADC_CTRL1, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Left PGA", ADAV80X_ADC_CTRL1, 1, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT("AIFOUT", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFIN", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_AIF_OUT("AIFAUXOUT", "Aux Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFAUXIN", "Aux Playback", 0, SND_SOC_NOPM, 0, 0), + + ADAV80X_MUX("Aux Capture Select", &adav80x_aux_capture_mux_ctrl), + ADAV80X_MUX("Capture Select", &adav80x_capture_mux_ctrl), + ADAV80X_MUX("DAC Select", &adav80x_dac_mux_ctrl), + + SND_SOC_DAPM_INPUT("VINR"), + SND_SOC_DAPM_INPUT("VINL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), + SND_SOC_DAPM_OUTPUT("VOUTL"), + + SND_SOC_DAPM_SUPPLY("SYSCLK", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", ADAV80X_PLL_CTRL1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2", ADAV80X_PLL_CTRL1, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("OSC", ADAV80X_PLL_CTRL1, 1, 1, NULL, 0), +}; + +static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + const char *clk; + + switch (adav80x->clk_src) { + case ADAV80X_CLK_PLL1: + clk = "PLL1"; + break; + case ADAV80X_CLK_PLL2: + clk = "PLL2"; + break; + case ADAV80X_CLK_XTAL: + clk = "OSC"; + break; + default: + return 0; + } + + return strcmp(source->name, clk) == 0; +} + +static int adav80x_dapm_pll_check(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + return adav80x->pll_src == ADAV80X_PLL_SRC_XTAL; +} + + +static const struct snd_soc_dapm_route adav80x_dapm_routes[] = { + { "DAC Select", "ADC", "ADC" }, + { "DAC Select", "Playback", "AIFIN" }, + { "DAC Select", "Aux Playback", "AIFAUXIN" }, + { "DAC", NULL, "DAC Select" }, + + { "Capture Select", "ADC", "ADC" }, + { "Capture Select", "Playback", "AIFIN" }, + { "Capture Select", "Aux Playback", "AIFAUXIN" }, + { "AIFOUT", NULL, "Capture Select" }, + + { "Aux Capture Select", "ADC", "ADC" }, + { "Aux Capture Select", "Playback", "AIFIN" }, + { "Aux Capture Select", "Aux Playback", "AIFAUXIN" }, + { "AIFAUXOUT", NULL, "Aux Capture Select" }, + + { "VOUTR", NULL, "DAC" }, + { "VOUTL", NULL, "DAC" }, + + { "Left PGA", NULL, "VINL" }, + { "Right PGA", NULL, "VINR" }, + { "ADC", NULL, "Left PGA" }, + { "ADC", NULL, "Right PGA" }, + + { "SYSCLK", NULL, "PLL1", adav80x_dapm_sysclk_check }, + { "SYSCLK", NULL, "PLL2", adav80x_dapm_sysclk_check }, + { "SYSCLK", NULL, "OSC", adav80x_dapm_sysclk_check }, + { "PLL1", NULL, "OSC", adav80x_dapm_pll_check }, + { "PLL2", NULL, "OSC", adav80x_dapm_pll_check }, + + { "ADC", NULL, "SYSCLK" }, + { "DAC", NULL, "SYSCLK" }, + { "AIFOUT", NULL, "SYSCLK" }, + { "AIFAUXOUT", NULL, "SYSCLK" }, + { "AIFIN", NULL, "SYSCLK" }, + { "AIFAUXIN", NULL, "SYSCLK" }, +}; + +static int adav80x_set_deemph(struct snd_soc_codec *codec) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (adav80x->deemph) { + switch (adav80x->rate) { + case 32000: + val = ADAV80X_DAC_CTRL2_DEEMPH_32; + break; + case 44100: + val = ADAV80X_DAC_CTRL2_DEEMPH_44; + break; + case 48000: + case 64000: + case 88200: + case 96000: + val = ADAV80X_DAC_CTRL2_DEEMPH_48; + break; + default: + val = ADAV80X_DAC_CTRL2_DEEMPH_NONE; + break; + } + } else { + val = ADAV80X_DAC_CTRL2_DEEMPH_NONE; + } + + return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + ADAV80X_DAC_CTRL2_DEEMPH_MASK, val); +} + +static int adav80x_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int deemph = ucontrol->value.enumerated.item[0]; + + if (deemph > 1) + return -EINVAL; + + adav80x->deemph = deemph; + + return adav80x_set_deemph(codec); +} + +static int adav80x_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = adav80x->deemph; + return 0; +}; + +static const DECLARE_TLV_DB_SCALE(adav80x_inpga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_MINMAX(adav80x_digital_tlv, -9563, 0); + +static const struct snd_kcontrol_new adav80x_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", ADAV80X_DAC_L_VOL, + ADAV80X_DAC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv), + SOC_DOUBLE_R_TLV("Master Capture Volume", ADAV80X_ADC_L_VOL, + ADAV80X_ADC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv), + + SOC_DOUBLE_R_TLV("PGA Capture