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

[alsa-devel] Issue with creative Xfi PCIe ca0110-IBG
by Guillem Solà 15 Oct '09

15 Oct '09

Hi, I have a Creative XFi PCIe with ca0110-IBG chip. It's primary use is audio input for streaming on a brand new Dell server with RHEL. I have been testing latest kernel 2.6.31 through it's releases candidates and the card stoped working on 2.6.31-rc6, so now I'm stuck at 2.6.31-rc5. With rc5 I made a 2 weeks test and it went flawlessly. There's another guy who referenced this issue on http://mailman.alsa-project.org/pipermail/alsa-devel/2009-September/020876.… and Takashi Iwai said that there is a communication error between the codec and the controller. Any workaround? Is there a bug created related to this issue? I tried to "extract" the alsa-driver on 2.6.31-rc5 and install it 2.6.31 final without success. Also tried to get old snapshots from alsa-driver and alsa-kmirror but I cannot compile them. Any place where get some info about how to create my output from alsa-info: http://www.alsa-project.org/db/?f=09087914dfdd85a07dca432cf71a9626b4fc4251 regards, Guillem Solà

2 25

[alsa-devel] [PATCH - try2] ASoC: Codec driver for Texas Instruments tlv320dac33 codec
by Peter Ujfalusi 15 Oct '09

15 Oct '09

Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo audio DAC. TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low power audio playback. The digital interface can use I2S, DSP (A or B), Right and Left justified formats. DAC33 has stereo analog input, which can be bypassed to the analog outputs. Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass' mode (default) and nSample mode (FIFO is in use). a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is working synchronously as a normal codec (it needs constant stream of data on the digital interface). b) The nSample mode implementation uses one interrupt line from DAC33 to the host: Alarm threshold is set to 10ms of audio data (limit by the driver implementation). DAC33 will signal an interrupt, when the FIFO level goes under the Alarm threshold. The host will write to nSample register a value (number of stereo samples), to tell DAC33 how many samples it should read in a burst from the host. When the DAC33 received the number of samples, it disables the clocks on the I2S bus. When the FIFO use again goes under the Alarm threshold, DAC33 signals the host with an interrupt, and the process is repeated. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- I think I have addressed all the comments given for the first series. I might have missed something, but in short: - Handling of mutex is reworked, so it is not taken for all I2C accesses, but still protects the chip_power variable. - nsample_switch handling changed - IRQ handling reworked - Handling of the hard and soft power states are cleaner - definitions are namespeced - Some generic cleanups include/sound/tlv320dac33-plat.h | 20 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320dac33.c | 1235 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320dac33.h | 267 ++++++++ 5 files changed, 1528 insertions(+), 0 deletions(-) create mode 100644 include/sound/tlv320dac33-plat.h create mode 100644 sound/soc/codecs/tlv320dac33.c create mode 100644 sound/soc/codecs/tlv320dac33.h diff --git a/include/sound/tlv320dac33-plat.h b/include/sound/tlv320dac33-plat.h new file mode 100644 index 0000000..5858d06 --- /dev/null +++ b/include/sound/tlv320dac33-plat.h @@ -0,0 +1,20 @@ +/* + * Platform header for Texas Instruments TLV320DAC33 codec driver + * + * Author: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> + * + * Copyright: (C) 2009 Nokia Corporation + * + * 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 __TLV320DAC33_PLAT_H +#define __TLV320DAC33_PLAT_H + +struct tlv320dac33_platform_data { + int power_gpio; +}; + +#endif /* __TLV320DAC33_PLAT_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index