[alsa-devel] [PATCH] ALSA SOC driver for s3c24xx with uda1341
This patch adds support for the UDA1341 codec and a sound card definition for one of these UDAs connected to the s3c24xx. It is *heavily* based on the one made by Zoltan Devai but:
1 since the UDA is the only use of the L3 protocol I can find I just embedded a stripped-down L3 bit-banging algorithm from the original RMK work. It is really small.
2 the driver has the possibility to specify the pins used by codec via platform data so it can work on SMDK2410, SMDK2440 or any custom design.
3 it tries to guess the right clock source and divider so it is not tied to a particular crystal used.
4 it fixes some bugs.
Thank you for reviews/comments.
Generated on 20081108 against v2.6.27
Signed-off-by: Christian Pellegrin chripell@fsfe.org --- include/sound/s3c24xx_uda1341.h | 41 ++ sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/uda1341/Makefile | 3 + sound/soc/codecs/uda1341/l3.c | 106 +++++ sound/soc/codecs/uda1341/l3.h | 8 + sound/soc/codecs/uda1341/uda1341.c | 537 ++++++++++++++++++++++ sound/soc/codecs/uda1341/uda1341.h | 47 ++ sound/soc/codecs/uda1341/uda1341_platform_data.h | 18 + sound/soc/s3c24xx/Kconfig | 7 +- sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/s3c24xx_uda1341.c | 314 +++++++++++++ 12 files changed, 1086 insertions(+), 1 deletions(-)
diff --git a/include/sound/s3c24xx_uda1341.h b/include/sound/s3c24xx_uda1341.h new file mode 100644 index 0000000..bd80bd3 --- /dev/null +++ b/include/sound/s3c24xx_uda1341.h @@ -0,0 +1,41 @@ +#ifndef _S3C24XX_UDA1341_H_ +#define _S3C24XX_UDA1341_H_ 1 + +/* + +Example usage (pins for SMDK2410). Put something like this in your +machine file: + +... + +static struct s3c24xx_uda1341_platform_data s3c24xx_uda1341_data = { + .l3_clk = S3C2410_GPB4, + .l3_data = S3C2410_GPB3, + .l3_mode = S3C2410_GPB2, +}; + +static struct platform_device s3c24xx_uda1341 = { + .name = "s3c24xx_uda1341", + .dev = { + .platform_data = &s3c24xx_uda1341_data, + } +}; + +... + +static struct platform_device *smdk2410_devices[] __initdata = { +... + &s3c24xx_uda1341, +... +}; + + */ + +struct s3c24xx_uda1341_platform_data { + int l3_clk; + int l3_mode; + int l3_data; + void (*power) (int); +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1db04a2..4b483b7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -50,3 +50,6 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_TLV320AIC3X tristate depends on I2C + +config SND_SOC_UDA1341 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d7b97ab..cbace60 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,3 +23,4 @@ obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_UDA1341) += uda1341/ diff --git a/sound/soc/codecs/uda1341/Makefile b/sound/soc/codecs/uda1341/Makefile new file mode 100644 index 0000000..f9869ba --- /dev/null +++ b/sound/soc/codecs/uda1341/Makefile @@ -0,0 +1,3 @@ +snd-soc-uda1341-objs := l3.o uda1341.o + +obj-$(CONFIG_SND_SOC_UDA1341) += snd-soc-uda1341.o \ No newline at end of file diff --git a/sound/soc/codecs/uda1341/l3.c b/sound/soc/codecs/uda1341/l3.c new file mode 100644 index 0000000..b65b352 --- /dev/null +++ b/sound/soc/codecs/uda1341/l3.c @@ -0,0 +1,106 @@ +/* + * L3 code + * + * Copyright (C) 2008, Christian Pellegrin chripell@evolware.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * + * based on: + * + * L3 bus algorithm module. + * + * Copyright (C) 2001 Russell King, All Rights Reserved. + * + * + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/init.h> +#include <linux/errno.h> +#include <linux/sched.h> + +#include "l3.h" + +/*#define L3_DEBUG 1*/ +#ifdef L3_DEBUG +#define DBG(format, arg...) \ + printk(KERN_DEBUG "L3: %s:" format "\n" , __func__, ## arg) +#else +#define DBG(format, arg...) do {} while (0) +#endif + +#define setdat(adap, val) (adap->setdat(adap, val)) +#define setclk(adap, val) (adap->setclk(adap, val)) +#define setmode(adap, val) (adap->setmode(adap, val)) + +/* + * Send one byte of data to the chip. Data is latched into the chip on + * the rising edge of the clock. + */ +static void sendbyte(struct uda1341_platform_data *adap, unsigned int byte) +{ + int i; + + DBG("%02x", byte); + + for (i = 0; i < 8; i++) { + setclk(adap, 0); + udelay(adap->data_hold); + setdat(adap, byte & 1); + udelay(adap->data_setup); + setclk(adap, 1); + udelay(adap->clock_high); + byte >>= 1; + } +} + +/* + * Send a set of bytes to the chip. We need to pulse the MODE line + * between each byte, but never at the start nor at the end of the + * transfer. + */ +static void sendbytes(struct uda1341_platform_data *adap, const u8 *buf, + int len) +{ + int i; + + for (i = 0; i < len; i++) { + if (i) { + udelay(adap->mode_hold); + setmode(adap, 0); + udelay(adap->mode); + } + setmode(adap, 1); + udelay(adap->mode_setup); + sendbyte(adap, buf[i]); + } +} + +int l3_write(struct uda1341_platform_data *adap, u8 addr, u8 *data, int len) +{ + setclk(adap, 1); + setdat(adap, 1); + setmode(adap, 1); + udelay(adap->mode); + + setmode(adap, 0); + udelay(adap->mode_setup); + sendbyte(adap, addr); + udelay(adap->mode_hold); + + sendbytes(adap, data, len); + + setclk(adap, 1); + setdat(adap, 1); + setmode(adap, 0); + + return len; +} + + diff --git a/sound/soc/codecs/uda1341/l3.h b/sound/soc/codecs/uda1341/l3.h new file mode 100644 index 0000000..2c31e0c --- /dev/null +++ b/sound/soc/codecs/uda1341/l3.h @@ -0,0 +1,8 @@ +#ifndef _L3_H_ +#define _L3_H_ 1 + +#include "uda1341_platform_data.h" + +int l3_write(struct uda1341_platform_data *adap, u8 addr, u8 *data, int len); + +#endif diff --git a/sound/soc/codecs/uda1341/uda1341.c b/sound/soc/codecs/uda1341/uda1341.c new file mode 100644 index 0000000..93cadf0 --- /dev/null +++ b/sound/soc/codecs/uda1341/uda1341.c @@ -0,0 +1,537 @@ +/* + * uda1341.c -- UDA1341 ALSA SoC Codec driver + * + * Modifications by Christian Pellegrin chripell@evolware.org + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "uda1341.h" +#include "l3.h" + +/*#define UDA1341_DEBUG 1*/ +#ifdef UDA1341_DEBUG +#define DBG(format, arg...) \ + printk(KERN_DEBUG "UDA1341: %s:" format "\n" , __func__, ## arg) +#else +#define DBG(format, arg...) do {} while (0) +#endif + +#define UDA1341_RATES SNDRV_PCM_RATE_8000_48000 +#define UDA1341_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) + +struct uda1341_priv { + int sysclk; + int dai_fmt; +}; + +/* In-data addresses are hard-coded into the reg-cache values */ +static const char uda1341_reg[UDA1341_REGS_NUM] = { + /* Extended address registers */ + 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, + /* Status, data regs */ + 0x00, 0x83, 0x00, 0x40, 0x80, 0x00, +}; + +/* + * The codec has no support for reading its registers except for peak level... + */ +static inline unsigned int uda1341_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= UDA1341_REGS_NUM) + return -1; + return cache[reg]; +} + +/* + * Write the register cache + */ +static inline void uda1341_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, unsigned int value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= UDA1341_REGS_NUM) + return; + cache[reg] = value; +} + +/* + * Write to the uda1341 registers + * + */ +static int uda1341_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + int ret; + u8 addr; + u8 data = value; + + DBG("reg: %02X, value:%02X", reg, value); + + if (reg >= UDA1341_REGS_NUM) { + DBG("Unkown register: reg: %d", reg); + return -EINVAL; + } + + uda1341_write_reg_cache(codec, reg, value); + + switch (reg) { + case UDA1341_STATUS0: + case UDA1341_STATUS1: + addr = UDA1341_STATUS_ADDR; + break; + case UDA1341_DATA000: + case UDA1341_DATA001: + case UDA1341_DATA010: + addr = UDA1341_DATA0_ADDR; + break; + case UDA1341_DATA1: + addr = UDA1341_DATA1_ADDR; + break; + default: + /* It's an extended address register */ + addr = (reg | UDA1341_EXTADDR_PREFIX); + + ret = l3_write((struct uda1341_platform_data *) + codec->control_data, + UDA1341_DATA0_ADDR, &addr, 1); + if (ret != 1) + return -EIO; + + addr = UDA1341_DATA0_ADDR; + data = (value | UDA1341_EXTDATA_PREFIX); + break; + } + + ret = l3_write((struct uda1341_platform_data *) codec->control_data, + addr, &data, 1); + if (ret != 1) + return -EIO; + + return 0; +} + +static inline void uda1341_reset(struct snd_soc_codec *codec) +{ + u8 reset_reg = uda1341_read_reg_cache(codec, UDA1341_STATUS0); + uda1341_write(codec, UDA1341_STATUS0, reset_reg | (1<<6)); + mdelay(1); + uda1341_write(codec, UDA1341_STATUS0, reset_reg & ~(1<<6)); +} + +static int uda1341_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = uda1341_read_reg_cache(codec, UDA1341_DATA010); + + DBG("mute: %d", mute); + + if (mute) + mute_reg |= (1<<2); + else + mute_reg &= ~(1<<2); + + uda1341_write(codec, UDA1341_DATA010, mute_reg & ~(1<<2)); + + return 0; +} + +static int uda1341_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda1341_priv *uda1341 = codec->private_data; + + u8 hw_params = uda1341_read_reg_cache(codec, UDA1341_STATUS0); + hw_params &= STATUS0_SYSCLK_MASK; + hw_params &= STATUS0_DAIFMT_MASK; + + DBG("sysclk: %d, rate:%d", uda1341->sysclk, params_rate(params)); + + /* set SYSCLK / fs ratio */ + switch (uda1341->sysclk / params_rate(params)) { + case 512: + break; + case 384: + hw_params |= (1<<4); + break; + case 256: + hw_params |= (1<<5); + break; + default: + return -EINVAL; + break; + } + + DBG("dai_fmt: %d, params_format:%d", uda1341->dai_fmt, + params_format(params)); + + + /* set DAI format and word length */ + switch (uda1341->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + hw_params |= (1<<1); + break; + case SNDRV_PCM_FORMAT_S18_3LE: + hw_params |= (1<<2); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + hw_params |= ((1<<2) | (1<<1)); + break; + default: + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_LEFT_J: + hw_params |= (1<<3); + break; + default: + return -EINVAL; + } + + uda1341_write(codec, UDA1341_STATUS0, hw_params); + + return 0; +} + +static int uda1341_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 uda1341_priv *uda1341 = codec->private_data; + + DBG("clk_id: %d, freq: %d, dir: %d", clk_id, freq, dir); + + /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable + we'll error out on set_hw_params if it's not OK */ + if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) { + uda1341->sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int uda1341_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda1341_priv *uda1341 = codec->private_data; + + DBG("fmt: %08X", fmt); + + /* codec supports only full slave mode */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + + /* no support for clock inversion */ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + + /* We can't setup DAI format here as it depends on the word bit num */ + /* so let's just store the value for later */ + uda1341->dai_fmt = fmt; + + return 0; +} + +static int uda1341_dapm_event(struct snd_soc_codec *codec, int event) +{ + u8 reg; + + DBG("event: %08X", event); + + switch (event) { + case SNDRV_CTL_POWER_D0: /* full On */ + /* ADC, DAC on */ + reg = uda1341_read_reg_cache(codec, UDA1341_STATUS1); + uda1341_write(codec, UDA1341_STATUS1, reg | 0x03); + break; + case SNDRV_CTL_POWER_D1: /* partial On */ + case SNDRV_CTL_POWER_D2: /* partial On */ + case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + break; + case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + /* mute, ADC, DAC power off */ + reg = uda1341_read_reg_cache(codec, UDA1341_STATUS1); + uda1341_write(codec, UDA1341_STATUS1, reg & ~(0x03)); + break; + } + return 0; +} + +static const char *uda1341_dsp_setting[] = {"Flat", "Minimum1", + "Minimum2", "Maximum"}; +static const char *uda1341_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *uda1341_mixmode[] = {"DD", "Input 2", + "Input 2", "Digital mixer"}; + +static const struct soc_enum uda1341_mixer_enum[] = { +SOC_ENUM_SINGLE(UDA1341_DATA010, 0, 0x03, uda1341_dsp_setting), +SOC_ENUM_SINGLE(UDA1341_DATA010, 3, 0x03, uda1341_deemph), +SOC_ENUM_SINGLE(UDA1341_EA010, 0, 0x03, uda1341_mixmode), +}; + +static const struct snd_kcontrol_new uda1341_snd_controls[] = { +SOC_SINGLE("Playback Volume", UDA1341_DATA000, 0, 0x3F, 1), +SOC_SINGLE("Mic gain", UDA1341_EA010, 2, 0x07, 0), +SOC_SINGLE("Channel 1 mixer gain", UDA1341_EA000, 0, 0x1F, 1), +SOC_SINGLE("Channgel 2 mixer gain", UDA1341_EA001, 0, 0x1F, 1), +SOC_SINGLE("Input channel 2 amp gain", UDA1341_EA101, 0, 0x1F, 0), + +SOC_SINGLE("Bass boost", UDA1341_DATA001, 2, 0xF, 0), +SOC_SINGLE("Treble", UDA1341_DATA001, 0, 3, 0), + +SOC_ENUM("DSP setting", uda1341_mixer_enum[0]), +SOC_ENUM("Playback De-emphasis", uda1341_mixer_enum[1]), +SOC_ENUM("Mixer mode", uda1341_mixer_enum[2]), + +/* This should be an ext control with own handler, if one wants + to set the values in 0.5dB steps instead of 3dB */ +SOC_SINGLE("AGC output level", UDA1341_EA110, 0, 0x03, 1), +SOC_SINGLE("AGC time const", UDA1341_EA110, 2, 0x07, 0), + +SOC_SINGLE("DAC Gain switch", UDA1341_STATUS1, 6, 1, 0), +SOC_SINGLE("ADC Gain switch", UDA1341_STATUS1, 5, 1, 0), +SOC_SINGLE("ADC Polarity switch", UDA1341_STATUS1, 4, 1, 0), +SOC_SINGLE("DAC Polarity switch", UDA1341_STATUS1, 3, 1, 0), +SOC_SINGLE("Double speed playback switch", UDA1341_STATUS1, 2, 1, 0), +}; + +static int uda1341_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(uda1341_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&uda1341_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +struct snd_soc_dai uda1341_dai = { + .name = "UDA1341", + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1341_RATES, + .formats = UDA1341_FORMATS, + }, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1341_RATES, + .formats = UDA1341_FORMATS, + }, + /* pcm operations */ + .ops = { + .hw_params = uda1341_hw_params, + }, + /* DAI operations */ + .dai_ops = { + .digital_mute = uda1341_mute, + .set_sysclk = uda1341_set_dai_sysclk, + .set_fmt = uda1341_set_dai_fmt, + } +}; +EXPORT_SYMBOL(uda1341_dai); + + +static int uda1341_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct uda1341_priv *uda1341; + void *codec_setup_data = socdev->codec_data; + int ret = -ENOMEM; + struct uda1341_platform_data *pd; + + printk(KERN_INFO "UDA1341 SoC Audio Codec\n"); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return ret; + + codec = socdev->codec; + + uda1341 = kzalloc(sizeof(struct uda1341_priv), GFP_KERNEL); + if (uda1341 == NULL) + goto priv_err; + + codec->private_data = uda1341; + + codec->reg_cache = kmemdup(uda1341_reg, sizeof(uda1341_reg), + GFP_KERNEL); + + if (codec->reg_cache == NULL) + goto reg_err; + + mutex_init(&codec->mutex); + + codec->reg_cache_size = sizeof(uda1341_reg); + codec->reg_cache_step = 1; + + codec->name = "UDA1341"; + codec->owner = THIS_MODULE; + codec->dai = &uda1341_dai; + codec->num_dai = 1; + codec->read = uda1341_read_reg_cache; + codec->write = uda1341_write; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + if (!codec_setup_data) { + printk(KERN_ERR "UDA1341 SoC codec: " + "missing L3 bitbang function\n"); + ret = -ENODEV; + goto pcm_err; + } + + codec->control_data = codec_setup_data; + pd = (struct uda1341_platform_data *) codec_setup_data; + + if (pd->power) + pd->power(1); + + uda1341_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "UDA1341: failed to register pcms\n"); + goto pcm_err; + } + + ret = uda1341_add_controls(codec); + if (ret < 0) { + printk(KERN_ERR "UDA1341: failed to register controls\n"); + goto pcm_err; + } + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "UDA1341: failed to register card\n"); + goto card_err; + } + + return 0; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); +reg_err: + kfree(codec->private_data); +priv_err: + kfree(codec); + return ret; +} + +/* power down chip */ +static int uda1341_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct uda1341_platform_data *pd; + + uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + + pd = (struct uda1341_platform_data *) codec->control_data; + if (pd->power) + pd->power(0); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} + +#if defined(CONFIG_PM) +static int uda1341_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct uda1341_platform_data *pd; + + uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + + pd = (struct uda1341_platform_data *) codec->control_data; + if (pd->power) + pd->power(0); + + return 0; +} + +static int uda1341_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct uda1341_platform_data *pd; + int i; + u8 *cache = codec->reg_cache; + + pd = (struct uda1341_platform_data *) codec->control_data; + if (pd->power) + pd->power(1); + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(uda1341_reg); i++) + codec->write(codec, i, *cache++); + uda1341_dapm_event(codec, SNDRV_CTL_POWER_D0); + return 0; +} +#endif /* CONFIG_PM */ + +struct snd_soc_codec_device soc_codec_dev_uda1341 = { + .probe = uda1341_soc_probe, + .remove = uda1341_soc_remove, +#if defined(CONFIG_PM) + .suspend = uda1341_soc_suspend, + .resume = uda1341_soc_resume, +#endif /* CONFIG_PM */ +}; +EXPORT_SYMBOL(soc_codec_dev_uda1341); + +MODULE_DESCRIPTION("UDA1341 ALSA soc codec driver"); +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin chripell@evolware.org"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda1341/uda1341.h b/sound/soc/codecs/uda1341/uda1341.h new file mode 100644 index 0000000..5eccfc4 --- /dev/null +++ b/sound/soc/codecs/uda1341/uda1341.h @@ -0,0 +1,47 @@ +/* + * uda1341.h -- UDA1341 ALSA SoC Codec driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _UDA1341_H +#define _UDA1341_H + +#define UDA1341_L3ADDR 5 +#define UDA1341_DATA0_ADDR ((UDA1341_L3ADDR << 2) | 0) +#define UDA1341_DATA1_ADDR ((UDA1341_L3ADDR << 2) | 1) +#define UDA1341_STATUS_ADDR ((UDA1341_L3ADDR << 2) | 2) + +#define UDA1341_EXTADDR_PREFIX 0xC0 +#define UDA1341_EXTDATA_PREFIX 0xE0 + +/* UDA1341 registers */ +#define UDA1341_EA000 0 +#define UDA1341_EA001 1 +#define UDA1341_EA010 2 +#define UDA1341_EA011 3 +#define UDA1341_EA100 4 +#define UDA1341_EA101 5 +#define UDA1341_EA110 6 +#define UDA1341_EA111 7 +#define UDA1341_STATUS0 8 +#define UDA1341_STATUS1 9 +#define UDA1341_DATA000 10 +#define UDA1341_DATA001 11 +#define UDA1341_DATA010 12 +#define UDA1341_DATA1 13 + +#define UDA1341_REGS_NUM 14 + +#define STATUS0_DAIFMT_MASK (~(7<<1)) +#define STATUS0_SYSCLK_MASK (~(3<<4)) + +extern struct snd_soc_dai uda1341_dai; +extern struct snd_soc_codec_device soc_codec_dev_uda1341; + +#endif /* _UDA1341_H */ diff --git a/sound/soc/codecs/uda1341/uda1341_platform_data.h b/sound/soc/codecs/uda1341/uda1341_platform_data.h new file mode 100644 index 0000000..26081cb --- /dev/null +++ b/sound/soc/codecs/uda1341/uda1341_platform_data.h @@ -0,0 +1,18 @@ +#ifndef _UDA1341_PLATFORM_DATA_H_ +#define _UDA1341_PLATFORM_DATA_H_ 1 + +struct uda1341_platform_data { + void (*setdat) (struct uda1341_platform_data *, int); + void (*setclk) (struct uda1341_platform_data *, int); + void (*setmode) (struct uda1341_platform_data *, int); + int data_hold; + int data_setup; + int clock_high; + int mode_hold; + int mode; + int mode_setup; + void *priv; + void (*power) (int); +}; + +#endif diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index b9f2353..cc8fa5e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -16,7 +16,7 @@ config SND_S3C2443_SOC_AC97 tristate select AC97_BUS select SND_SOC_AC97_BUS - + config SND_S3C24XX_SOC_NEO1973_WM8753 tristate "SoC I2S Audio support for NEO1973 - WM8753" depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01 @@ -44,3 +44,8 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650 Say Y if you want to add support for SoC audio on ln2440sbc with the ALC650.