Volume", ADAV80X_PGA_L_VOL, + ADAV80X_PGA_R_VOL, 0, 0x30, 0, adav80x_inpga_tlv), + + SOC_DOUBLE("Master Playback Switch", ADAV80X_DAC_CTRL1, 0, 1, 1, 0), + SOC_DOUBLE("Master Capture Switch", ADAV80X_ADC_CTRL1, 2, 3, 1, 1), + + SOC_SINGLE("ADC High Pass Filter Switch", ADAV80X_ADC_CTRL1, 6, 1, 0), + + SOC_SINGLE_BOOL_EXT("Playback De-emphasis Switch", 0, + adav80x_get_deemph, adav80x_put_deemph), +}; + +static unsigned int adav80x_port_ctrl_regs[2][2] = { + { ADAV80X_REC_CTRL, ADAV80X_PLAYBACK_CTRL, }, + { ADAV80X_AUX_OUT_CTRL, ADAV80X_AUX_IN_CTRL }, +}; + +static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int capture = 0x00; + unsigned int playback = 0x00; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + capture |= ADAV80X_CAPTURE_MODE_MASTER; + playback |= ADAV80X_PLAYBACK_MODE_MASTER; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + capture |= ADAV80X_CAPTURE_MODE_I2S; + playback |= ADAV80X_PLAYBACK_MODE_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + capture |= ADAV80X_CAPTURE_MODE_LEFT_J; + playback |= ADAV80X_PLAYBACK_MODE_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + capture |= ADAV80X_CAPTURE_MODE_RIGHT_J; + playback |= ADAV80X_PLAYBACK_MODE_RIGHT_J_24; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER, + capture); + snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback); + + adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + return 0; +} + +static int adav80x_set_adc_clock(struct snd_soc_codec *codec, + unsigned int sample_rate) +{ + unsigned int val; + + if (sample_rate <= 48000) + val = ADAV80X_ADC_CTRL1_MODULATOR_128FS; + else + val = ADAV80X_ADC_CTRL1_MODULATOR_64FS; + + snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1, + ADAV80X_ADC_CTRL1_MODULATOR_MASK, val); + + return 0; +} + +static int adav80x_set_dac_clock(struct snd_soc_codec *codec, + unsigned int sample_rate) +{ + unsigned int val; + + if (sample_rate <= 48000) + val = ADAV80X_DAC_CTRL2_DIV1 | ADAV80X_DAC_CTRL2_INTERPOL_256FS; + else + val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS; + + snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK, + val); + + return 0; +} + +static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, + struct snd_soc_dai *dai, snd_pcm_format_t format) +{ + unsigned int val; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAV80X_CAPTURE_WORD_LEN16; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + val = ADAV80X_CAPTRUE_WORD_LEN18; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAV80X_CAPTURE_WORD_LEN20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAV80X_CAPTURE_WORD_LEN24; + break; + default: + break; + } + + snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + ADAV80X_CAPTURE_WORD_LEN_MASK, val); + + return 0; +} + +static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, + struct snd_soc_dai *dai, snd_pcm_format_t format) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (adav80x->dai_fmt[dai->id] != SND_SOC_DAIFMT_RIGHT_J) + return 0; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_16; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_18; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24; + break; + default: + break; + } + + snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1], + ADAV80X_PLAYBACK_MODE_MASK, val); + + return 0; +} + +static int adav80x_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 adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + + if (rate * 256 != adav80x->sysclk) + return -EINVAL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + adav80x_set_playback_pcm_format(codec, dai, + params_format(params)); + adav80x_set_dac_clock(codec, rate); + } else { + adav80x_set_capture_pcm_format(codec, dai, + params_format(params)); + adav80x_set_adc_clock(codec, rate); + } + adav80x->rate = rate; + adav80x_set_deemph(codec); + + return 0; +} + +static int adav80x_set_sysclk(struct snd_soc_codec *codec, + int clk_id, unsigned int freq, int dir) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case ADAV80X_CLK_XIN: + case ADAV80X_CLK_XTAL: + case ADAV80X_CLK_MCLKI: + case ADAV80X_CLK_PLL1: + case ADAV80X_CLK_PLL2: + break; + default: + return -EINVAL; + } + + adav80x->sysclk = freq; + + if (adav80x->clk_src != clk_id) { + unsigned int iclk_ctrl1, iclk_ctrl2; + + adav80x->clk_src = clk_id; + if (clk_id == ADAV80X_CLK_XTAL) + clk_id = ADAV80X_CLK_XIN; + + iclk_ctrl1 = ADAV80X_ICLK_CTRL1_DAC_SRC(clk_id) | + ADAV80X_ICLK_CTRL1_ADC_SRC(clk_id) | + ADAV80X_ICLK_CTRL1_ICLK2_SRC(clk_id); + iclk_ctrl2 = ADAV80X_ICLK_CTRL2_ICLK1_SRC(clk_id); + + snd_soc_write(codec, ADAV80X_ICLK_CTRL1, iclk_ctrl1); + snd_soc_write(codec, ADAV80X_ICLK_CTRL2, iclk_ctrl2); + + snd_soc_dapm_sync(&codec->dapm); + } + + return 0; +} + +static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int pll_ctrl1 = 0; + unsigned int pll_ctrl2 = 0; + unsigned int pll_src; + + switch (source) { + case ADAV80X_PLL_SRC_XTAL: + case ADAV80X_PLL_SRC_XIN: + case ADAV80X_PLL_SRC_MCLKI: + break; + default: + return -EINVAL; + } + + if (!freq_out) + return 0; + + switch (freq_in) { + case 27000000: + break; + case 54000000: + if (source == ADAV80X_PLL_SRC_XIN) { + pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV; + break; + } + default: + return -EINVAL; + } + + if (freq_out > 12288000) { + pll_ctrl2 |= ADAV80X_PLL_CTRL2_DOUB(pll_id); + freq_out /= 2; + } + + /* freq_out = sample_rate * 256 */ + switch (freq_out) { + case 8192000: + pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_32(pll_id); + break; + case 11289600: + pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_44(pll_id); + break; + case 12288000: + pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_48(pll_id); + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAV80X_PLL_CTRL1, ADAV80X_PLL_CTRL1_PLLDIV, + pll_ctrl1); + snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2, + ADAV80X_PLL_CTRL2_PLL_MASK(pll_id), pll_ctrl2); + + if (source != adav80x->pll_src) { + if (source == ADAV80X_PLL_SRC_MCLKI) + pll_src = ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll_id); + else + pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll_id); + + snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC, + ADAV80X_PLL_CLK_SRC_PLL_MASK(pll_id), pll_src); + + adav80x->pll_src = source; + + snd_soc_dapm_sync(&codec->dapm); + } + + return 0; +} + +static int adav80x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int mask = ADAV80X_DAC_CTRL1_PD; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +/* Enforce the same sample rate on all audio interfaces */ +static int adav80x_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + if (!codec->active || !adav80x->rate) + return 0; + + return snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate); +} + +static void adav80x_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + if (!codec->active) + adav80x->rate = 0; +} + +static const struct snd_soc_dai_ops adav80x_dai_ops = { + .set_fmt = adav80x_set_dai_fmt, + .hw_params = adav80x_hw_params, + .startup = adav80x_dai_startup, + .shutdown = adav80x_dai_shutdown, +}; + +#define ADAV80X_PLAYBACK_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000) + +#define ADAV80X_CAPTURE_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) + +#define ADAV80X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver adav80x_dais[] = { + { + .name = "adav80x-hifi", + .id = 0, + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_PLAYBACK_RATES, + .formats = ADAV80X_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_CAPTURE_RATES, + .formats = ADAV80X_FORMATS, + }, + .ops = &adav80x_dai_ops, + }, + { + .name = "adav80x-aux", + .id = 1, + .playback = { + .stream_name = "Aux Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_PLAYBACK_RATES, + .formats = ADAV80X_FORMATS, + }, + .capture = { + .stream_name = "Aux Capture", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_CAPTURE_RATES, + .formats = ADAV80X_FORMATS, + }, + .ops = &adav80x_dai_ops, + }, +}; + +static int adav80x_probe(struct snd_soc_codec *codec) +{ + int ret; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type); + if (ret) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Power down SYSCLK output, power down S/PDIF receiver */ + snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x27); + /* Disable DAC zero flag */ + snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6); + + return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); +} + +static