[alsa-devel] [PATCH 1/5 v1] ASoC Add TLV320AIC23 codec driver
ASoC codec driver for TLV320AIC23 device
Signed-off-by: Arun KS arunks@mistralsolutions.com --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic23.c | 631 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23.h | 125 ++++++++ 4 files changed, 763 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic23.c create mode 100644 sound/soc/codecs/tlv320aic23.h
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0507fcf..bdead2d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -7,6 +7,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 select SND_SOC_CS4270 select SND_SOC_SSM2602 + select SND_SOC_TLV320AIC23 select SND_SOC_TLV320AIC26 select SND_SOC_TLV320AIC3X select SND_SOC_UDA1380 @@ -62,6 +63,10 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_SSM2602 tristate
+config SND_SOC_TLV320AIC23 + tristate + depends on I2C + config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" depends on SND_SOC && SPI diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 0731844..90f0a58 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -4,6 +4,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-uda1380-objs := uda1380.o @@ -25,6 +26,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c new file mode 100644 index 0000000..b602c87 --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.c @@ -0,0 +1,631 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, arunks@mistralsolutions.com + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/tlv320aic3x.c by Vladimir Barinov + * + * 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. + * + * Notes: + * The AIC23 is a driver for a low power stereo audio + * codec tlv320aic23 + * + * The machine layer should disable unsupported inputs/outputs by + * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "tlv320aic23.h" + +#define AUDIO_NAME "tlv320aic23" +#define AIC23_VERSION "0.1" + +/* codec private data */ +struct tlv320aic23_priv { + unsigned int sysclk; + int master; +}; + +struct tlv320aic23_samplerate_reg_info { + u32 sample_rate; + u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ + u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ +}; + +/* + * AIC23 register cache + */ +static const u16 tlv320aic23_reg[] = { + 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ + 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ + 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ +}; + +/* + * read tlv320aic23 register cache + */ +static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec + *codec, unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return -1; + return cache[reg]; +} + +/* + * write tlv320aic23 register cache + */ +static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u16 value) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return; + cache[reg] = value; +} + +/* + * write to the tlv320aic23 register space + */ +static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + + u8 data; + + /* TLV320AIC23 has 7 bit address and 9 bits of data + * so we need to switch one data bit into reg and rest + * of data into val + */ + + if ((reg < 0 || reg > 9) && (reg != 15)) { + printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg); + return -1; + } + + data = (reg << 1) | (value >> 8 & 0x01); + + tlv320aic23_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, (u8) data, + (u8) (value & 0xff)) == 0) + return 0; + + printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__, + value, reg); + + return -EIO; +} + +static