fab01c9..49bab93 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C + select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_UDA134X @@ -142,6 +143,9 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC3X tristate +config SND_SOC_TLV320DAC33 + tristate + config SND_SOC_TWL4030 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 2f14391..8f519ee 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -17,6 +17,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-twl4030-objs := twl4030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o @@ -70,6 +71,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(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-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c new file mode 100644 index 0000000..a28b98a --- /dev/null +++ b/sound/soc/codecs/tlv320dac33.c @@ -0,0 +1,1235 @@ +/* + * ALSA SoC Texas Instruments TLV320DAC33 codec driver + * + * Author: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> + * + * Copyright: (C) 2009 Nokia Corporation + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/gpio.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 <sound/tlv.h> + +#include <sound/tlv320dac33-plat.h> +#include "tlv320dac33.h" + +#define DAC33_BUFFER_SIZE_BYTES 24576 /* bytes, 12288 16 bit words, + * 6144 stereo */ +#define DAC33_BUFFER_SIZE_SAMPLES 6144 + +#define NSAMPLE_MAX 5700 + +#define LATENCY_TIME_MS 20 + +static struct snd_soc_codec *tlv320dac33_codec; + +enum dac33_state { + DAC33_IDLE = 0, + DAC33_PREFILL, + DAC33_PLAYBACK, + DAC33_FLUSH, +}; + +struct tlv320dac33_priv { + struct mutex mutex; + struct workqueue_struct *dac33_wq; + struct work_struct work; + struct snd_soc_codec codec; + int power_gpio; + int chip_power; + int irq; + unsigned int refclk; + + unsigned int alarm_threshold; /* set to be half of LATENCY_TIME_MS */ + unsigned int nsample_min; /* nsample should not be lower than + * this */ + unsigned int nsample_max; /* nsample should not be higher than + * this */ + unsigned int nsample_switch; /* Use FIFO or bypass FIFO switch */ + unsigned int nsample; /* burst read amount from host */ + + enum dac33_state state; +}; + +static const u8 dac33_reg[DAC33_CACHEREGNUM] = { +0x00, 0x00, 0x00, 0x00, /* 0x00 - 0x03 */ +0x00, 0x00, 0x00, 0x00, /* 0x04 - 0x07 */ +0x00, 0x00, 0x00, 0x00, /* 0x08 - 0x0b */ +0x00, 0x00, 0x00, 0x00, /* 0x0c - 0x0f */ +0x00, 0x00, 0x00, 0x00, /* 0x10 - 0x13 */ +0x00, 0x00, 0x00, 0x00, /* 0x14 - 0x17 */ +0x00, 0x00, 0x00, 0x00, /* 0x18 - 0x1b */ +0x00, 0x00, 0x00, 0x00, /* 0x1c - 0x1f */ +0x00, 0x00, 0x00, 0x00, /* 0x20 - 0x23 */ +0x00, 0x00, 0x00, 0x00, /* 0x24 - 0x27 */ +0x00, 0x00, 0x00, 0x00, /* 0x28 - 0x2b */ +0x00, 0x00, 0x00, 0x80, /* 0x2c - 0x2f */ +0x80, 0x00, 0x00, 0x00, /* 0x30 - 0x33 */ +0x00, 0x00, 0x00, 0x00, /* 0x34 - 0x37 */ +0x00, 0x00, /* 0x38 - 0x39 */ +/* Registers 0x3a - 0x3f are reserved */ + 0x00, 0x00, /* 0x3a - 0x3b */ +0x00, 0x00, 0x00, 0x00, /* 0x3c - 0x3f */ + +0x00, 0x00, 0x00, 0x00, /* 0x40 - 0x43 */ +0x00, 0x80, /* 0x44 - 0x45 */ +/* Registers 0x46 - 0x47 are reserved */ + 0x80, 0x80, /* 0x46 - 0x47 */ + +0x80, 0x00, 0x00, /* 0x48 - 0x4a */ +/* Registers 0x4b - 0x7c are reserved */ + 0x00, /* 0x4b */ +0x00, 0x00, 0x00, 0x00, /* 0x4c - 0x4f */ +0x00, 0x00, 0x00, 0x00, /* 0x50 - 0x53 */ +0x00, 0x00, 0x00, 0x00, /* 0x54 - 0x57 */ +0x00, 0x00, 0x00, 0x00, /* 0x58 - 0x5b */ +0x00, 0x00, 0x00, 0x00, /* 0x5c - 0x5f */ +0x00, 0x00, 0x00, 0x00, /* 0x60 - 0x63 */ +0x00, 0x00, 0x00, 0x00, /* 0x64 - 0x67 */ +0x00, 0x00, 0x00, 0x00, /* 0x68 - 0x6b */ +0x00, 0x00, 0x00, 0x00, /* 0x6c - 0x6f */ +0x00, 0x00, 0x00, 0x00, /* 0x70 - 0x73 */ +0x00, 0x00, 0x00, 0x00, /* 0x74 - 0x77 */ +0x00, 0x00, 0x00, 0x00, /* 0x78 - 0x7b */ +0x00, /* 0x7c */ + + 0xda, 0x33, 0x03, /* 0x7d - 0x7f */ +}; + +/* Register read and write */ +static inline unsigned int dac33_read_reg_cache(struct