int adav80x_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int adav80x_resume(struct snd_soc_codec *codec) +{ + adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + codec->cache_sync = 1; + snd_soc_cache_sync(codec); + + return 0; +} + +static int adav80x_remove(struct snd_soc_codec *codec) +{ + return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static struct snd_soc_codec_driver adav80x_codec_driver = { + .probe = adav80x_probe, + .remove = adav80x_remove, + .suspend = adav80x_suspend, + .resume = adav80x_resume, + .set_bias_level = adav80x_set_bias_level, + + .set_pll = adav80x_set_pll, + .set_sysclk = adav80x_set_sysclk, + + .reg_word_size = sizeof(u8), + .reg_cache_size = ARRAY_SIZE(adav80x_default_regs), + .reg_cache_default = adav80x_default_regs, + + .controls = adav80x_controls, + .num_controls = ARRAY_SIZE(adav80x_controls), + .dapm_widgets = adav80x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adav80x_dapm_widgets), + .dapm_routes = adav80x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes), +}; + +static int __devinit adav80x_bus_probe(struct device *dev, + enum snd_soc_control_type control_type) +{ + struct adav80x *adav80x; + int ret; + + adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL); + if (!adav80x) + return -ENOMEM; + + dev_set_drvdata(dev, adav80x); + adav80x->control_type = control_type; + + ret = snd_soc_register_codec(dev, &adav80x_codec_driver, + adav80x_dais, ARRAY_SIZE(adav80x_dais)); + if (ret) + kfree(adav80x); + + return ret; +} + +static int __devexit adav80x_bus_remove(struct device *dev) +{ + snd_soc_unregister_codec(dev); + kfree(dev_get_drvdata(dev)); + return 0; +} + +#if defined(CONFIG_SPI_MASTER) +static int __devinit adav80x_spi_probe(struct spi_device *spi) +{ + return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); +} + +static int __devexit adav80x_spi_remove(struct spi_device *spi) +{ + return adav80x_bus_remove(&spi->dev); +} + +static struct spi_driver adav80x_spi_driver = { + .driver = { + .name = "adav801", + .owner = THIS_MODULE, + }, + .probe = adav80x_spi_probe, + .remove = __devexit_p(adav80x_spi_remove), +}; +#endif + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static const struct i2c_device_id adav80x_id[] = { + { "adav803", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adav80x_id); + +static int __devinit adav80x_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + return adav80x_bus_probe(&client->dev, SND_SOC_I2C); +} + +static int __devexit adav80x_i2c_remove(struct i2c_client *client) +{ + return adav80x_bus_remove(&client->dev); +} + +static struct i2c_driver adav80x_i2c_driver = { + .driver = { + .name = "adav803", + .owner = THIS_MODULE, + }, + .probe = adav80x_i2c_probe, + .remove = __devexit_p(adav80x_i2c_remove), + .id_table = adav80x_id, +}; +#endif + +static int __init adav80x_init(void) +{ + int ret = 0; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&adav80x_i2c_driver); + if (ret) + return ret; +#endif + +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&adav80x_spi_driver); +#endif + + return ret; +} +module_init(adav80x_init); + +static void __exit adav80x_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&adav80x_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&adav80x_spi_driver); +#endif +} +module_exit(adav80x_exit); + +MODULE_DESCRIPTION("ASoC ADAV80x driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_AUTHOR("Yi Li <yi.li(a)analog.com>>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adav80x.h b/sound/soc/codecs/adav80x.h new file mode 100644 index 0000000..a13aefe --- /dev/null +++ b/sound/soc/codecs/adav80x.h @@ -0,0 +1,31 @@ +/* + * header file for ADAV80X parts + * + * Copyright 2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef _ADAV80X_H +#define _ADAV80X_H + +enum adav80x_pll_src { + ADAV80X_PLL_SRC_XIN, + ADAV80X_PLL_SRC_XTAL, + ADAV80X_PLL_SRC_MCLKI, +}; + +enum adav80x_pll { + ADAV80X_PLL1 = 0, + ADAV80X_PLL2 = 1, +}; + +enum adav80x_clk_src { + ADAV80X_CLK_XIN = 0, + ADAV80X_CLK_MCLKI = 1, + ADAV80X_CLK_PLL1 = 2, + ADAV80X_CLK_PLL2 = 3, + ADAV80X_CLK_XTAL = 6, +}; + +#endif -- 1.7.2.5