const char *tlv320aic23_rec_src[] = { "Line", "Mic" }; +static const char *tlv320aic23_sidetone_src[] = + { "-6db", "-9db", "-12db", "-18db", "0db" }; + +static const struct soc_enum tlv320aic23_enum[] = { + SOC_ENUM_SINGLE(TLV320AIC23_ANALOG_AUDIO_CONTROL, 2, 2, + tlv320aic23_rec_src), + SOC_ENUM_SINGLE(TLV320AIC23_ANALOG_AUDIO_CONTROL, 6, 5, + tlv320aic23_sidetone_src), +}; + +static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = +SOC_DAPM_ENUM("Route", tlv320aic23_enum[0]); + +static const struct snd_kcontrol_new tlv320aic23_sidetone_src_controls = +SOC_DAPM_ENUM("Route", tlv320aic23_enum[1]); + +static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { + SOC_DOUBLE_R("PCM Playback Volume", TLV320AIC23_LEFT_CHANNEL_VOLUME, + TLV320AIC23_RIGHT_CHANNEL_VOLUME, 0, 0x7f, 0), + SOC_SINGLE("PCM Playback Switch", TLV320AIC23_DIGITAL_AUDIO_CONTROL, 3, + 0x01, 1), + SOC_DOUBLE_R("Line Input Mute", TLV320AIC23_LEFT_LINE_VOLUME, + TLV320AIC23_RIGHT_LINE_VOLUME, 7, 0x01, 0), + SOC_DOUBLE_R("Line Input Volume", TLV320AIC23_LEFT_LINE_VOLUME, + TLV320AIC23_RIGHT_LINE_VOLUME, 0, 0x1f, 0), + SOC_SINGLE("Mic Switch", TLV320AIC23_ANALOG_AUDIO_CONTROL, 1, 0x01, 0), + SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANALOG_AUDIO_CONTROL, 0, + 0x01, 0), + SOC_SINGLE("Sidetone Switch", TLV320AIC23_ANALOG_AUDIO_CONTROL, 5, 1, + 0), + SOC_ENUM("Sidetone Gain", tlv320aic23_enum[1]), + SOC_SINGLE("Line Bypass", TLV320AIC23_ANALOG_AUDIO_CONTROL, 3, 1, 0), + +}; + +/* add non dapm controls */ +static int tlv320aic23_add_controls(struct snd_soc_codec *codec) +{ + + int err, i; + + for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&tlv320aic23_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; + +} + +/* PGA Mixer controls for Line and Mic switch */ +static const struct snd_kcontrol_new tlv320aic23_pga_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Switch", TLV320AIC23_POWER_DOWN_CONTROL, 0, 1, 1), + SOC_DAPM_SINGLE("Mic Switch", TLV320AIC23_POWER_DOWN_CONTROL, 1, 1, 1), +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_POWER_DOWN_CONTROL, 3, + 1), + SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_POWER_DOWN_CONTROL, 2, + 1), + SND_SOC_DAPM_MIXER("PGA Mixer", SND_SOC_NOPM, 0, 0, + &tlv320aic23_pga_mixer_controls[0], + ARRAY_SIZE(tlv320aic23_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0, + &tlv320aic23_rec_src_mux_controls), + + SND_SOC_DAPM_OUTPUT("LHPOUT"), + SND_SOC_DAPM_OUTPUT("RHPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_INPUT("LLINEIN"), + SND_SOC_DAPM_INPUT("RLINEIN"), + SND_SOC_DAPM_INPUT("MICIN"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + {"LHPOUT", NULL, "DAC"}, + {"RHPOUT", NULL, "DAC"}, + + {"LOUT", NULL, "DAC"}, + {"ROUT", NULL, "DAC"}, + + {"Capture Source", "Line", "LLINEIN"}, + {"Capture Source", "Line", "RLINEIN"}, + + {"Capture Source", "Mic", "MICIN"}, + + {"PGA Mixer", "Line Switch", "Capture Source"}, + {"PGA Mixer", "Mic Switch", "Capture Source"}, + + {"ADC", NULL, "PGA Mixer"}, +}; + +/* tlv320aic23 related */ +static const struct tlv320aic23_samplerate_reg_info rate_reg_info[] = { + {4000, 0x06, 1}, /* 4000 */ + {8000, 0x06, 0}, /* 8000 */ + {16000, 0x0C, 1}, /* 16000 */ + {22050, 0x11, 1}, /* 22050 */ + {24000, 0x00, 1}, /* 24000 */ + {32000, 0x0C, 0}, /* 32000 */ + {44100, 0x11, 0}, /* 44100 */ + {48000, 0x00, 0}, /* 48000 */ + {88200, 0x1F, 0}, /* 88200 */ + {96000, 0x0E, 0}, /* 96000 */ +}; + +static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int tlv320aic23_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; + u16 