+config SND_S3C24XX_SOC_S3C24XX_UDA1341 + tristate "SoC I2S Audio support UDA1341 wired to a S3C24XX" + depends on SND_S3C24XX_SOC + select SND_S3C24XX_SOC_I2S + select SND_SOC_UDA1341 \ No newline at end of file diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 0aa5fb0..155f5a4 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -13,7 +13,9 @@ obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o +snd-soc-s3c24xx-uda1341-objs := s3c24xx_uda1341.o
obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o +obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA1341) += snd-soc-s3c24xx-uda1341.o diff --git a/sound/soc/s3c24xx/s3c24xx_uda1341.c b/sound/soc/s3c24xx/s3c24xx_uda1341.c new file mode 100644 index 0000000..fc3c13e --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_uda1341.c @@ -0,0 +1,314 @@ +/* + * Modifications by Christian Pellegrin chripell@evolware.org + * + * s3c24xx_uda1341.c -- S3C24XX_UDA1341 ALSA SoC Audio board driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/clk.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/s3c24xx_uda1341.h> + +#include <asm/mach-types.h> +#include <asm/plat-s3c24xx/regs-iis.h> +#include <mach/regs-gpio.h> +#include <mach/regs-gpioj.h> +#include <mach/hardware.h> + +#include "../codecs/uda1341/uda1341.h" +#include "../codecs/uda1341/uda1341_platform_data.h" +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" + +/*#define S3C24XX_UDA1341_DEBUG 1*/ +#ifdef S3C24XX_UDA1341_DEBUG +#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x) +#else +#define DBG(x...) +#endif + +static struct clk *xtal; +static struct clk *pclk; + +static unsigned long s3c24xx_uda1341_calc_error(unsigned long rate, + unsigned long clk_rate, + unsigned int div, + unsigned int fs) +{ + long err; + + err = clk_rate / (div * fs); + err -= rate; + if (err < 0) + err = -err; + DBG("rate %lu clk %lu div %u fs %u err %ld\n", + rate, clk_rate, div, fs, err); + return err; +} + +static int s3c24xx_uda1341_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int clk = 0; + unsigned int div = 0, cdiv; + int ret = 0; + int clk_source, fs_mode; + unsigned long mpllin_rate = clk_get_rate(xtal); + unsigned long pclk_rate = clk_get_rate(pclk); + unsigned long rate = params_rate(params); + unsigned long err, cerr; + + DBG("mpllin %ld pclk %ld rate %lu\n", mpllin_rate, pclk_rate, rate); + + div = pclk_rate / (256 * rate); + if (div == 0) + div = 1; + if (div > 32) + div = 32; + err = s3c24xx_uda1341_calc_error(rate, pclk_rate, div, 256); + fs_mode = S3C2410_IISMOD_256FS; + clk_source = S3C24XX_CLKSRC_PCLK; + + if (div < 32) { + cdiv = div + 1; + cerr = s3c24xx_uda1341_calc_error(rate, pclk_rate, cdiv, 256); + if (cerr < err) { + err = cerr; + div = cdiv; + } + } + + cdiv = pclk_rate / (384 * rate); + if (cdiv == 0) + cdiv = 1; + if (cdiv > 32) + cdiv = 32; + cerr = s3c24xx_uda1341_calc_error(rate, pclk_rate, cdiv, 384); + if (cerr < err) { + err = cerr; + div = cdiv; + fs_mode = S3C2410_IISMOD_384FS; + } + + if (cdiv < 32) { + cdiv = cdiv + 1; + cerr = s3c24xx_uda1341_calc_error(rate, pclk_rate, cdiv, 384); + if (cerr < err) { + err = cerr; + div = cdiv; + fs_mode = S3C2410_IISMOD_384FS; + } + } + + cerr = s3c24xx_uda1341_calc_error(rate, mpllin_rate, 1, 256); + if (cerr < err) { + err = cerr; + fs_mode = S3C2410_IISMOD_256FS; + clk_source = S3C24XX_CLKSRC_MPLL; + } + + cerr = s3c24xx_uda1341_calc_error(rate, mpllin_rate, 1, 384); + if (cerr < err) { + err = cerr; + fs_mode = S3C2410_IISMOD_384FS; + clk_source = S3C24XX_CLKSRC_MPLL; + } + + clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate; + div = div - 1; + DBG("Will use: %s %s %d sysclk %d err %ld\n", + fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", + clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", + div, clk, err); + + ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, clk_source , clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + fs_mode); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops s3c24xx_uda1341_ops = { + .hw_params = s3c24xx_uda1341_hw_params, +}; + +static struct snd_soc_dai_link s3c24xx_uda1341_dai_link = { + .name = "UDA1341", + .stream_name = "UDA1341", + .codec_dai = &uda1341_dai, + .cpu_dai = &s3c24xx_i2s_dai, + .ops = &s3c24xx_uda1341_ops, +}; + +static struct snd_soc_machine snd_soc_machine_s3c24xx_uda1341 = { + .name = "S3C24XX_UDA1341", + .dai_link = &s3c24xx_uda1341_dai_link, + .num_links = 1, +}; + +static struct s3c24xx_uda1341_platform_data *s3c24xx_uda1341_l3_pins; + +static void setdat(struct uda1341_platform_data *p, int v) +{ + s3c2410_gpio_setpin(s3c24xx_uda1341_l3_pins->l3_data, v > 0); + s3c2410_gpio_cfgpin(s3c24xx_uda1341_l3_pins->l3_data, + S3C2410_GPIO_OUTPUT); +} + +static void setclk(struct uda1341_platform_data *p, int v) +{ + s3c2410_gpio_setpin(s3c24xx_uda1341_l3_pins->l3_clk, v > 0); + s3c2410_gpio_cfgpin(s3c24xx_uda1341_l3_pins->l3_clk, + S3C2410_GPIO_OUTPUT); +} + +static void setmode(struct uda1341_platform_data *p, int v) +{ + s3c2410_gpio_setpin(s3c24xx_uda1341_l3_pins->l3_mode, v > 0); + s3c2410_gpio_cfgpin(s3c24xx_uda1341_l3_pins->l3_mode, + S3C2410_GPIO_OUTPUT); +} + +static void s3c24xx_uda1341_power(int en) +{ + if (s3c24xx_uda1341_l3_pins->power) + s3c24xx_uda1341_l3_pins->power(en); +} + +static struct uda1341_platform_data s3c24xx_uda1341 = { + .setdat = setdat, + .setclk = setclk, + .setmode = setmode, + .data_hold = 1, + .data_setup = 1, + .clock_high = 1, + .mode_hold = 1, + .mode = 1, + .mode_setup = 1, + .power = s3c24xx_uda1341_power, +}; + +static struct snd_soc_device s3c24xx_uda1341_snd_devdata = { + .machine = &snd_soc_machine_s3c24xx_uda1341, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_uda1341, + .codec_data = &s3c24xx_uda1341, +}; + +static struct platform_device *s3c24xx_uda1341_snd_device; + +static int s3c24xx_uda1341_probe(struct platform_device *pdev) +{ + int ret; + + printk(KERN_INFO "S3C24XX_UDA1341 SoC Audio driver\n"); + + s3c24xx_uda1341_l3_pins = pdev->dev.platform_data; + if (s3c24xx_uda1341_l3_pins == NULL) { + printk(KERN_ERR "S3C24XX_UDA1341 SoC Audio: " + "unable to find platform data\n"); + return -ENODEV; + } + + s3c24xx_uda1341_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_uda1341_snd_device) { + printk(KERN_ERR "S3C24XX_UDA1341 SoC Audio: " + "Unable to register\n"); + return -ENOMEM; + } + + platform_set_drvdata(s3c24xx_uda1341_snd_device, + &s3c24xx_uda1341_snd_devdata); + s3c24xx_uda1341_snd_devdata.dev = &s3c24xx_uda1341_snd_device->dev; + ret = platform_device_add(s3c24xx_uda1341_snd_device); + + if (ret) { + printk(KERN_ERR "S3C24XX_UDA1341 SoC Audio: Unable to add\n"); + platform_device_put(s3c24xx_uda1341_snd_device); + } + + xtal = clk_get(NULL, "xtal"); + pclk = clk_get(NULL, "pclk"); + + return ret; +} + +static int s3c24xx_uda1341_remove(struct platform_device *pdev) +{ + platform_device_unregister(s3c24xx_uda1341_snd_device); + clk_put(xtal); + clk_put(pclk); + return 0; +} + +static struct platform_driver s3c24xx_uda1341_driver = { + .probe = s3c24xx_uda1341_probe, + .remove = s3c24xx_uda1341_remove, + .driver = { + .name = "s3c24xx_uda1341", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c24xx_uda1341_init(void) +{ + return platform_driver_register(&s3c24xx_uda1341_driver); +} + +static void __exit s3c24xx_uda1341_exit(void) +{ + platform_driver_unregister(&s3c24xx_uda1341_driver); +} + + +module_init(s3c24xx_uda1341_init); +module_exit(s3c24xx_uda1341_exit); + +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin chripell@evolware.org"); +MODULE_DESCRIPTION("S3C24XX_UDA1341 ALSA SoC audio driver"); +MODULE_LICENSE("GPL"); -- 1.4.4.4
On Sat, Nov 08, 2008 at 08:43:55AM +0100, Christian Pellegrin wrote:
This patch adds support for the UDA1341 codec and a sound card definition for one of these UDAs connected to the s3c24xx. It is
First up, thanks for doing this work - it'd be good to see this merged into ASoC.
*heavily* based on the one made by Zoltan Devai but:
Ideally this should be submitted as multiple patches - at least one for the codec and one for the board, probably also one for the l3 interface.
Generated on 20081108 against v2.6.27
Please generate patches against current git - the topic/asoc branch of:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git
In general, once the merge window is past it's best to submit patches against at least -rc1, submitting against the previous release makes it much more likely that your patch will be out of date as has happened here.
+++ b/sound/soc/codecs/Kconfig @@ -50,3 +50,6 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_TLV320AIC3X tristate depends on I2C
+config SND_SOC_UDA1341
- tristate
You should also add this to SND_SOC_ALL_CODECS. There's a bunch of other things like this that have changed between 2.6.27 and the current code - for example, the bias level configuration.
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d7b97ab..cbace60 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,3 +23,4 @@ obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_UDA1341) += uda1341/
There doesn't seem to be much benefit to adding a subdirectory for two source files here.
--- /dev/null +++ b/sound/soc/codecs/uda1341/l3.c
+/*#define L3_DEBUG 1*/ +#ifdef L3_DEBUG +#define DBG(format, arg...) \
- printk(KERN_DEBUG "L3: %s:" format "\n" , __func__, ## arg)
+#else +#define DBG(format, arg...) do {} while (0) +#endif
Please use the standard pr_debug() macro rather than defining custom ones.
+#define setdat(adap, val) (adap->setdat(adap, val)) +#define setclk(adap, val) (adap->setclk(adap, val)) +#define setmode(adap, val) (adap->setmode(adap, val))
These should be inline functions for type safety.
+/*#define UDA1341_DEBUG 1*/ +#ifdef UDA1341_DEBUG +#define DBG(format, arg...) \
- printk(KERN_DEBUG "UDA1341: %s:" format "\n" , __func__, ## arg)
+#else +#define DBG(format, arg...) do {} while (0) +#endif
Please use the standard pr_debug() macro.
+static int uda1341_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
+{
- int ret;
- u8 addr;
- u8 data = value;
- DBG("reg: %02X, value:%02X", reg, value);
- if (reg >= UDA1341_REGS_NUM) {
DBG("Unkown register: reg: %d", reg);return -EINVAL;- }
That should probably print the error unconditionally.
+static int uda1341_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+{
- /* set SYSCLK / fs ratio */
- switch (uda1341->sysclk / params_rate(params)) {
- default:
return -EINVAL;break;- }
No need for the break here. It's probably best to log an explicit error saying why the failure occurred - otherwise it'll be a bit obscure to users what's going on. A similar comment applies to the other error cases.
+static int uda1341_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)+{
- /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
- we'll error out on set_hw_params if it's not OK */
This implies rather more flexibility than actually exists - hw_params() requires an exact multiplier here. You should probably add a note explaining that it's up to the machine driver to make sure that the rate is OK.
+SOC_SINGLE("Channel 1 mixer gain", UDA1341_EA000, 0, 0x1F, 1), +SOC_SINGLE("Channgel 2 mixer gain", UDA1341_EA001, 0, 0x1F, 1),
There's a typo here. Also, this and many of the other control names don't fit in with the ALSA control name spec so won't be displayed correctly by applications. For example, these should be "Volume" rather than "gain", and "switch" should be "Switch". There's documentation of the standard names in:
Documentation/sound/alsa/ControlNames.txt
+SOC_SINGLE("DAC Gain switch", UDA1341_STATUS1, 6, 1, 0), +SOC_SINGLE("ADC Gain switch", UDA1341_STATUS1, 5, 1, 0),
What exactly do these controls do? "Gain switch" sounds wrong - I'd not expect the gain to be a boolean.
+static int uda1341_soc_suspend(struct platform_device *pdev,
pm_message_t state)+{
- struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_codec *codec = socdev->codec;
- struct uda1341_platform_data *pd;
- uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
- pd = (struct uda1341_platform_data *) codec->control_data;
- if (pd->power)
pd->power(0);
I'd expect that dapm_event() (which is now called set_bias_level()) would be handling the power callback too.
+struct snd_soc_codec_device soc_codec_dev_uda1341 = {
- .probe = uda1341_soc_probe,
- .remove = uda1341_soc_remove,
+#if defined(CONFIG_PM)
- .suspend = uda1341_soc_suspend,
- .resume = uda1341_soc_resume,
+#endif /* CONFIG_PM */
The standard idiom for this is to have an else defining the functions to NULL pointers above and then no ifdef here.
- void (*power) (int);
I'd really expect to see this being passed an argument specifying what it's interacting with.
@@ -16,7 +16,7 @@ config SND_S3C2443_SOC_AC97 tristate select AC97_BUS select SND_SOC_AC97_BUS
config SND_S3C24XX_SOC_NEO1973_WM8753 tristate "SoC I2S Audio support for NEO1973 - WM8753" depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01
Random whitespace change here...
--- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_uda1341.c
+/*#define S3C24XX_UDA1341_DEBUG 1*/ +#ifdef S3C24XX_UDA1341_DEBUG +#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x) +#else +#define DBG(x...) +#endif
pr_debug().
+static int s3c24xx_uda1341_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)+{
I can follow this function but it'd be nice to see more comments in here. It looks like you could clarify it by iterating over a table of possible clock sources rather than writing each out in code.
It also looks like this should be imposing constraints which prevent the configuration of different rates for the DAC and ADC - several existing codec drivers like the wm8903 provide examples of doing this.
- ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, clk_source , clk,
SND_SOC_CLOCK_IN);- if (ret < 0)
return ret;
You want to run this through scripts/checkpatch.pl.
+static void setdat(struct uda1341_platform_data *p, int v) +{
- s3c2410_gpio_setpin(s3c24xx_uda1341_l3_pins->l3_data, v > 0);
- s3c2410_gpio_cfgpin(s3c24xx_uda1341_l3_pins->l3_data,
S3C2410_GPIO_OUTPUT);+}
Is there any reason for setting the pins to output mode each time? It looks like you could just configure the mode once at startup and then only set the pin status here.
+static void s3c24xx_uda1341_power(int en) +{
- if (s3c24xx_uda1341_l3_pins->power)
s3c24xx_uda1341_l3_pins->power(en);+}
This feels like it ought to be initialised conditionally rather than having the check for the pin it.
- ret = platform_device_add(s3c24xx_uda1341_snd_device);
- xtal = clk_get(NULL, "xtal");
- pclk = clk_get(NULL, "pclk");
This should be done in the init function for the device and should really check the return value in case it can't get the clock for some reason. Ideally there'd be a dev there, but I'd need to check if the s3c24xx stuff does that.
On Mon, 10 Nov 2008 13:34:52 +0000 Mark Brown broonie@sirena.org.uk wrote:
On Sat, Nov 08, 2008 at 08:43:55AM +0100, Christian Pellegrin wrote:
This patch adds support for the UDA1341 codec and a sound card definition for one of these UDAs connected to the s3c24xx. It is
First up, thanks for doing this work - it'd be good to see this merged into ASoC.
I'm equally interested in this patch, seeing as I got an hp jornada 7xx with uda1344 waiting.
Have you any plans on creating an 134x seeing as there is very little difference between chipsets?
*heavily* based on the one made by Zoltan Devai but:
Ideally this should be submitted as multiple patches - at least one for the codec and one for the board, probably also one for the l3 interface.
Generated on 20081108 against v2.6.27
Please generate patches against current git - the topic/asoc branch of:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git
In general, once the merge window is past it's best to submit patches against at least -rc1, submitting against the previous release makes it much more likely that your patch will be out of date as has happened here.
+++ b/sound/soc/codecs/Kconfig @@ -50,3 +50,6 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_TLV320AIC3X tristate depends on I2C
+config SND_SOC_UDA1341
- tristate
You should also add this to SND_SOC_ALL_CODECS. There's a bunch of other things like this that have changed between 2.6.27 and the current code - for example, the bias level configuration.
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d7b97ab..cbace60 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,3 +23,4 @@ obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_UDA1341) += uda1341/
There doesn't seem to be much benefit to adding a subdirectory for two source files here.
--- /dev/null +++ b/sound/soc/codecs/uda1341/l3.c
+/*#define L3_DEBUG 1*/ +#ifdef L3_DEBUG +#define DBG(format, arg...) \
- printk(KERN_DEBUG "L3: %s:" format "\n" , __func__, ## arg)
+#else +#define DBG(format, arg...) do {} while (0) +#endif
Please use the standard pr_debug() macro rather than defining custom ones.
+#define setdat(adap, val) (adap->setdat(adap, val)) +#define setclk(adap, val) (adap->setclk(adap, val)) +#define setmode(adap, val) (adap->setmode(adap, val))
These should be inline functions for type safety.
+/*#define UDA1341_DEBUG 1*/ +#ifdef UDA1341_DEBUG +#define DBG(format, arg...) \
- printk(KERN_DEBUG "UDA1341: %s:" format "\n" , __func__, ## arg)
+#else +#define DBG(format, arg...) do {} while (0) +#endif
Please use the standard pr_debug() macro.
+static int uda1341_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
+{
- int ret;
- u8 addr;
- u8 data = value;
- DBG("reg: %02X, value:%02X", reg, value);
- if (reg >= UDA1341_REGS_NUM) {
DBG("Unkown register: reg: %d", reg);return -EINVAL;- }
That should probably print the error unconditionally.
+static int uda1341_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+{
- /* set SYSCLK / fs ratio */
- switch (uda1341->sysclk / params_rate(params)) {
- default:
return -EINVAL;break;- }
No need for the break here. It's probably best to log an explicit error saying why the failure occurred - otherwise it'll be a bit obscure to users what's going on. A similar comment applies to the other error cases.
+static int uda1341_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)+{
- /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
- we'll error out on set_hw_params if it's not OK */
This implies rather more flexibility than actually exists - hw_params() requires an exact multiplier here. You should probably add a note explaining that it's up to the machine driver to make sure that the rate is OK.
+SOC_SINGLE("Channel 1 mixer gain", UDA1341_EA000, 0, 0x1F, 1), +SOC_SINGLE("Channgel 2 mixer gain", UDA1341_EA001, 0, 0x1F, 1),
There's a typo here. Also, this and many of the other control names don't fit in with the ALSA control name spec so won't be displayed correctly by applications. For example, these should be "Volume" rather than "gain", and "switch" should be "Switch". There's documentation of the standard names in:
Documentation/sound/alsa/ControlNames.txt
+SOC_SINGLE("DAC Gain switch", UDA1341_STATUS1, 6, 1, 0), +SOC_SINGLE("ADC Gain switch", UDA1341_STATUS1, 5, 1, 0),
What exactly do these controls do? "Gain switch" sounds wrong - I'd not expect the gain to be a boolean.
+static int uda1341_soc_suspend(struct platform_device *pdev,
pm_message_t state)+{
- struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_codec *codec = socdev->codec;
- struct uda1341_platform_data *pd;
- uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
- pd = (struct uda1341_platform_data *) codec->control_data;
- if (pd->power)
pd->power(0);I'd expect that dapm_event() (which is now called set_bias_level()) would be handling the power callback too.
+struct snd_soc_codec_device soc_codec_dev_uda1341 = {
- .probe = uda1341_soc_probe,
- .remove = uda1341_soc_remove,
+#if defined(CONFIG_PM)
- .suspend = uda1341_soc_suspend,
- .resume = uda1341_soc_resume,
+#endif /* CONFIG_PM */
The standard idiom for this is to have an else defining the functions to NULL pointers above and then no ifdef here.
- void (*power) (int);
I'd really expect to see this being passed an argument specifying what it's interacting with.
@@ -16,7 +16,7 @@ config SND_S3C2443_SOC_AC97 tristate select AC97_BUS select SND_SOC_AC97_BUS
config SND_S3C24XX_SOC_NEO1973_WM8753 tristate "SoC I2S Audio support for NEO1973 - WM8753" depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01
Random whitespace change here...
--- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_uda1341.c
+/*#define S3C24XX_UDA1341_DEBUG 1*/ +#ifdef S3C24XX_UDA1341_DEBUG +#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x) +#else +#define DBG(x...) +#endif
pr_debug().
+static int s3c24xx_uda1341_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)+{
I can follow this function but it'd be nice to see more comments in here. It looks like you could clarify it by iterating over a table of possible clock sources rather than writing each out in code.
It also looks like this should be imposing constraints which prevent the configuration of different rates for the DAC and ADC - several existing codec drivers like the wm8903 provide examples of doing this.
- ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, clk_source , clk,
SND_SOC_CLOCK_IN);- if (ret < 0)
return ret;You want to run this through scripts/checkpatch.pl.
+static void setdat(struct uda1341_platform_data *p, int v) +{
- s3c2410_gpio_setpin(s3c24xx_uda1341_l3_pins->l3_data, v > 0);
- s3c2410_gpio_cfgpin(s3c24xx_uda1341_l3_pins->l3_data,
S3C2410_GPIO_OUTPUT);+}
Is there any reason for setting the pins to output mode each time? It looks like you could just configure the mode once at startup and then only set the pin status here.
+static void s3c24xx_uda1341_power(int en) +{
- if (s3c24xx_uda1341_l3_pins->power)
s3c24xx_uda1341_l3_pins->power(en);+}
This feels like it ought to be initialised conditionally rather than having the check for the pin it.
- ret = platform_device_add(s3c24xx_uda1341_snd_device);
- xtal = clk_get(NULL, "xtal");
- pclk = clk_get(NULL, "pclk");
This should be done in the init function for the device and should really check the return value in case it can't get the clock for some reason. Ideally there'd be a dev there, but I'd need to check if the s3c24xx stuff does that.
List admin: http://lists.arm.linux.org.uk/mailman/listinfo/linux-arm-kernel FAQ: http://www.arm.linux.org.uk/mailinglists/faq.php Etiquette: http://www.arm.linux.org.uk/mailinglists/etiquette.php
On Mon, Nov 10, 2008 at 6:58 PM, Kristoffer Ericson kristoffer.ericson@gmail.com wrote:
On Mon, 10 Nov 2008 13:34:52 +0000
I'm equally interested in this patch, seeing as I got an hp jornada 7xx with uda1344 waiting.
Have you any plans on creating an 134x seeing as there is very little difference between chipsets?
Hi,
I had a quick look at the UDA1344 data sheet. When operated in L3-mode it's basically a subset of the UDA1341 so I can do the changes. Anyway I'm not able to test them so could you give me a hand testing? Anyway if you won't use special mixer settings the patch I sent should work since the important configuration registers and the volume are the same for both chips.
On Mon, 10 Nov 2008 20:08:00 +0100 chri chripell@gmail.com wrote:
On Mon, Nov 10, 2008 at 6:58 PM, Kristoffer Ericson kristoffer.ericson@gmail.com wrote:
On Mon, 10 Nov 2008 13:34:52 +0000
I'm equally interested in this patch, seeing as I got an hp jornada 7xx with uda1344 waiting.
Have you any plans on creating an 134x seeing as there is very little difference between chipsets?
Hi,
I had a quick look at the UDA1344 data sheet. When operated in L3-mode it's basically a subset of the UDA1341 so I can do the changes. Anyway I'm not able to test them so could you give me a hand testing? Anyway if you won't use special mixer settings the patch I sent should work since the important configuration registers and the volume are the same for both chips.
It would be good if the driver could be used for both 1341/1344. Some minor ifdef's should sort it out. Please drop me a mail with the patches (missed the old one) and I'll start testing them them out.
-- Christian Pellegrin, see http://www.evolware.org/chri/ "Real Programmers don't play tennis, or any other sport which requires you to change clothes. Mountain climbing is OK, and Real Programmers wear their climbing boots to work in case a mountain should suddenly spring up in the middle of the computer room."
On Mon, Nov 10, 2008 at 09:22:33PM +0100, Kristoffer Ericson wrote:
On Mon, 10 Nov 2008 20:08:00 +0100 chri chripell@gmail.com wrote:
On Mon, Nov 10, 2008 at 6:58 PM, Kristoffer Ericson kristoffer.ericson@gmail.com wrote:
On Mon, 10 Nov 2008 13:34:52 +0000
I'm equally interested in this patch, seeing as I got an hp jornada 7xx with uda1344 waiting.
Have you any plans on creating an 134x seeing as there is very little difference between chipsets?
Hi,
I had a quick look at the UDA1344 data sheet. When operated in L3-mode it's basically a subset of the UDA1341 so I can do the changes. Anyway I'm not able to test them so could you give me a hand testing? Anyway if you won't use special mixer settings the patch I sent should work since the important configuration registers and the volume are the same for both chips.
It would be good if the driver could be used for both 1341/1344. Some minor ifdef's should sort it out. Please drop me a mail with the patches (missed the old one) and I'll start testing them them out.
eww, ifdefs...
-- Christian Pellegrin, see http://www.evolware.org/chri/ "Real Programmers don't play tennis, or any other sport which requires you to change clothes. Mountain climbing is OK, and Real Programmers wear their climbing boots to work in case a mountain should suddenly spring up in the middle of the computer room."
-- Kristoffer Ericson kristoffer.ericson@gmail.com
List admin: http://lists.arm.linux.org.uk/mailman/listinfo/linux-arm-kernel FAQ: http://www.arm.linux.org.uk/mailinglists/faq.php Etiquette: http://www.arm.linux.org.uk/mailinglists/etiquette.php
On Mon, Nov 10, 2008 at 11:10 PM, Ben Dooks ben-linux@fluff.org wrote:
It would be good if the driver could be used for both 1341/1344. Some minor ifdef's should sort it out. Please drop me a mail with the patches (missed the old one) and I'll start testing them them out.
eww, ifdefs...