3 5

[alsa-devel] ASoC updates for 3.1
by Mark Brown 24 Jun '11

24 Jun '11

The following changes since commit f192c0ab2420aa59d6fae26564b14adf97456063: ASoC: SMDKV310: Enable SPDIF device (2011-06-10 11:05:09 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Arnaud Patard (2): ASoC: imx: Remove unused Kconfig SND_MXC_SOC_SSI entry ASoC: imx: add missing module informations Barry Song (1): ASoC: AD1836: rename suspend/resume funcs Dan Carpenter (1): ASoC: adau1701: signedness bug in adau1701_write() Daniel Mack (1): ASoC: pxa-ssp: Correct check for stream presence Johannes Stezenbach (2): ASoC: add STA32X codec driver ASoC: add WM8782 ADC Codec Driver Lars-Peter Clausen (2): ASoC: Add ADAU1701 codec driver ASoC: Blackfin: Add bf5xx-adau1701 machine driver Liam Girdwood (1): ASoC: dapm - Refactor widget IO functions in preparation for platform widgets. Mark Brown (15): ASoC: Factor out redundant read() functions ASoC: Move register I/O code into a separate file ASoC: Add weak routes for sidetone style paths ASoC: Don't use codec->control_data in bulk write ASoC: Remove unused and about to be broken SND_SOC_CUSTOM I/O bus Merge branch 'for-3.0' into for-3.1 ASoC: Fix mismerge with release branch ASoC: Clean up some coding style nits in the bf5xx-i2s-pcm driver ASoC: Add adau* drivers to the Analog Devices file patterns ASoC: Remove adau1701 from SND_SOC_ALL_CODECS due to Sigma dependency ASoC: Fix DAPM sequence run for per-widget I/O methods ASoC: Trigger wm_hubs series update startup off a separate flag ASoC: Allow suppression of series updates on wm_hubs devices ASoC: Add basic WM8918 support Merge branch 'for-3.0' into for-3.1 Mike Frysinger (5): ASoC: AD1836: clean up comment headers ASoC: AD1836: drop unnecessary spi register check ASoC: AD1836: fix intermixed tab/space indentation ASoC: AD1836: fix codec name ASoC: Blackfin: allow SPI for SSM2602 parts Taylor Hutt (1): ASoC: codecs: Max98095: Fix logging of hardware revision. MAINTAINERS | 1 + include/sound/soc-dapm.h | 3 + include/sound/soc.h | 3 +- sound/soc/Makefile | 3 +- sound/soc/blackfin/Kconfig | 14 +- sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf5xx-i2s-pcm.c | 6 +- sound/soc/blackfin/bfin-eval-adau1701.c | 139 ++++++ sound/soc/codecs/Kconfig | 14 +- sound/soc/codecs/Makefile | 6 + sound/soc/codecs/ad1836.c | 78 ++-- sound/soc/codecs/ad1836.h | 21 +- sound/soc/codecs/adau1701.c | 549 ++++++++++++++++++++++ sound/soc/codecs/adau1701.h | 17 + sound/soc/codecs/max98095.c | 8 +- sound/soc/codecs/sta32x.c | 777 +++++++++++++++++++++++++++++++ sound/soc/codecs/sta32x.h | 210 +++++++++ sound/soc/codecs/wm8782.c | 80 ++++ sound/soc/codecs/wm8904.c | 1 + sound/soc/codecs/wm8993.c | 1 + sound/soc/codecs/wm8994.c | 1 + sound/soc/codecs/wm_hubs.c | 5 +- sound/soc/codecs/wm_hubs.h | 2 + sound/soc/imx/Kconfig | 7 - sound/soc/imx/imx-pcm-dma-mx2.c | 2 + sound/soc/imx/imx-ssi.c | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 4 +- sound/soc/soc-cache.c | 416 ----------------- sound/soc/soc-dapm.c | 142 ++++++- sound/soc/soc-io.c | 396 ++++++++++++++++ 30 files changed, 2399 insertions(+), 511 deletions(-) create mode 100644 sound/soc/blackfin/bfin-eval-adau1701.c create mode 100644 sound/soc/codecs/adau1701.c create mode 100644 sound/soc/codecs/adau1701.h create mode 100644 sound/soc/codecs/sta32x.c create mode 100644 sound/soc/codecs/sta32x.h create mode 100644 sound/soc/codecs/wm8782.c create mode 100644 sound/soc/soc-io.c