iface_reg, data; + u8 count = 0; + + iface_reg = tlv320aic23_read_reg_cache(codec, + TLV320AIC23_DIGITAL_AUDIO_FORMAT) + & ~(0x03 << 2); + + /* Search for the right sample rate */ + /* Verify what happens if the rate is not supported + * now it goes to 96Khz */ + while ((rate_reg_info[count].sample_rate != params_rate(params)) && + (count < ARRAY_SIZE(rate_reg_info))) { + count++; + } + + data = (rate_reg_info[count].divider << CLKIN_SHIFT) | + (rate_reg_info[count].control << BOSR_SHIFT) | USB_CLK_ON; + + tlv320aic23_write(codec, TLV320AIC23_SAMPLE_RATE_CONTROL, data); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_reg |= (0x01 << 2); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_reg |= (0x02 << 2); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_reg |= (0x03 << 2); + break; + } + tlv320aic23_write(codec, TLV320AIC23_DIGITAL_AUDIO_FORMAT, iface_reg); + + return 0; +} + +static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg; + + reg = + tlv320aic23_read_reg_cache(codec, + TLV320AIC23_DIGITAL_AUDIO_CONTROL) & + ~DACM_MUTE; + + if (mute) + tlv320aic23_write(codec, TLV320AIC23_DIGITAL_AUDIO_CONTROL, + reg | DACM_MUTE); + + else + tlv320aic23_write(codec, TLV320AIC23_DIGITAL_AUDIO_CONTROL, + reg); + + return 0; +} + +static int tlv320aic23_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 tlv320aic23_priv *tlv320aic23 = codec->private_data; + + tlv320aic23->sysclk = freq; + return 0; +} + +static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tlv320aic23_priv *tlv320aic23 = codec->private_data; + u16 iface_reg; + + iface_reg = + tlv320aic23_read_reg_cache(codec, + TLV320AIC23_DIGITAL_AUDIO_FORMAT) & + (~0x03); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + tlv320aic23->master = 1; + iface_reg |= MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + tlv320aic23->master = 0; + break; + default: + return -EINVAL; + + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface_reg |= FOR_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= FOR_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg |= FOR_LJUST; + break; + default: + return -EINVAL; + + } + + tlv320aic23_write(codec, TLV320AIC23_DIGITAL_AUDIO_FORMAT, iface_reg); + + return 0; +} + +#define AIC23_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai tlv320aic23_dai = { + .name = "tlv320aic23", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .ops = { + .hw_params = tlv320aic23_hw_params, + }, + .dai_ops = { + .digital_mute = tlv320aic23_mute, + .set_sysclk = tlv320aic23_set_dai_sysclk, + .set_fmt = tlv320aic23_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(tlv320aic23_dai); + +static int tlv320aic23_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; + u16 reg; + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_POWER_DOWN_CONTROL); + tlv320aic23_write(codec, TLV320AIC23_POWER_DOWN_CONTROL, + reg | DEVICE_POWER_OFF | OUT_OFF); + + return 0; +} + +static int tlv320aic23_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 reg; + u8 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(tlv320aic23_reg); i++) { + data[0] = i; + data[1] = cache[i]; + codec->hw_write(codec->control_data, data, 2); + } + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_POWER_DOWN_CONTROL); + tlv320aic23_write(codec, TLV320AIC23_POWER_DOWN_CONTROL, + reg & (~DEVICE_POWER_OFF) & (~OUT_OFF)); + + return 0; +} + +/* + * initialise the AIC23 driver + * register the mixer and dsp interfaces with the kernel + */ +static int