I'll try to put the common code in a uda_common.c and the do 2 small drivers.
On Tue, 11 Nov 2008 15:15:11 +0100 chri chripell@gmail.com wrote:
On Mon, Nov 10, 2008 at 11:10 PM, Ben Dooks ben-linux@fluff.org wrote:
It would be good if the driver could be used for both 1341/1344. Some minor ifdef's should sort it out. Please drop me a mail with the patches (missed the old one) and I'll start testing them them out.
eww, ifdefs...
I'll try to put the common code in a uda_common.c and the do 2 small drivers.
Yeah sounds resonable. And Ben, ifdefs do serve a purpose, atleast at minimal use :)
-- Christian Pellegrin, see http://www.evolware.org/chri/ "Real Programmers don't play tennis, or any other sport which requires you to change clothes. Mountain climbing is OK, and Real Programmers wear their climbing boots to work in case a mountain should suddenly spring up in the middle of the computer room."
On Tue, 11 Nov 2008 15:15:11 +0100 chri chripell@gmail.com wrote:
On Mon, Nov 10, 2008 at 11:10 PM, Ben Dooks ben-linux@fluff.org wrote:
It would be good if the driver could be used for both 1341/1344. Some minor ifdef's should sort it out. Please drop me a mail with the patches (missed the old one) and I'll start testing them them out.
eww, ifdefs...
Oh, just wanted to ask, is the L3 code platform specific or universally crunchable?
I'll try to put the common code in a uda_common.c and the do 2 small drivers.
-- Christian Pellegrin, see http://www.evolware.org/chri/ "Real Programmers don't play tennis, or any other sport which requires you to change clothes. Mountain climbing is OK, and Real Programmers wear their climbing boots to work in case a mountain should suddenly spring up in the middle of the computer room."
On Tue, Nov 11, 2008 at 9:53 PM, Kristoffer Ericson kristoffer.ericson@gmail.com wrote:
On Tue, 11 Nov 2008 15:15:11 +0100
Oh, just wanted to ask, is the L3 code platform specific or universally crunchable?
In the patch that is coming it will be more reusable.
Hi,
On Mon, Nov 10, 2008 at 2:34 PM, Mark Brown broonie@sirena.org.uk wrote:
Ideally this should be submitted as multiple patches - at least one for the codec and one for the board, probably also one for the l3 interface.
ack
Please generate patches against current git - the topic/asoc branch of:
ack.
You should also add this to SND_SOC_ALL_CODECS. There's a bunch of other things like this that have changed between 2.6.27 and the current code - for example, the bias level configuration.
ack.
There doesn't seem to be much benefit to adding a subdirectory for two source files here.
ack
Please use the standard pr_debug() macro rather than defining custom ones.
ack. I was tempted to use pr_debug (and even better dev_dbg) but I went for the old printk method for uniformity with the rest of the drivers.
These should be inline functions for type safety.
ack
That should probably print the error unconditionally.
ack
switch (uda1341->sysclk / params_rate(params)) {
default:return -EINVAL;break;}No need for the break here. It's probably best to log an explicit error saying why the failure occurred - otherwise it'll be a bit obscure to users what's going on. A similar comment applies to the other error cases.
ack
+static int uda1341_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)+{
/* Anything between 256fs*8Khz and 512fs*48Khz should be acceptablewe'll error out on set_hw_params if it's not OK */This implies rather more flexibility than actually exists - hw_params() requires an exact multiplier here. You should probably add a note explaining that it's up to the machine driver to make sure that the rate is OK.
ack
There's a typo here. Also, this and many of the other control names don't fit in with the ALSA control name spec so won't be displayed correctly by applications. For example, these should be "Volume" rather than "gain", and "switch" should be "Switch". There's documentation of the standard names in:
Documentation/sound/alsa/ControlNames.txt
ack. I have to admit that I didn't check the names. I will read this document and do the changes accordingly
+SOC_SINGLE("DAC Gain switch", UDA1341_STATUS1, 6, 1, 0), +SOC_SINGLE("ADC Gain switch", UDA1341_STATUS1, 5, 1, 0),
What exactly do these controls do? "Gain switch" sounds wrong - I'd not expect the gain to be a boolean.
they turn on/off an amplifier with 6db gain in the record and playback respectively.
pd = (struct uda1341_platform_data *) codec->control_data;if (pd->power)pd->power(0);I'd expect that dapm_event() (which is now called set_bias_level()) would be handling the power callback too.
ack
+struct snd_soc_codec_device soc_codec_dev_uda1341 = {
.probe = uda1341_soc_probe,.remove = uda1341_soc_remove,+#if defined(CONFIG_PM)
.suspend = uda1341_soc_suspend,.resume = uda1341_soc_resume,+#endif /* CONFIG_PM */
The standard idiom for this is to have an else defining the functions to NULL pointers above and then no ifdef here.
ack
void (*power) (int);I'd really expect to see this being passed an argument specifying what it's interacting with.
ack
config SND_S3C24XX_SOC_NEO1973_WM8753 tristate "SoC I2S Audio support for NEO1973 - WM8753" depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01
Random whitespace change here...
ack, sorry for that
+static int s3c24xx_uda1341_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)+{
I can follow this function but it'd be nice to see more comments in here. It looks like you could clarify it by iterating over a table of possible clock sources rather than writing each out in code.
unfortunately the 2 clock sources are handled differently: PCLK can be divided, MPLL_in no. So I don't see much convenience in using a (complicated) table
It also looks like this should be imposing constraints which prevent the configuration of different rates for the DAC and ADC - several existing codec drivers like the wm8903 provide examples of doing this.
ack
ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, clk_source , clk,SND_SOC_CLOCK_IN);if (ret < 0)return ret;You want to run this through scripts/checkpatch.pl.
I already did it
+static void setdat(struct uda1341_platform_data *p, int v) +{
s3c2410_gpio_setpin(s3c24xx_uda1341_l3_pins->l3_data, v > 0);s3c2410_gpio_cfgpin(s3c24xx_uda1341_l3_pins->l3_data,S3C2410_GPIO_OUTPUT);+}
Is there any reason for setting the pins to output mode each time? It looks like you could just configure the mode once at startup and then only set the pin status here.
no, I'll clean this up too.
+static void s3c24xx_uda1341_power(int en) +{
if (s3c24xx_uda1341_l3_pins->power)s3c24xx_uda1341_l3_pins->power(en);+}
This feels like it ought to be initialised conditionally rather than having the check for the pin it.
It's not clear to me what you mean here.
ret = platform_device_add(s3c24xx_uda1341_snd_device);
xtal = clk_get(NULL, "xtal");pclk = clk_get(NULL, "pclk");This should be done in the init function for the device and should really check the return value in case it can't get the clock for some reason. Ideally there'd be a dev there, but I'd need to check if the s3c24xx stuff does that.
I'll looka at some other s3c24xx clock user and do the changes.
Thanks,
On Mon, Nov 10, 2008 at 08:04:49PM +0100, chri wrote:
On Mon, Nov 10, 2008 at 2:34 PM, Mark Brown broonie@sirena.org.uk wrote:
ack. I was tempted to use pr_debug (and even better dev_dbg) but I went for the old printk method for uniformity with the rest of the drivers.
If you work with current git you'll see that the other drivers have been converted to pr_debug().
+SOC_SINGLE("DAC Gain switch", UDA1341_STATUS1, 6, 1, 0), +SOC_SINGLE("ADC Gain switch", UDA1341_STATUS1, 5, 1, 0),
What exactly do these controls do? "Gain switch" sounds wrong - I'd not expect the gain to be a boolean.
they turn on/off an amplifier with 6db gain in the record and playback respectively.
Should be "ADC +6dB Switch" or similar then.
+static int s3c24xx_uda1341_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)+{
I can follow this function but it'd be nice to see more comments in here. It looks like you could clarify it by iterating over a table of possible clock sources rather than writing each out in code.
unfortunately the 2 clock sources are handled differently: PCLK can be divided, MPLL_in no. So I don't see much convenience in using a (complicated) table
A pointer to a table of dividers would handle it. In any case, it needs more comments.
ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, clk_source , clk,SND_SOC_CLOCK_IN);if (ret < 0)return ret;
You want to run this through scripts/checkpatch.pl.
I already did it
I'd have expected checkpatch to catch the extra space you've got after clk_source there...
+static void s3c24xx_uda1341_power(int en) +{
if (s3c24xx_uda1341_l3_pins->power)s3c24xx_uda1341_l3_pins->power(en);+}
This feels like it ought to be initialised conditionally rather than having the check for the pin it.
It's not clear to me what you mean here.
This function is going to be used in a vtable where IIRC it can be optional but it needs to check the value of power in s3c24xx_uda1341_l3_pins. It feels like you've got too many layers of indirection here and that, for example, the l3_pins power should be assignable directly to the vtable.
On Tue, Nov 11, 2008 at 11:39 AM, Mark Brown broonie@sirena.org.uk wrote:
If you work with current git you'll see that the other drivers have been converted to pr_debug().
I see, I'm cloning git repository *now*.
they turn on/off an amplifier with 6db gain in the record and playback respectively.
Should be "ADC +6dB Switch" or similar then.
ack
A pointer to a table of dividers would handle it. In any case, it needs more comments.
ack,
You want to run this through scripts/checkpatch.pl.
I already did it
I'd have expected checkpatch to catch the extra space you've got after clk_source there...
I agree
This function is going to be used in a vtable where IIRC it can be optional but it needs to check the value of power in s3c24xx_uda1341_l3_pins. It feels like you've got too many layers of indirection here and that, for example, the l3_pins power should be assignable directly to the vtable.
ack
Thanks,
On Mon, Nov 10, 2008 at 01:34:52PM +0000, Mark Brown wrote:
- xtal = clk_get(NULL, "xtal");
- pclk = clk_get(NULL, "pclk");
This should be done in the init function for the device and should really check the return value in case it can't get the clock for some reason. Ideally there'd be a dev there, but I'd need to check if the s3c24xx stuff does that.
And should not be passing a NULL device to clk_get().
On Mon, Nov 10, 2008 at 8:22 PM, Russell King - ARM Linux linux@arm.linux.org.uk wrote:
On Mon, Nov 10, 2008 at 01:34:52PM +0000, Mark Brown wrote:
- xtal = clk_get(NULL, "xtal");
- pclk = clk_get(NULL, "pclk");
This should be done in the init function for the device and should really check the return value in case it can't get the clock for some reason. Ideally there'd be a dev there, but I'd need to check if the s3c24xx stuff does that.
And should not be passing a NULL device to clk_get().
ack. I just picked the *wrong* example driver as a template :-/
On Sat, Nov 08, 2008 at 08:43:55AM +0100, Christian Pellegrin wrote:
This patch adds support for the UDA1341 codec and a sound card definition for one of these UDAs connected to the s3c24xx. It is *heavily* based on the one made by Zoltan Devai but:
1 since the UDA is the only use of the L3 protocol I can find I just embedded a stripped-down L3 bit-banging algorithm from the original RMK work. It is really small.
2 the driver has the possibility to specify the pins used by codec via platform data so it can work on SMDK2410, SMDK2440 or any custom design.
And, iirc, the h1940 ipaq.
3 it tries to guess the right clock source and divider so it is not tied to a particular crystal used.
4 it fixes some bugs.
Thank you for reviews/comments.
Generated on 20081108 against v2.6.27
Signed-off-by: Christian Pellegrin chripell@fsfe.org
include/sound/s3c24xx_uda1341.h | 41 ++ sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/uda1341/Makefile | 3 + sound/soc/codecs/uda1341/l3.c | 106 +++++ sound/soc/codecs/uda1341/l3.h | 8 + sound/soc/codecs/uda1341/uda1341.c | 537 ++++++++++++++++++++++ sound/soc/codecs/uda1341/uda1341.h | 47 ++ sound/soc/codecs/uda1341/uda1341_platform_data.h | 18 + sound/soc/s3c24xx/Kconfig | 7 +- sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/s3c24xx_uda1341.c | 314 +++++++++++++ 12 files changed, 1086 insertions(+), 1 deletions(-)
diff --git a/include/sound/s3c24xx_uda1341.h b/include/sound/s3c24xx_uda1341.h new file mode 100644 index 0000000..bd80bd3 --- /dev/null +++ b/include/sound/s3c24xx_uda1341.h @@ -0,0 +1,41 @@ +#ifndef _S3C24XX_UDA1341_H_ +#define _S3C24XX_UDA1341_H_ 1
+/*
+Example usage (pins for SMDK2410). Put something like this in your +machine file:
+...