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[alsa-devel] ASoC updates for 3.0
by Mark Brown 24 Jun '11

24 Jun '11

The following changes since commit 33195500edf260e8c8809ab9dfc67f50e0ce031f: ASoC: SAMSUNG: Fix the incorrect referencing of I2SCON register (2011-06-10 10:54:46 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 Arnaud Patard (2): ASoC: imx: Remove unused Kconfig SND_MXC_SOC_SSI entry ASoC: imx: add missing module informations Daniel Mack (1): ASoC: pxa-ssp: Correct check for stream presence Mark Brown (1): ASoC: Remove unused and about to be broken SND_SOC_CUSTOM I/O bus include/sound/soc.h | 3 +-- sound/soc/imx/Kconfig | 7 ------- sound/soc/imx/imx-pcm-dma-mx2.c | 2 ++ sound/soc/imx/imx-ssi.c | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 4 ++-- sound/soc/soc-cache.c | 3 --- 6 files changed, 6 insertions(+), 15 deletions(-)

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Re: [alsa-devel] [PATCH 01/37] Remove unneeded version.h includes from sound/
by Takashi Iwai 24 Jun '11

24 Jun '11

At Thu, 23 Jun 2011 23:54:40 +0200 (CEST), Jesper Juhl wrote: > > In the sound/ directory there are two files (flagged by 'make > versioncheck'); sound/pci/asihpi/asihpi.c and > sound/soc/codecs/wm8991.c that include linux/version.h although they > don't need it. This patch removes the unneeded includes. > > Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> Thanks, applied to sound git tree now. Takashi > --- > sound/pci/asihpi/asihpi.c | 1 - > sound/soc/codecs/wm8991.c | 1 - > 2 files changed, 0 insertions(+), 2 deletions(-) > > diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c > index 2ca6f4f..e3569bd 100644 > --- a/sound/pci/asihpi/asihpi.c > +++ b/sound/pci/asihpi/asihpi.c > @@ -27,7 +27,6 @@ > #include "hpioctl.h" > > #include <linux/pci.h> > -#include <linux/version.h> > #include <linux/init.h> > #include <linux/jiffies.h> > #include <linux/slab.h> > diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c > index 3c2ee1b..6af23d0 100644 > --- a/sound/soc/codecs/wm8991.c > +++ b/sound/soc/codecs/wm8991.c > @@ -13,7 +13,6 @@ > > #include <linux/module.h> > #include <linux/moduleparam.h> > -#include <linux/version.h> > #include <linux/kernel.h> > #include <linux/init.h> > #include <linux/delay.h> > -- > 1.7.5.2 > > > -- > Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ > Don't top-post http://www.catb.org/jargon/html/T/top-post.html > Plain text mails only, please. >

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[alsa-devel] [PATCH] ALSA: HDA: Pinfix quirk for HP Z200 Workstation
by David Henningsson 24 Jun '11

24 Jun '11

BIOS lists the internal speaker as an internal line-out. Change to internal speaker + model=auto for better auto-mute capabilities. Relevant alsa-info is here: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/754964/+attachme… -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] [ASoc] TDM over one SSC channel on AT91 possible?
by Friedrich Lobenstock 24 Jun '11

24 Jun '11

Hello! The background: I'm just designing an ATMEL AT91RM9200 based board which got two TLV320AIC33 on board. It is either possible to hook up the two TLV320AIC33 to a separate SSC (synchronous serial control) channel each or use one SSC channel and configure it for TDM (time division multiplex) mode. There is already a driver for TLV320AIC14k/TLV320AIC2k in the development tree, so adding support for the TLV320AIC33 should be feasible. What I'm not sure about is the fact if ASOC is prepared for using channels, especially SSC, in TDM mode. Is the core ASOC structure prepared to handle that? Would we be better off instantiate one drive twice each time with different parameters for the SSC channel? Thanks for any feedback. -- MfG / Regards Friedrich Lobenstock _________________________________________________________ SCOTTY Group Austria GmbH a SCOTTY Group plc company, http://www.scottygroup.com/ _________________________________________________________

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