tlv320aic23_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + u16 reg; + + codec->name = "tlv320aic23"; + codec->owner = THIS_MODULE; + codec->read = tlv320aic23_read_reg_cache; + codec->write = tlv320aic23_write; + codec->dai = &tlv320aic23_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg); + codec->reg_cache = + kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* Reset codec */ + tlv320aic23_write(codec, TLV320AIC23_RESET_CONTROL, 0); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to create pcms\n"); + goto pcm_err; + } + /* Power on the device */ + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_POWER_DOWN_CONTROL); + tlv320aic23_write(codec, TLV320AIC23_POWER_DOWN_CONTROL, + reg & (~DEVICE_POWER_OFF) & (~OUT_OFF)); + + tlv320aic23_write(codec, TLV320AIC23_DIGITAL_AUDIO_CONTROL, DEEMP_44K); + + /* Unmute input */ + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LEFT_LINE_VOLUME); + tlv320aic23_write(codec, TLV320AIC23_LEFT_LINE_VOLUME, + (reg & (~LIM_MUTED)) | (LRS_ENABLED)); + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RIGHT_LINE_VOLUME); + tlv320aic23_write(codec, TLV320AIC23_RIGHT_LINE_VOLUME, + (reg & (~LIM_MUTED)) | LRS_ENABLED); + reg = + tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANALOG_AUDIO_CONTROL); + tlv320aic23_write(codec, TLV320AIC23_ANALOG_AUDIO_CONTROL, + (reg | DAC_SELECTED) & (~MICM_MUTED) & (~BYPASS_ON)); + + /* Default output volume */ + tlv320aic23_write(codec, TLV320AIC23_LEFT_CHANNEL_VOLUME, + DEFAULT_OUTPUT_VOLUME & OUTPUT_VOLUME_MASK); + tlv320aic23_write(codec, TLV320AIC23_RIGHT_CHANNEL_VOLUME, + DEFAULT_OUTPUT_VOLUME & OUTPUT_VOLUME_MASK); + + /* Activate interface */ + reg = + tlv320aic23_read_reg_cache(codec, + TLV320AIC23_DIGITAL_INTERFACE_ACT); + tlv320aic23_write(codec, TLV320AIC23_DIGITAL_INTERFACE_ACT, + reg | ACT_ON); + + tlv320aic23_add_controls(codec); + tlv320aic23_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} +static struct snd_soc_device *tlv320aic23_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * If the i2c layer weren't so broken, we could pass this kind of data + * around + */ +static int tlv320aic23_codec_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct snd_soc_device *socdev = tlv320aic23_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = tlv320aic23_init(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} +static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) +{ + put_device(&i2c->dev); + return 0; +} + +static const struct i2c_device_id tlv320aic23_id[] = { + {"tlv320aic23", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tlv320aic23_id); + +static struct i2c_driver tlv320aic23_i2c_driver = { + .driver = { + .name = "tlv320aic23", + }, + .probe = tlv320aic23_codec_probe, + .remove = __exit_p(tlv320aic23_i2c_remove), + .id_table = tlv320aic23_id, +}; + +#endif + +static int tlv320aic23_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct tlv320aic23_priv *tlv320aic23; + int ret = 0; + + printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + tlv320aic23 = kzalloc(sizeof(struct tlv320aic23_priv), GFP_KERNEL); + if (tlv320aic23 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = tlv320aic23; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + tlv320aic23_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data; + codec->hw_read = NULL; + ret = i2c_add_driver(&tlv320aic23_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); +#endif + return ret; +} + +static int