+static struct s3c24xx_uda1341_platform_data s3c24xx_uda1341_data = {
- .l3_clk = S3C2410_GPB4,
- .l3_data = S3C2410_GPB3,
- .l3_mode = S3C2410_GPB2,
+};
+static struct platform_device s3c24xx_uda1341 = {
- .name = "s3c24xx_uda1341",
- .dev = {
.platform_data = &s3c24xx_uda1341_data,- }
+};
+...
+static struct platform_device *smdk2410_devices[] __initdata = { +...
- &s3c24xx_uda1341,
+... +};
- */
example into a doc file please, or point at an implementation of this.
+struct s3c24xx_uda1341_platform_data {
- int l3_clk;
- int l3_mode;
- int l3_data;
- void (*power) (int);
+};
+#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1db04a2..4b483b7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -50,3 +50,6 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_TLV320AIC3X tristate depends on I2C
+config SND_SOC_UDA1341
- tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d7b97ab..cbace60 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,3 +23,4 @@ obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_UDA1341) += uda1341/ diff --git a/sound/soc/codecs/uda1341/Makefile b/sound/soc/codecs/uda1341/Makefile new file mode 100644 index 0000000..f9869ba --- /dev/null +++ b/sound/soc/codecs/uda1341/Makefile @@ -0,0 +1,3 @@ +snd-soc-uda1341-objs := l3.o uda1341.o
+obj-$(CONFIG_SND_SOC_UDA1341) += snd-soc-uda1341.o \ No newline at end of file diff --git a/sound/soc/codecs/uda1341/l3.c b/sound/soc/codecs/uda1341/l3.c new file mode 100644 index 0000000..b65b352 --- /dev/null +++ b/sound/soc/codecs/uda1341/l3.c @@ -0,0 +1,106 @@ +/*
- L3 code
- Copyright (C) 2008, Christian Pellegrin chripell@evolware.org
- This program is free software; you can redistribute it and/or modify
- it under the terms of the GNU General Public License version 2 as
- published by the Free Software Foundation.
- based on:
- L3 bus algorithm module.
- Copyright (C) 2001 Russell King, All Rights Reserved.
- */
+#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/init.h> +#include <linux/errno.h> +#include <linux/sched.h>
+#include "l3.h"
+/*#define L3_DEBUG 1*/ +#ifdef L3_DEBUG +#define DBG(format, arg...) \
- printk(KERN_DEBUG "L3: %s:" format "\n" , __func__, ## arg)
+#else +#define DBG(format, arg...) do {} while (0) +#endif
+#define setdat(adap, val) (adap->setdat(adap, val)) +#define setclk(adap, val) (adap->setclk(adap, val)) +#define setmode(adap, val) (adap->setmode(adap, val))
inline functions are better than #define.
+/*
- Send one byte of data to the chip. Data is latched into the chip on
- the rising edge of the clock.
- */
+static void sendbyte(struct uda1341_platform_data *adap, unsigned int byte) +{
- int i;
- DBG("%02x", byte);
- for (i = 0; i < 8; i++) {
setclk(adap, 0);udelay(adap->data_hold);setdat(adap, byte & 1);udelay(adap->data_setup);setclk(adap, 1);udelay(adap->clock_high);byte >>= 1;- }
+}
+/*
- Send a set of bytes to the chip. We need to pulse the MODE line
- between each byte, but never at the start nor at the end of the
- transfer.
- */
+static void sendbytes(struct uda1341_platform_data *adap, const u8 *buf,
int len)+{
- int i;
- for (i = 0; i < len; i++) {
if (i) {udelay(adap->mode_hold);setmode(adap, 0);udelay(adap->mode);}setmode(adap, 1);udelay(adap->mode_setup);sendbyte(adap, buf[i]);- }
+}
+int l3_write(struct uda1341_platform_data *adap, u8 addr, u8 *data, int len) +{
- setclk(adap, 1);
- setdat(adap, 1);
- setmode(adap, 1);
- udelay(adap->mode);
- setmode(adap, 0);
- udelay(adap->mode_setup);
- sendbyte(adap, addr);
- udelay(adap->mode_hold);
- sendbytes(adap, data, len);
- setclk(adap, 1);
- setdat(adap, 1);
- setmode(adap, 0);
- return len;
+}
diff --git a/sound/soc/codecs/uda1341/l3.h b/sound/soc/codecs/uda1341/l3.h new file mode 100644 index 0000000..2c31e0c --- /dev/null +++ b/sound/soc/codecs/uda1341/l3.h @@ -0,0 +1,8 @@ +#ifndef _L3_H_ +#define _L3_H_ 1
+#include "uda1341_platform_data.h"
please find as better name for that file.
+int l3_write(struct uda1341_platform_data *adap, u8 addr, u8 *data, int len);
+#endif diff --git a/sound/soc/codecs/uda1341/uda1341.c b/sound/soc/codecs/uda1341/uda1341.c new file mode 100644 index 0000000..93cadf0 --- /dev/null +++ b/sound/soc/codecs/uda1341/uda1341.c @@ -0,0 +1,537 @@ +/*
- uda1341.c -- UDA1341 ALSA SoC Codec driver
- Modifications by Christian Pellegrin chripell@evolware.org
- Copyright 2007 Dension Audio Systems Ltd.
- Author: Zoltan Devai
- Based on the WM87xx drivers by Liam Girdwood and Richard Purdie
- This program is free software; you can redistribute it and/or modify
- it under the terms of the GNU General Public License version 2 as
- published by the Free Software Foundation.
- */
+#include <linux/module.h> +#include <linux/delay.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h>
+#include "uda1341.h" +#include "l3.h"
+/*#define UDA1341_DEBUG 1*/ +#ifdef UDA1341_DEBUG +#define DBG(format, arg...) \
- printk(KERN_DEBUG "UDA1341: %s:" format "\n" , __func__, ## arg)
+#else +#define DBG(format, arg...) do {} while (0) +#endif
+#define UDA1341_RATES SNDRV_PCM_RATE_8000_48000 +#define UDA1341_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE)+struct uda1341_priv {
- int sysclk;
- int dai_fmt;
+};
+/* In-data addresses are hard-coded into the reg-cache values */ +static const char uda1341_reg[UDA1341_REGS_NUM] = {
- /* Extended address registers */
- 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00,
- /* Status, data regs */
- 0x00, 0x83, 0x00, 0x40, 0x80, 0x00,
+};
+/*
- The codec has no support for reading its registers except for peak level...
- */
+static inline unsigned int uda1341_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
+{
- u8 *cache = codec->reg_cache;
- if (reg >= UDA1341_REGS_NUM)
return -1;- return cache[reg];
+}
+/*
- Write the register cache
- */
+static inline void uda1341_write_reg_cache(struct snd_soc_codec *codec,
- u8 reg, unsigned int value)
+{
- u8 *cache = codec->reg_cache;
- if (reg >= UDA1341_REGS_NUM)
return;- cache[reg] = value;
+}
+/*
- Write to the uda1341 registers
- */
+static int uda1341_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
+{
- int ret;
- u8 addr;
- u8 data = value;
- DBG("reg: %02X, value:%02X", reg, value);
- if (reg >= UDA1341_REGS_NUM) {
DBG("Unkown register: reg: %d", reg);return -EINVAL;- }
- uda1341_write_reg_cache(codec, reg, value);
- switch (reg) {
- case UDA1341_STATUS0:
- case UDA1341_STATUS1:
addr = UDA1341_STATUS_ADDR;break;- case UDA1341_DATA000:
- case UDA1341_DATA001:
- case UDA1341_DATA010:
addr = UDA1341_DATA0_ADDR;break;- case UDA1341_DATA1:
addr = UDA1341_DATA1_ADDR;break;- default:
/* It's an extended address register */addr = (reg | UDA1341_EXTADDR_PREFIX);ret = l3_write((struct uda1341_platform_data *)codec->control_data,UDA1341_DATA0_ADDR, &addr, 1);if (ret != 1)return -EIO;addr = UDA1341_DATA0_ADDR;data = (value | UDA1341_EXTDATA_PREFIX);break;- }
- ret = l3_write((struct uda1341_platform_data *) codec->control_data,
addr, &data, 1);- if (ret != 1)
return -EIO;- return 0;
+}
+static inline void uda1341_reset(struct snd_soc_codec *codec) +{
- u8 reset_reg = uda1341_read_reg_cache(codec, UDA1341_STATUS0);
- uda1341_write(codec, UDA1341_STATUS0, reset_reg | (1<<6));
- mdelay(1);
- uda1341_write(codec, UDA1341_STATUS0, reset_reg & ~(1<<6));
+}
do you really want to be busy-waiting the cpu for that long? how about msleep?