snd_soc_codec *codec, + unsigned reg) +{ + u8 *cache = codec->reg_cache; + if (reg >= DAC33_CACHEREGNUM) + return 0; + + return cache[reg]; +} + +static inline void dac33_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + if (reg >= DAC33_CACHEREGNUM) + return; + + cache[reg] = value; +} + +static int dac33_read(struct snd_soc_codec *codec, unsigned int reg, + u8 *value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + int val; + + *value = reg & 0xff; + + /* If powered off, return the cached value */ + if (dac33->chip_power) { + val = i2c_smbus_read_byte_data(codec->control_data, value[0]); + if (val < 0) { + dev_err(codec->dev, "Read failed (%d)\n", val); + value[0] = dac33_read_reg_cache(codec, reg); + } else { + value[0] = val; + dac33_write_reg_cache(codec, reg, val); + } + } else { + value[0] = dac33_read_reg_cache(codec, reg); + } + + return 0; +} + +static int dac33_write_locked(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 data[2]; + int ret = 0; + + /* + * data is + * D15..D8 dac33 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + dac33_write_reg_cache(codec, data[0], data[1]); + mutex_lock(&dac33->mutex); + if (dac33->chip_power) { + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + else + ret = 0; + } + mutex_unlock(&dac33->mutex); + + return ret; +} + +static int dac33_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 data[2]; + int ret = 0; + + /* + * data is + * D15..D8 dac33 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + dac33_write_reg_cache(codec, data[0], data[1]); + if (dac33->chip_power) { + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + else + ret = 0; + } + + return ret; +} + +#define DAC33_I2C_ADDR_AUTOINC 0x80 +static int dac33_write16(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 data[3]; + int ret = 0; + + /* + * data is + * D23..D16 dac33 register offset + * D15..D8 register data MSB + * D7...D0 register data LSB + */ + data[0] = reg & 0xff; + data[1] = (value >> 8) & 0xff; + data[2] = value & 0xff; + + dac33_write_reg_cache(codec, data[0], data[1]); + dac33_write_reg_cache(codec, data[0] + 1, data[2]); + + if (dac33->chip_power) { + /* We need to set autoincrement mode for 16 bit writes */ + data[0] |= DAC33_I2C_ADDR_AUTOINC; + ret = codec->hw_write(codec->control_data, data, 3); + if (ret != 3) + dev_err(codec->dev, "Write failed (%d)\n", ret); + else + ret = 0; + } + + return ret; +} + +static void dac33_restore_regs(struct snd_soc_codec *codec) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 *cache = codec->reg_cache; + u8 data[2]; + int i, ret; + + if (!dac33->chip_power) + return; + + for (i = DAC33_PWR_CTRL; i <= DAC33_INTP_CTRL_B; i++) { + data[0] = i; + data[1] = cache[i]; + /* Skip the read only registers */ + if ((i >= DAC33_INT_OSC_STATUS && + i <= DAC33_INT_OSC_FREQ_RAT_READ_B) || + (i >= DAC33_FIFO_WPTR_MSB && i <= DAC33_FIFO_IRQ_FLAG) || + i == DAC33_DAC_STATUS_FLAGS || + i == DAC33_SRC_EST_REF_CLK_RATIO_A || + i == DAC33_SRC_EST_REF_CLK_RATIO_B) + continue; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + } + for (i = DAC33_LDAC_PWR_CTRL; i <= DAC33_LINEL_TO_LLO_VOL; i++) { + data[0] = i; + data[1] = cache[i]; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + } + for (i = DAC33_LINER_TO_RLO_VOL; i <= DAC33_OSC_TRIM; i++) { + data[0] = i; + data[1] = cache[i]; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + } +} + +static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) +{ + u8 reg; + + reg = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + if (power) + reg |= DAC33_PDNALLB; + else + reg &= ~DAC33_PDNALLB; + dac33_write(codec, DAC33_PWR_CTRL, reg); +} + +static void dac33_hard_power(struct snd_soc_codec *codec, int power) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + + mutex_lock(&dac33->mutex); + if (power) { + if (dac33->power_gpio >= 0) { + gpio_set_value(dac33->power_gpio, 1); + dac33->chip_power = 1; + /* Restore