tlv320aic23_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&tlv320aic23_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} +struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = { + .probe = tlv320aic23_probe, + .remove = tlv320aic23_remove, + .suspend = tlv320aic23_suspend, + .resume = tlv320aic23_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); +MODULE_AUTHOR("Arun KS arunks@mistralsolutions.com"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h new file mode 100644 index 0000000..7ffeaf6 --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.h @@ -0,0 +1,125 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, arunks@mistralsolutions.com + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd + * + * 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 _TLV320AIC23_H +#define _TLV320AIC23_H + +/* Codec TLV320AIC23 */ +#define TLV320AIC23_LEFT_LINE_VOLUME 0x00 +#define TLV320AIC23_RIGHT_LINE_VOLUME 0x01 +#define TLV320AIC23_LEFT_CHANNEL_VOLUME 0x02 +#define TLV320AIC23_RIGHT_CHANNEL_VOLUME 0x03 +#define TLV320AIC23_ANALOG_AUDIO_CONTROL 0x04 +#define TLV320AIC23_DIGITAL_AUDIO_CONTROL 0x05 +#define TLV320AIC23_POWER_DOWN_CONTROL 0x06 +#define TLV320AIC23_DIGITAL_AUDIO_FORMAT 0x07 +#define TLV320AIC23_SAMPLE_RATE_CONTROL 0x08 +#define TLV320AIC23_DIGITAL_INTERFACE_ACT 0x09 +#define TLV320AIC23_RESET_CONTROL 0x0F + +/* Left (right) line input volume control register */ +#define LRS_ENABLED 0x0100 +#define LIM_MUTED 0x0080 +#define LIV_DEFAULT 0x0017 +#define LIV_MAX 0x001f +#define LIV_MIN 0x0000 + +/* Left (right) channel headphone volume control register */ +#define LZC_ON 0x0080 +#define LHV_DEFAULT 0x0079 +#define LHV_MAX 0x007f +#define LHV_MIN 0x0000 + +/* Analog audio path control register */ +#define STA_REG(x) ((x)<<6) +#define STE_ENABLED 0x0020 +#define DAC_SELECTED 0x0010 +#define BYPASS_ON 0x0008 +#define INSEL_MIC 0x0004 +#define MICM_MUTED 0x0002 +#define MICB_20DB 0x0001 + +/* Digital audio path control register */ +#define DACM_MUTE 0x0008 +#define DEEMP_32K 0x0002 +#define DEEMP_44K 0x0004 +#define DEEMP_48K 0x0006 +#define ADCHP_ON 0x0001 + +/* Power control down register */ +#define DEVICE_POWER_OFF 0x0080 +#define CLK_OFF 0x0040 +#define OSC_OFF 0x0020 +#define OUT_OFF 0x0010 +#define DAC_OFF 0x0008 +#define ADC_OFF 0x0004 +#define MIC_OFF 0x0002 +#define LINE_OFF 0x0001 + +/* Digital audio interface register */ +#define MS_MASTER 0x0040 +#define LRSWAP_ON 0x0020 +#define LRP_ON 0x0010 +#define IWL_16 0x0000 +#define IWL_20 0x0004 +#define IWL_24 0x0008 +#define IWL_32 0x000C +#define FOR_I2S 0x0002 +#define FOR_DSP 0x0003 +#define FOR_LJUST 0x0001 + +/* Sample rate control register */ +#define CLKOUT_HALF 0x0080 +#define CLKIN_HALF 0x0040 +#define BOSR_384fs 0x0002 /* BOSR_272fs when in USB mode */ +#define USB_CLK_ON 0x0001 +#define SR_MASK 0xf +#define CLKOUT_SHIFT 7 +#define CLKIN_SHIFT 6 +#define SR_SHIFT 2 +#define BOSR_SHIFT 1 + +/* Digital interface register */ +#define ACT_ON 0x0001 + +/* + * AUDIO related MACROS + */ + +#define DEFAULT_OUTPUT_VOLUME 0x70 +#define DEFAULT_INPUT_VOLUME 0x10 /* 0 ==> mute line in */ + +#define OUTPUT_VOLUME_MIN LHV_MIN +#define OUTPUT_VOLUME_MAX LHV_MAX +#define OUTPUT_VOLUME_RANGE (OUTPUT_VOLUME_MAX - OUTPUT_VOLUME_MIN) +#define OUTPUT_VOLUME_MASK OUTPUT_VOLUME_MAX + +#define INPUT_VOLUME_MIN LIV_MIN +#define INPUT_VOLUME_MAX LIV_MAX +#define INPUT_VOLUME_RANGE (INPUT_VOLUME_MAX - INPUT_VOLUME_MIN) +#define INPUT_VOLUME_MASK INPUT_VOLUME_MAX + +#define SIDETONE_MASK 0x1c0 +#define SIDETONE_0 0x100 +#define SIDETONE_6 0x000 +#define SIDETONE_9 0x040 +#define SIDETONE_12 0x080 +#define SIDETONE_18 0x0c0 + +#define DEFAULT_ANALOG_AUDIO_CONTROL (DAC_SELECTED | STE_ENABLED | \ + INSEL_MIC | MICB_20DB) + + +extern struct snd_soc_dai tlv320aic23_dai; +extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23; + +#endif /* _TLV320AIC23_H */ +