+static int uda1341_mute(struct snd_soc_dai *dai, int mute) +{
- struct snd_soc_codec *codec = dai->codec;
- u8 mute_reg = uda1341_read_reg_cache(codec, UDA1341_DATA010);
- DBG("mute: %d", mute);
- if (mute)
mute_reg |= (1<<2);- else
mute_reg &= ~(1<<2);- uda1341_write(codec, UDA1341_DATA010, mute_reg & ~(1<<2));
- return 0;
+}
+static int uda1341_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_device *socdev = rtd->socdev;
- struct snd_soc_codec *codec = socdev->codec;
- struct uda1341_priv *uda1341 = codec->private_data;
- u8 hw_params = uda1341_read_reg_cache(codec, UDA1341_STATUS0);
- hw_params &= STATUS0_SYSCLK_MASK;
- hw_params &= STATUS0_DAIFMT_MASK;
- DBG("sysclk: %d, rate:%d", uda1341->sysclk, params_rate(params));
- /* set SYSCLK / fs ratio */
- switch (uda1341->sysclk / params_rate(params)) {
- case 512:
break;- case 384:
hw_params |= (1<<4);break;- case 256:
hw_params |= (1<<5);break;- default:
return -EINVAL;break;- }
- DBG("dai_fmt: %d, params_format:%d", uda1341->dai_fmt,
params_format(params));- /* set DAI format and word length */
- switch (uda1341->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
- case SND_SOC_DAIFMT_I2S:
break;- case SND_SOC_DAIFMT_RIGHT_J:
switch (params_format(params)) {case SNDRV_PCM_FORMAT_S16_LE:hw_params |= (1<<1);break;case SNDRV_PCM_FORMAT_S18_3LE:hw_params |= (1<<2);break;case SNDRV_PCM_FORMAT_S20_3LE:hw_params |= ((1<<2) | (1<<1));break;default:return -EINVAL;}break;- case SND_SOC_DAIFMT_LEFT_J:
hw_params |= (1<<3);break;- default:
return -EINVAL;- }
- uda1341_write(codec, UDA1341_STATUS0, hw_params);
- return 0;
+}
+static int uda1341_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 uda1341_priv *uda1341 = codec->private_data;
- DBG("clk_id: %d, freq: %d, dir: %d", clk_id, freq, dir);
- /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
- we'll error out on set_hw_params if it's not OK */
- if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) {
uda1341->sysclk = freq;return 0;- }
- return -EINVAL;
+}
+static int uda1341_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)+{
- struct snd_soc_codec *codec = codec_dai->codec;
- struct uda1341_priv *uda1341 = codec->private_data;
- DBG("fmt: %08X", fmt);
- /* codec supports only full slave mode */
- if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
return -EINVAL;- /* no support for clock inversion */
- if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF)
return -EINVAL;- /* We can't setup DAI format here as it depends on the word bit num */
- /* so let's just store the value for later */
- uda1341->dai_fmt = fmt;
- return 0;
+}
+static int uda1341_dapm_event(struct snd_soc_codec *codec, int event) +{
- u8 reg;
- DBG("event: %08X", event);
- switch (event) {
- case SNDRV_CTL_POWER_D0: /* full On */
/* ADC, DAC on */reg = uda1341_read_reg_cache(codec, UDA1341_STATUS1);uda1341_write(codec, UDA1341_STATUS1, reg | 0x03);break;- case SNDRV_CTL_POWER_D1: /* partial On */
- case SNDRV_CTL_POWER_D2: /* partial On */
- case SNDRV_CTL_POWER_D3hot: /* Off, with power */
break;- case SNDRV_CTL_POWER_D3cold: /* Off, without power */
/* mute, ADC, DAC power off */reg = uda1341_read_reg_cache(codec, UDA1341_STATUS1);uda1341_write(codec, UDA1341_STATUS1, reg & ~(0x03));break;- }
- return 0;
+}
+static const char *uda1341_dsp_setting[] = {"Flat", "Minimum1",
"Minimum2", "Maximum"};+static const char *uda1341_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *uda1341_mixmode[] = {"DD", "Input 2",
"Input 2", "Digital mixer"};+static const struct soc_enum uda1341_mixer_enum[] = { +SOC_ENUM_SINGLE(UDA1341_DATA010, 0, 0x03, uda1341_dsp_setting), +SOC_ENUM_SINGLE(UDA1341_DATA010, 3, 0x03, uda1341_deemph), +SOC_ENUM_SINGLE(UDA1341_EA010, 0, 0x03, uda1341_mixmode), +};
+static const struct snd_kcontrol_new uda1341_snd_controls[] = { +SOC_SINGLE("Playback Volume", UDA1341_DATA000, 0, 0x3F, 1), +SOC_SINGLE("Mic gain", UDA1341_EA010, 2, 0x07, 0), +SOC_SINGLE("Channel 1 mixer gain", UDA1341_EA000, 0, 0x1F, 1), +SOC_SINGLE("Channgel 2 mixer gain", UDA1341_EA001, 0, 0x1F, 1), +SOC_SINGLE("Input channel 2 amp gain", UDA1341_EA101, 0, 0x1F, 0),
+SOC_SINGLE("Bass boost", UDA1341_DATA001, 2, 0xF, 0), +SOC_SINGLE("Treble", UDA1341_DATA001, 0, 3, 0),
+SOC_ENUM("DSP setting", uda1341_mixer_enum[0]), +SOC_ENUM("Playback De-emphasis", uda1341_mixer_enum[1]), +SOC_ENUM("Mixer mode", uda1341_mixer_enum[2]),
+/* This should be an ext control with own handler, if one wants
to set the values in 0.5dB steps instead of 3dB */+SOC_SINGLE("AGC output level", UDA1341_EA110, 0, 0x03, 1), +SOC_SINGLE("AGC time const", UDA1341_EA110, 2, 0x07, 0),
+SOC_SINGLE("DAC Gain switch", UDA1341_STATUS1, 6, 1, 0), +SOC_SINGLE("ADC Gain switch", UDA1341_STATUS1, 5, 1, 0), +SOC_SINGLE("ADC Polarity switch", UDA1341_STATUS1, 4, 1, 0), +SOC_SINGLE("DAC Polarity switch", UDA1341_STATUS1, 3, 1, 0), +SOC_SINGLE("Double speed playback switch", UDA1341_STATUS1, 2, 1, 0), +};
+static int uda1341_add_controls(struct snd_soc_codec *codec) +{
- int err, i;
- for (i = 0; i < ARRAY_SIZE(uda1341_snd_controls); i++) {
err = snd_ctl_add(codec->card,snd_soc_cnew(&uda1341_snd_controls[i],codec, NULL));if (err < 0)return err;- }
- return 0;
+}
+struct snd_soc_dai uda1341_dai = {
- .name = "UDA1341",
- /* playback capabilities */
- .playback = {
.stream_name = "Playback",.channels_min = 1,.channels_max = 2,.rates = UDA1341_RATES,.formats = UDA1341_FORMATS,- },
- /* capture capabilities */
- .capture = {
.stream_name = "Capture",.channels_min = 1,.channels_max = 2,.rates = UDA1341_RATES,.formats = UDA1341_FORMATS,- },
- /* pcm operations */
- .ops = {
.hw_params = uda1341_hw_params,- },
- /* DAI operations */
- .dai_ops = {
.digital_mute = uda1341_mute,.set_sysclk = uda1341_set_dai_sysclk,.set_fmt = uda1341_set_dai_fmt,- }
+}; +EXPORT_SYMBOL(uda1341_dai);
+static int uda1341_soc_probe(struct platform_device *pdev) +{
- struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_codec *codec;
- struct uda1341_priv *uda1341;
- void *codec_setup_data = socdev->codec_data;
- int ret = -ENOMEM;
- struct uda1341_platform_data *pd;
- printk(KERN_INFO "UDA1341 SoC Audio Codec\n");
- socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
- if (socdev->codec == NULL)
return ret;- codec = socdev->codec;
- uda1341 = kzalloc(sizeof(struct uda1341_priv), GFP_KERNEL);
- if (uda1341 == NULL)
goto priv_err;- codec->private_data = uda1341;
- codec->reg_cache = kmemdup(uda1341_reg, sizeof(uda1341_reg),
GFP_KERNEL);- if (codec->reg_cache == NULL)
goto reg_err;- mutex_init(&codec->mutex);
- codec->reg_cache_size = sizeof(uda1341_reg);
- codec->reg_cache_step = 1;
- codec->name = "UDA1341";
- codec->owner = THIS_MODULE;
- codec->dai = &uda1341_dai;
- codec->num_dai = 1;
- codec->read = uda1341_read_reg_cache;
- codec->write = uda1341_write;
- INIT_LIST_HEAD(&codec->dapm_widgets);
- INIT_LIST_HEAD(&codec->dapm_paths);
- if (!codec_setup_data) {
printk(KERN_ERR "UDA1341 SoC codec: ""missing L3 bitbang function\n");ret = -ENODEV;goto pcm_err;- }
- codec->control_data = codec_setup_data;
- pd = (struct uda1341_platform_data *) codec_setup_data;
- if (pd->power)
pd->power(1);- uda1341_reset(codec);
- /* register pcms */
- ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
- if (ret < 0) {
printk(KERN_ERR "UDA1341: failed to register pcms\n");goto pcm_err;- }
- ret = uda1341_add_controls(codec);
- if (ret < 0) {
printk(KERN_ERR "UDA1341: failed to register controls\n");goto pcm_err;- }
- ret = snd_soc_register_card(socdev);
- if (ret < 0) {
printk(KERN_ERR "UDA1341: failed to register card\n");goto card_err;- }
- return 0;
+card_err:
- snd_soc_free_pcms(socdev);
- snd_soc_dapm_free(socdev);
+pcm_err:
- kfree(codec->reg_cache);
+reg_err:
- kfree(codec->private_data);
+priv_err:
- kfree(codec);
- return ret;
+}
+/* power down chip */ +static int uda1341_soc_remove(struct platform_device *pdev) +{
- struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_codec *codec = socdev->codec;
- struct uda1341_platform_data *pd;
- uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
- pd = (struct uda1341_platform_data *) codec->control_data;
- if (pd->power)
pd->power(0);- snd_soc_free_pcms(socdev);
- snd_soc_dapm_free(socdev);
- kfree(codec->private_data);
- kfree(codec->reg_cache);
- kfree(codec);
- return 0;
+}
+#if defined(CONFIG_PM) +static int uda1341_soc_suspend(struct platform_device *pdev,
pm_message_t state)+{
- struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_codec *codec = socdev->codec;
- struct uda1341_platform_data *pd;
- uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
- pd = (struct uda1341_platform_data *) codec->control_data;
- if (pd->power)
pd->power(0);- return 0;
+}
+static int uda1341_soc_resume(struct platform_device *pdev) +{
- struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_codec *codec = socdev->codec;
- struct uda1341_platform_data *pd;
- int i;
- u8 *cache = codec->reg_cache;
- pd = (struct uda1341_platform_data *) codec->control_data;
- if (pd->power)
pd->power(1);- /* Sync reg_cache with the hardware */
- for (i = 0; i < ARRAY_SIZE(uda1341_reg); i++)
codec->write(codec, i, *cache++);- uda1341_dapm_event(codec, SNDRV_CTL_POWER_D0);
- return 0;
+} +#endif /* CONFIG_PM */
+struct snd_soc_codec_device soc_codec_dev_uda1341 = {
- .probe = uda1341_soc_probe,
- .remove = uda1341_soc_remove,
+#if defined(CONFIG_PM)
- .suspend = uda1341_soc_suspend,
- .resume = uda1341_soc_resume,
+#endif /* CONFIG_PM */ +}; +EXPORT_SYMBOL(soc_codec_dev_uda1341);
+MODULE_DESCRIPTION("UDA1341 ALSA soc codec driver"); +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin chripell@evolware.org"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda1341/uda1341.h b/sound/soc/codecs/uda1341/uda1341.h new file mode 100644 index 0000000..5eccfc4 --- /dev/null +++ b/sound/soc/codecs/uda1341/uda1341.h @@ -0,0 +1,47 @@ +/*
- uda1341.h -- UDA1341 ALSA SoC Codec driver
- Copyright 2007 Dension Audio Systems Ltd.
- Author: Zoltan Devai
- This program is free software; you can redistribute it and/or modify
- it under the terms of the GNU General Public License version 2 as
- published by the Free Software Foundation.
- */
+#ifndef _UDA1341_H +#define _UDA1341_H
+#define UDA1341_L3ADDR 5 +#define UDA1341_DATA0_ADDR ((UDA1341_L3ADDR << 2) | 0) +#define UDA1341_DATA1_ADDR ((UDA1341_L3ADDR << 2) | 1) +#define UDA1341_STATUS_ADDR ((UDA1341_L3ADDR << 2) | 2)
+#define UDA1341_EXTADDR_PREFIX 0xC0 +#define UDA1341_EXTDATA_PREFIX 0xE0
+/* UDA1341 registers */ +#define UDA1341_EA000 0 +#define UDA1341_EA001 1 +#define UDA1341_EA010 2 +#define UDA1341_EA011 3 +#define UDA1341_EA100 4 +#define UDA1341_EA101 5 +#define UDA1341_EA110 6 +#define UDA1341_EA111 7 +#define UDA1341_STATUS0 8 +#define UDA1341_STATUS1 9 +#define UDA1341_DATA000 10 +#define UDA1341_DATA001 11 +#define UDA1341_DATA010 12 +#define UDA1341_DATA1 13
+#define UDA1341_REGS_NUM 14
+#define STATUS0_DAIFMT_MASK (~(7<<1)) +#define STATUS0_SYSCLK_MASK (~(3<<4))
+extern struct snd_soc_dai uda1341_dai; +extern struct snd_soc_codec_device soc_codec_dev_uda1341;
+#endif /* _UDA1341_H */ diff --git a/sound/soc/codecs/uda1341/uda1341_platform_data.h b/sound/soc/codecs/uda1341/uda1341_platform_data.h new file mode 100644 index 0000000..26081cb --- /dev/null +++ b/sound/soc/codecs/uda1341/uda1341_platform_data.h @@ -0,0 +1,18 @@ +#ifndef _UDA1341_PLATFORM_DATA_H_ +#define _UDA1341_PLATFORM_DATA_H_ 1
+struct uda1341_platform_data {
- void (*setdat) (struct uda1341_platform_data *, int);
- void (*setclk) (struct uda1341_platform_data *, int);
- void (*setmode) (struct uda1341_platform_data *, int);
no spaces ^
- int data_hold;
- int data_setup;
- int clock_high;
- int mode_hold;
- int mode;
- int mode_setup;
- void *priv;
- void (*power) (int);
+};
+#endif diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index b9f2353..cc8fa5e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -16,7 +16,7 @@ config SND_S3C2443_SOC_AC97 tristate select AC97_BUS select SND_SOC_AC97_BUS
config SND_S3C24XX_SOC_NEO1973_WM8753 tristate "SoC I2S Audio support for NEO1973 - WM8753" depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01 @@ -44,3 +44,8 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650 Say Y if you want to add support for SoC audio on ln2440sbc with the ALC650.