registers */ + dac33_restore_regs(codec); + } + dac33_soft_power(codec, 1); + } else { + dac33_soft_power(codec, 0); + if (dac33->power_gpio >= 0) { + gpio_set_value(dac33->power_gpio, 0); + dac33->chip_power = 0; + } + } + mutex_unlock(&dac33->mutex); + +} + +static int dac33_get_nsample(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + + ucontrol->value.integer.value[0] = dac33->nsample; + + return 0; +} + +static int dac33_set_nsample(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret = 0; + + if (dac33->nsample == ucontrol->value.integer.value[0]) + return 0; + + if (ucontrol->value.integer.value[0] < dac33->nsample_min || + ucontrol->value.integer.value[0] > dac33->nsample_max) + ret = -EINVAL; + else + dac33->nsample = ucontrol->value.integer.value[0]; + + return ret; +} + +static int dac33_get_nsample_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + + ucontrol->value.integer.value[0] = dac33->nsample_switch; + + return 0; +} + +static int dac33_set_nsample_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret = 0; + + if (dac33->nsample_switch == ucontrol->value.integer.value[0]) + return 0; + /* Do not allow changes while stream is running*/ + if (codec->active) + return -EPERM; + + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] > 1) + ret = -EINVAL; + else + dac33->nsample_switch = ucontrol->value.integer.value[0]; + + return ret; +} + +/* + * DACL/R digital volume control: + * from 0 dB to -63.5 in 0.5 dB steps + * Need to be inverted later on: + * 0x00 == 0 dB + * 0x7f == -63.5 dB + */ +static DECLARE_TLV_DB_SCALE(dac_digivol_tlv, -6350, 50, 0); + +static const struct snd_kcontrol_new dac33_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC Digital Playback Volume", + DAC33_LDAC_DIG_VOL_CTRL, DAC33_RDAC_DIG_VOL_CTRL, + 0, 0x7f, 1, dac_digivol_tlv), + SOC_DOUBLE_R("DAC Digital Playback Switch", + DAC33_LDAC_DIG_VOL_CTRL, DAC33_RDAC_DIG_VOL_CTRL, 7, 1, 1), + SOC_DOUBLE_R("Line to Line Out Volume", + DAC33_LINEL_TO_LLO_VOL, DAC33_LINER_TO_RLO_VOL, 0, 127, 1), + SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0, + dac33_get_nsample, dac33_set_nsample), + SOC_SINGLE_EXT("nSample Switch", 0, 0, 1, 0, + dac33_get_nsample_switch, dac33_set_nsample_switch), +}; + +/* Analog bypass */ +static const struct snd_kcontrol_new dac33_dapm_abypassl_control = + SOC_DAPM_SINGLE("Switch", DAC33_LINEL_TO_LLO_VOL, 7, 1, 1); + +static const struct snd_kcontrol_new dac33_dapm_abypassr_control = + SOC_DAPM_SINGLE("Switch", DAC33_LINER_TO_RLO_VOL, 7, 1, 1); + +static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("LEFT_LO"), + SND_SOC_DAPM_OUTPUT("RIGHT_LO"), + + SND_SOC_DAPM_INPUT("LINEL"), + SND_SOC_DAPM_INPUT("LINER"), + + SND_SOC_DAPM_DAC("DACL", "Left Playback", DAC33_LDAC_PWR_CTRL, 2, 0), + SND_SOC_DAPM_DAC("DACR", "Right Playback", DAC33_RDAC_PWR_CTRL, 2, 0), + + /* Analog bypass */ + SND_SOC_DAPM_SWITCH("Analog Left Bypass", SND_SOC_NOPM, 0, 0, + &dac33_dapm_abypassl_control), + SND_SOC_DAPM_SWITCH("Analog Right Bypass", SND_SOC_NOPM, 0, 0, + &dac33_dapm_abypassr_control), + + SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Left Amp Power", + DAC33_OUT_AMP_PWR_CTRL, 6, 3, 3, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amp Power", + DAC33_OUT_AMP_PWR_CTRL, 4, 3, 3, 0), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Analog bypass */ + {"Analog Left Bypass", "Switch", "LINEL"}, + {"Analog Right Bypass", "Switch", "LINER"}, + + {"Output Left Amp Power", NULL, "DACL"}, + {"Output Right Amp Power", NULL, "DACR"}, + + {"Output Left Amp Power", NULL, "Analog Left Bypass"}, + {"Output Right Amp Power", NULL, "Analog Right Bypass"}, + + /* output */ + {"LEFT_LO", NULL, "Output Left Amp Power"}, + {"RIGHT_LO", NULL, "Output Right Amp Power"}, +}; + +static int dac33_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dac33_dapm_widgets, + ARRAY_SIZE(dac33_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int