On Tue, Sep 30, 2008 at 03:29:32PM +0530, Arun KS wrote:
Thanks! It looks like you've addressed pretty much all the comments from the first round of reviews but a few additional (fairly minor comments):
+static const char *tlv320aic23_rec_src[] = { "Line", "Mic" }; +static const char *tlv320aic23_sidetone_src[] =
- { "-6db", "-9db", "-12db", "-18db", "0db" };
+static const struct soc_enum tlv320aic23_enum[] = {
- SOC_ENUM_SINGLE(TLV320AIC23_ANALOG_AUDIO_CONTROL, 2, 2,
tlv320aic23_rec_src),- SOC_ENUM_SINGLE(TLV320AIC23_ANALOG_AUDIO_CONTROL, 6, 5,
tlv320aic23_sidetone_src),+};
sidetone_src really ought to be a SOC_SINGLE_TLV rather than an enum from the looks of it - it's not selecting between sources for the sidetone, it's determining the level of the signal so a gain control with TLV information would present better in UIs.
+static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = +SOC_DAPM_ENUM("Route", tlv320aic23_enum[0]);
+static const struct snd_kcontrol_new tlv320aic23_sidetone_src_controls = +SOC_DAPM_ENUM("Route", tlv320aic23_enum[1]);
I know some of the older codec drivers do it but there's no need to have the enums be an array and it doesn't help clarity. Equally well, it's not a blocker for merge.
+static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
- SOC_DOUBLE_R("PCM Playback Volume", TLV320AIC23_LEFT_CHANNEL_VOLUME,
TLV320AIC23_RIGHT_CHANNEL_VOLUME, 0, 0x7f, 0),
TLV information might be nice here too (but it's not required). I suspect "Digital Playback Volume" would be a better name, but I'm not 100% sure on that one.
- SOC_DOUBLE_R("Line Input Mute", TLV320AIC23_LEFT_LINE_VOLUME,
TLV320AIC23_RIGHT_LINE_VOLUME, 7, 0x01, 0),
This should be "Line Input Switch" for user interfaces to pick it up properly.
+static const struct snd_soc_dapm_route intercon[] = {
- {"LHPOUT", NULL, "DAC"},
- {"RHPOUT", NULL, "DAC"},
- {"LOUT", NULL, "DAC"},
- {"ROUT", NULL, "DAC"},
- {"Capture Source", "Line", "LLINEIN"},
- {"Capture Source", "Line", "RLINEIN"},
- {"Capture Source", "Mic", "MICIN"},
- {"PGA Mixer", "Line Switch", "Capture Source"},
- {"PGA Mixer", "Mic Switch", "Capture Source"},
- {"ADC", NULL, "PGA Mixer"},
+};
There are no routes here for your bypass path(s) - this will mean that DAPM won't power them on unless there's an active playback and record. At present that's fine for digital only bypass paths but if there are analogue ones they should be visible here.
+static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) +{
- struct snd_soc_codec *codec = dai->codec;
- u16 reg;
- reg =
tlv320aic23_read_reg_cache(codec,TLV320AIC23_DIGITAL_AUDIO_CONTROL) &~DACM_MUTE;
Have you resolved the word wrapping issues with your mailer? I've not tried applying the patch but things like this make me think that it has been mangled.
+static int tlv320aic23_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 tlv320aic23_priv *tlv320aic23 = codec->private_data;
- tlv320aic23->sysclk = freq;
- return 0;
+}
You don't actually use sysclk for anything so you may as well drop this function and the local variable.
tlv320aic23->master = 1;iface_reg |= MS_MASTER;break;- case SND_SOC_DAIFMT_CBS_CFS:
tlv320aic23->master = 0;break;
You don't seem to use master anywhere either so it could also be dropped?