+config SND_S3C24XX_SOC_S3C24XX_UDA1341
tristate "SoC I2S Audio support UDA1341 wired to a S3C24XX"depends on SND_S3C24XX_SOCselect SND_S3C24XX_SOC_I2Sselect SND_SOC_UDA1341\ No newline at end of file diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 0aa5fb0..155f5a4 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -13,7 +13,9 @@ obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o +snd-soc-s3c24xx-uda1341-objs := s3c24xx_uda1341.o
obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o +obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA1341) += snd-soc-s3c24xx-uda1341.o diff --git a/sound/soc/s3c24xx/s3c24xx_uda1341.c b/sound/soc/s3c24xx/s3c24xx_uda1341.c new file mode 100644 index 0000000..fc3c13e --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_uda1341.c @@ -0,0 +1,314 @@ +/*
- Modifications by Christian Pellegrin chripell@evolware.org
- s3c24xx_uda1341.c -- S3C24XX_UDA1341 ALSA SoC Audio board driver
- Copyright 2007 Dension Audio Systems Ltd.
- Author: Zoltan Devai
- This program is free software; you can redistribute it and/or modify
- it under the terms of the GNU General Public License version 2 as
- published by the Free Software Foundation.
- */
+#include <linux/module.h> +#include <linux/clk.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/s3c24xx_uda1341.h>
+#include <asm/mach-types.h> +#include <asm/plat-s3c24xx/regs-iis.h> +#include <mach/regs-gpio.h> +#include <mach/regs-gpioj.h>
nope, don't include these as noted above.
+#include <mach/hardware.h>
+#include "../codecs/uda1341/uda1341.h" +#include "../codecs/uda1341/uda1341_platform_data.h"
find somewhere for these to live that can be included without resorting to knowing the directory layout.
+#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h"
+/*#define S3C24XX_UDA1341_DEBUG 1*/ +#ifdef S3C24XX_UDA1341_DEBUG +#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x) +#else +#define DBG(x...) +#endif
+static struct clk *xtal; +static struct clk *pclk;
+static unsigned long s3c24xx_uda1341_calc_error(unsigned long rate,
unsigned long clk_rate,unsigned int div,unsigned int fs)+{
- long err;
- err = clk_rate / (div * fs);
- err -= rate;
- if (err < 0)
err = -err;- DBG("rate %lu clk %lu div %u fs %u err %ld\n",
rate, clk_rate, div, fs, err);- return err;
+}
+static int s3c24xx_uda1341_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)+{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
- unsigned int clk = 0;
- unsigned int div = 0, cdiv;
- int ret = 0;
- int clk_source, fs_mode;
- unsigned long mpllin_rate = clk_get_rate(xtal);
- unsigned long pclk_rate = clk_get_rate(pclk);
- unsigned long rate = params_rate(params);
- unsigned long err, cerr;
- DBG("mpllin %ld pclk %ld rate %lu\n", mpllin_rate, pclk_rate, rate);
- div = pclk_rate / (256 * rate);
- if (div == 0)
div = 1;- if (div > 32)
div = 32;- err = s3c24xx_uda1341_calc_error(rate, pclk_rate, div, 256);
- fs_mode = S3C2410_IISMOD_256FS;
- clk_source = S3C24XX_CLKSRC_PCLK;
- if (div < 32) {
cdiv = div + 1;cerr = s3c24xx_uda1341_calc_error(rate, pclk_rate, cdiv, 256);if (cerr < err) {err = cerr;div = cdiv;}- }
- cdiv = pclk_rate / (384 * rate);
- if (cdiv == 0)
cdiv = 1;- if (cdiv > 32)
cdiv = 32;- cerr = s3c24xx_uda1341_calc_error(rate, pclk_rate, cdiv, 384);
- if (cerr < err) {
err = cerr;div = cdiv;fs_mode = S3C2410_IISMOD_384FS;- }
- if (cdiv < 32) {
cdiv = cdiv + 1;cerr = s3c24xx_uda1341_calc_error(rate, pclk_rate, cdiv, 384);if (cerr < err) {err = cerr;div = cdiv;fs_mode = S3C2410_IISMOD_384FS;}- }
- cerr = s3c24xx_uda1341_calc_error(rate, mpllin_rate, 1, 256);
- if (cerr < err) {
err = cerr;fs_mode = S3C2410_IISMOD_256FS;clk_source = S3C24XX_CLKSRC_MPLL;- }
- cerr = s3c24xx_uda1341_calc_error(rate, mpllin_rate, 1, 384);
- if (cerr < err) {
err = cerr;fs_mode = S3C2410_IISMOD_384FS;clk_source = S3C24XX_CLKSRC_MPLL;- }
- clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate;
- div = div - 1;
- DBG("Will use: %s %s %d sysclk %d err %ld\n",
fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS",clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK",div, clk, err);- ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);- if (ret < 0)
return ret;- ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);- if (ret < 0)
return ret;- ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, clk_source , clk,
SND_SOC_CLOCK_IN);- if (ret < 0)
return ret;- ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
fs_mode);- if (ret < 0)
return ret;- ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
S3C2410_IISMOD_32FS);- if (ret < 0)
return ret;- ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
S3C24XX_PRESCALE(div, div));- if (ret < 0)
return ret;- /* set the codec system clock for DAC and ADC */
- ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, clk,
SND_SOC_CLOCK_OUT);- if (ret < 0)
return ret;- return 0;
+}
+static struct snd_soc_ops s3c24xx_uda1341_ops = {
- .hw_params = s3c24xx_uda1341_hw_params,
+};
+static struct snd_soc_dai_link s3c24xx_uda1341_dai_link = {
- .name = "UDA1341",
- .stream_name = "UDA1341",
- .codec_dai = &uda1341_dai,
- .cpu_dai = &s3c24xx_i2s_dai,
- .ops = &s3c24xx_uda1341_ops,
+};
+static struct snd_soc_machine snd_soc_machine_s3c24xx_uda1341 = {
- .name = "S3C24XX_UDA1341",
- .dai_link = &s3c24xx_uda1341_dai_link,
- .num_links = 1,
+};
+static struct s3c24xx_uda1341_platform_data *s3c24xx_uda1341_l3_pins;
+static void setdat(struct uda1341_platform_data *p, int v) +{
- s3c2410_gpio_setpin(s3c24xx_uda1341_l3_pins->l3_data, v > 0);
- s3c2410_gpio_cfgpin(s3c24xx_uda1341_l3_pins->l3_data,
S3C2410_GPIO_OUTPUT);+}
+static void setclk(struct uda1341_platform_data *p, int v) +{
- s3c2410_gpio_setpin(s3c24xx_uda1341_l3_pins->l3_clk, v > 0);
- s3c2410_gpio_cfgpin(s3c24xx_uda1341_l3_pins->l3_clk,
S3C2410_GPIO_OUTPUT);+}
+static void setmode(struct uda1341_platform_data *p, int v) +{
- s3c2410_gpio_setpin(s3c24xx_uda1341_l3_pins->l3_mode, v > 0);
- s3c2410_gpio_cfgpin(s3c24xx_uda1341_l3_pins->l3_mode,
S3C2410_GPIO_OUTPUT);+}
please use the generic gpio layer, the s3c24xx functions are going to be deprecated and removed as soon as possible, probably after 2.6.28.
+static void s3c24xx_uda1341_power(int en) +{
- if (s3c24xx_uda1341_l3_pins->power)
s3c24xx_uda1341_l3_pins->power(en);+}
+static struct uda1341_platform_data s3c24xx_uda1341 = {
- .setdat = setdat,
- .setclk = setclk,
- .setmode = setmode,
- .data_hold = 1,
- .data_setup = 1,
- .clock_high = 1,
- .mode_hold = 1,
- .mode = 1,
- .mode_setup = 1,
- .power = s3c24xx_uda1341_power,
+};
+static struct snd_soc_device s3c24xx_uda1341_snd_devdata = {
- .machine = &snd_soc_machine_s3c24xx_uda1341,
- .platform = &s3c24xx_soc_platform,
- .codec_dev = &soc_codec_dev_uda1341,
- .codec_data = &s3c24xx_uda1341,
+};
+static struct platform_device *s3c24xx_uda1341_snd_device;
+static int s3c24xx_uda1341_probe(struct platform_device *pdev) +{
- int ret;
- printk(KERN_INFO "S3C24XX_UDA1341 SoC Audio driver\n");
- s3c24xx_uda1341_l3_pins = pdev->dev.platform_data;
- if (s3c24xx_uda1341_l3_pins == NULL) {
printk(KERN_ERR "S3C24XX_UDA1341 SoC Audio: ""unable to find platform data\n");return -ENODEV;- }
- s3c24xx_uda1341_snd_device = platform_device_alloc("soc-audio", -1);
- if (!s3c24xx_uda1341_snd_device) {
printk(KERN_ERR "S3C24XX_UDA1341 SoC Audio: ""Unable to register\n");return -ENOMEM;- }
- platform_set_drvdata(s3c24xx_uda1341_snd_device,
&s3c24xx_uda1341_snd_devdata);- s3c24xx_uda1341_snd_devdata.dev = &s3c24xx_uda1341_snd_device->dev;
- ret = platform_device_add(s3c24xx_uda1341_snd_device);
- if (ret) {
printk(KERN_ERR "S3C24XX_UDA1341 SoC Audio: Unable to add\n");platform_device_put(s3c24xx_uda1341_snd_device);- }
- xtal = clk_get(NULL, "xtal");
- pclk = clk_get(NULL, "pclk");
check for errors. really you should be passing a device for pclk.
- return ret;
+}
+static int s3c24xx_uda1341_remove(struct platform_device *pdev) +{
- platform_device_unregister(s3c24xx_uda1341_snd_device);
- clk_put(xtal);
- clk_put(pclk);
- return 0;
+}
+static struct platform_driver s3c24xx_uda1341_driver = {
- .probe = s3c24xx_uda1341_probe,
- .remove = s3c24xx_uda1341_remove,
- .driver = {
.name = "s3c24xx_uda1341",.owner = THIS_MODULE,- },
+};
+static int __init s3c24xx_uda1341_init(void) +{
- return platform_driver_register(&s3c24xx_uda1341_driver);
+}
+static void __exit s3c24xx_uda1341_exit(void) +{
- platform_driver_unregister(&s3c24xx_uda1341_driver);
+}
+module_init(s3c24xx_uda1341_init); +module_exit(s3c24xx_uda1341_exit);
+MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin chripell@evolware.org"); +MODULE_DESCRIPTION("S3C24XX_UDA1341 ALSA SoC audio driver"); +MODULE_LICENSE("GPL");
On Mon, Nov 10, 2008 at 11:09 PM, Ben Dooks ben-linux@fluff.org wrote:
example into a doc file please, or point at an implementation of this.
ack
+#define setdat(adap, val) (adap->setdat(adap, val)) +#define setclk(adap, val) (adap->setclk(adap, val)) +#define setmode(adap, val) (adap->setmode(adap, val))
inline functions are better than #define.
ack
+#include "uda1341_platform_data.h"
please find as better name for that file.
I always have problems as far as platform data is concerned because it should be separated from the private data structures of the driver; so it's rather common to just have a small ad-hoc file for them. In socketcan a decision was made to put all the platform data in a subdirectory so the name can be short and one knows where to look. Do you have any suggestion where is the best place to put this file and for a consistent naming convention?
+}
do you really want to be busy-waiting the cpu for that long? how about msleep?
ack, I'll use msleep
void (*setmode) (struct uda1341_platform_data *, int);no spaces ^
ack. It's strange that checkpatch.pl didn't catch this
find somewhere for these to live that can be included without resorting to knowing the directory layout.
ack, I'll try include/sound
+}
please use the generic gpio layer, the s3c24xx functions are going to be deprecated and removed as soon as possible, probably after 2.6.28.
ack.
pclk = clk_get(NULL, "pclk");check for errors. really you should be passing a device for pclk.
ack
Thanks,
On Tue, Nov 11, 2008 at 03:14:04PM +0100, chri wrote:
On Mon, Nov 10, 2008 at 11:09 PM, Ben Dooks ben-linux@fluff.org wrote:
find somewhere for these to live that can be included without resorting to knowing the directory layout.
ack, I'll try include/sound
If they (you cut context so I'm not sure what exactly they are) are only referenced within ASoC leave them in sound/soc/codecs which is idiomatic for that. Otherwise either include/sound or arch/arm depending on if it's ARM/Samsung specific or not.
participants (6)
-
Ben Dooks -
chri -
Christian Pellegrin
-
Kristoffer Ericson -
Mark Brown -
Russell King - ARM Linux