+/* Left (right) line input volume control register */ +#define LRS_ENABLED 0x0100 +#define LIM_MUTED 0x0080 +#define LIV_DEFAULT 0x0017 +#define LIV_MAX 0x001f +#define LIV_MIN 0x0000
All the definitions in the header should be namespaced - there's things like:
+/* Power control down register */ +#define DEVICE_POWER_OFF 0x0080 +#define CLK_OFF 0x0040 +#define OSC_OFF 0x0020 +#define OUT_OFF 0x0010 +#define DAC_OFF 0x0008 +#define ADC_OFF 0x0004 +#define MIC_OFF 0x0002 +#define LINE_OFF 0x0001
which are very likely to collide with other chips.
On Tue, Sep 30, 2008 at 3:49 PM, Mark Brown broonie@sirena.org.uk wrote:
On Tue, Sep 30, 2008 at 03:29:32PM +0530, Arun KS wrote:
Thanks! It looks like you've addressed pretty much all the comments from the first round of reviews but a few additional (fairly minor comments):
+static const char *tlv320aic23_rec_src[] = { "Line", "Mic" }; +static const char *tlv320aic23_sidetone_src[] =
- { "-6db", "-9db", "-12db", "-18db", "0db" };
+static const struct soc_enum tlv320aic23_enum[] = {
SOC_ENUM_SINGLE(TLV320AIC23_ANALOG_AUDIO_CONTROL, 2, 2,tlv320aic23_rec_src),SOC_ENUM_SINGLE(TLV320AIC23_ANALOG_AUDIO_CONTROL, 6, 5,tlv320aic23_sidetone_src),+};
sidetone_src really ought to be a SOC_SINGLE_TLV rather than an enum from the looks of it - it's not selecting between sources for the sidetone, it's determining the level of the signal so a gain control with TLV information would present better in UIs.
Thanks for the comments. Will do that.
+static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = +SOC_DAPM_ENUM("Route", tlv320aic23_enum[0]);
+static const struct snd_kcontrol_new tlv320aic23_sidetone_src_controls = +SOC_DAPM_ENUM("Route", tlv320aic23_enum[1]);
I know some of the older codec drivers do it but there's no need to have the enums be an array and it doesn't help clarity. Equally well, it's not a blocker for merge.
+static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
SOC_DOUBLE_R("PCM Playback Volume", TLV320AIC23_LEFT_CHANNEL_VOLUME,TLV320AIC23_RIGHT_CHANNEL_VOLUME, 0, 0x7f, 0),TLV information might be nice here too (but it's not required). I suspect "Digital Playback Volume" would be a better name, but I'm not 100% sure on that one.
SOC_DOUBLE_R("Line Input Mute", TLV320AIC23_LEFT_LINE_VOLUME,TLV320AIC23_RIGHT_LINE_VOLUME, 7, 0x01, 0),This should be "Line Input Switch" for user interfaces to pick it up prop
+static const struct snd_soc_dapm_route intercon[] = {
{"LHPOUT", NULL, "DAC"},{"RHPOUT", NULL, "DAC"},{"LOUT", NULL, "DAC"},{"ROUT", NULL, "DAC"},{"Capture Source", "Line", "LLINEIN"},{"Capture Source", "Line", "RLINEIN"},{"Capture Source", "Mic", "MICIN"},{"PGA Mixer", "Line Switch", "Capture Source"},{"PGA Mixer", "Mic Switch", "Capture Source"},{"ADC", NULL, "PGA Mixer"},+};
There are no routes here for your bypass path(s) - this will mean that DAPM won't power them on unless there's an active playback and record. At present that's fine for digital only bypass paths but if there are analogue ones they should be visible here.
I made the bypass and sidetone as controls. The user can switchon them through amixer controls.
sh-3.00# amixer controls numid=2,iface=MIXER,name='PCM Playback Switch' numid=1,iface=MIXER,name='PCM Playback Volume' numid=3,iface=MIXER,name='Line Input Mute' numid=4,iface=MIXER,name='Line Input Volume' numid=9,iface=MIXER,name='Line Bypass' numid=6,iface=MIXER,name='Mic Booster Switch' numid=5,iface=MIXER,name='Mic Switch' numid=10,iface=MIXER,name='Capture Source' numid=11,iface=MIXER,name='PGA Mixer Line Switch' numid=12,iface=MIXER,name='PGA Mixer Mic Switch' numid=8,iface=MIXER,name='Sidetone Gain' numid=7,iface=MIXER,name='Sidetone Switch'
sh-3.00# amixer cset numid=9 1 numid=9,iface=MIXER,name='Line Bypass' ; type=BOOLEAN,access=rw------,values=1 : values=on sh-3.00# amixer cset numid=11 1 numid=11,iface=MIXER,name='PGA Mixer Line Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on
Its working.
+static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) +{
struct snd_soc_codec *codec = dai->codec;u16 reg;reg =tlv320aic23_read_reg_cache(codec,TLV320AIC23_DIGITAL_AUDIO_CONTROL) &~DACM_MUTE;Have you resolved the word wrapping issues with your mailer? I've not tried applying the patch but things like this make me think that it has been mangled.
+static int tlv320aic23_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 tlv320aic23_priv *tlv320aic23 = codec->private_data;tlv320aic23->sysclk = freq;return 0;+}
You don't actually use sysclk for anything so you may as well drop this function and the local variable.
I'll do it.
tlv320aic23->master = 1;iface_reg |= MS_MASTER;break;case SND_SOC_DAIFMT_CBS_CFS:tlv320aic23->master = 0;break;You don't seem to use master anywhere either so it could also be dropped?
Ok i'll drop.
+/* Left (right) line input volume control register */ +#define LRS_ENABLED 0x0100 +#define LIM_MUTED 0x0080 +#define LIV_DEFAULT 0x0017 +#define LIV_MAX 0x001f +#define LIV_MIN 0x0000
All the definitions in the header should be namespaced - there's things like:
+/* Power control down register */ +#define DEVICE_POWER_OFF 0x0080 +#define CLK_OFF 0x0040 +#define OSC_OFF 0x0020 +#define OUT_OFF 0x0010 +#define DAC_OFF 0x0008 +#define ADC_OFF 0x0004 +#define MIC_OFF 0x0002 +#define LINE_OFF 0x0001
which are very likely to collide with other chips. _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
On Tue, Sep 30, 2008 at 04:17:38PM +0530, Arun KS wrote:
On Tue, Sep 30, 2008 at 3:49 PM, Mark Brown broonie@sirena.org.uk wrote:
There are no routes here for your bypass path(s) - this will mean that DAPM won't power them on unless there's an active playback and record. At present that's fine for digital only bypass paths but if there are analogue ones they should be visible here.
I made the bypass and sidetone as controls. The user can switchon them through amixer controls.
Right, they should be visible DAPM controls to allow user control.
sh-3.00# amixer cset numid=9 1 numid=9,iface=MIXER,name='Line Bypass' ; type=BOOLEAN,access=rw------,values=1 : values=on sh-3.00# amixer cset numid=11 1 numid=11,iface=MIXER,name='PGA Mixer Line Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on
Its working.
Hrm. That suggests that there's a bug in the power management somewhere - without routes DAPM won't know to power up any of the output stages when the bypass paths are in use, leaving them unpowered.
Note that a lot of PGAs will pass through audio when powered down so you need to check that things like mute and gain controls are functioning as expected rather than just testing for the presence of a signal. Then again, looking at your driver again I don't see any power managent for the output stages either...
You can see what DAPM is powering up by looking at:
/sys/bus/platform/devices/soc-audio/dapm_widgets
Even if this works within the device DAPM still needs to know about the bypass paths so that it can manage components outside the codec.
participants (2)
-
Arun KS -
Mark Brown