dac33_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + dac33_soft_power(codec, 1); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + dac33_soft_power(codec, 0); + break; + case SND_SOC_BIAS_OFF: + break; + } + codec->bias_level = level; + + return 0; +} + +static void dac33_work(struct work_struct *work) +{ + struct snd_soc_codec *codec; + struct tlv320dac33_priv *dac33; + u8 reg; + + dac33 = container_of(work, struct tlv320dac33_priv, work); + codec = &dac33->codec; + + mutex_lock(&dac33->mutex); + switch (dac33->state) { + case DAC33_PREFILL: + dac33->state = DAC33_PLAYBACK; + dac33_write16(codec, DAC33_NSAMPLE_MSB, + DAC33_THRREG(dac33->nsample)); + dac33_write16(codec, DAC33_PREFILL_MSB, + DAC33_THRREG(dac33->alarm_threshold)); + break; + case DAC33_PLAYBACK: + dac33_write16(codec, DAC33_NSAMPLE_MSB, + DAC33_THRREG(dac33->nsample)); + break; + case DAC33_IDLE: + break; + case DAC33_FLUSH: + /* flush fifo */ + dac33->state = DAC33_IDLE; + reg = dac33_read_reg_cache(codec, DAC33_FIFO_CTRL_A); + reg |= DAC33_FIFOFLUSH; + dac33_write(codec, DAC33_FIFO_CTRL_A, reg); + break; + } + mutex_unlock(&dac33->mutex); +} + +static irqreturn_t dac33_interrupt_handler(int irq, void *dev) +{ + struct snd_soc_codec *codec = dev; + struct tlv320dac33_priv *dac33 = codec->private_data; + + if (dac33->state == DAC33_PLAYBACK) + queue_work(dac33->dac33_wq, &dac33->work); + + return IRQ_HANDLED; +} + +static void dac33_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + unsigned int pwr_ctrl; + + /* Stop pending workqueue */ + if (dac33->nsample_switch) + cancel_work_sync(&dac33->work); + + mutex_lock(&dac33->mutex); + pwr_ctrl = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + pwr_ctrl &= ~(DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB); + dac33_write(codec, DAC33_PWR_CTRL, pwr_ctrl); + mutex_unlock(&dac33->mutex); +} + +static void dac33_oscwait(struct snd_soc_codec *codec) +{ + int timeout = 20; + u8 reg; + + do { + msleep(1); + dac33_read(codec, DAC33_INT_OSC_STATUS, &reg); + } while (((reg & 0x03) != DAC33_OSCSTATUS_NORMAL) && timeout--); + if ((reg & 0x03) != DAC33_OSCSTATUS_NORMAL) + dev_err(codec->dev, + "internal oscillator calibration failed\n"); +} + +static int dac33_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + /* Check parameters for validity */ + switch (params_rate(params)) { + case 44100: + case 48000: + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", + params_rate(params)); + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + dev_err(codec->dev, "unsupported format %d\n", + params_format(params)); + return -EINVAL; + } + + return 0; +} + +#define CALC_OSCSET(rate, refclk) ( \ + ((((rate * 10000) / refclk) * 4096) + 5000) / 10000) +#define CALC_RATIOSET(rate, refclk) ( \ + ((((refclk * 100000) / rate) * 16384) + 50000) / 100000) + +/* + * tlv320dac33 is strict on the sequence of the register writes, if the register + * writes happens in different order, than dac33 might end up in unknown state. + * Use the known, working sequence of register writes to initialize the dac33. + */ +static int dac33_prepare_chip(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + unsigned int oscset, ratioset, pwr_ctrl, reg_tmp; + u8 aictrl_a, fifoctrl_a; + + switch (substream->runtime->rate) { + case 44100: + case 48000: + oscset = CALC_OSCSET(substream->runtime->rate, dac33->refclk); + ratioset = CALC_RATIOSET(substream->runtime->rate, + dac33->refclk); + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", + substream->runtime->rate); + return -EINVAL; + } + + + aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A); + aictrl_a &= ~(DAC33_NCYCL_MASK | DAC33_WLEN_MASK); + fifoctrl_a = dac33_read_reg_cache(codec, DAC33_FIFO_CTRL_A); + fifoctrl_a &= ~DAC33_WIDTH; + switch (substream->runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + aictrl_a |= (DAC33_NCYCL_16 | DAC33_WLEN_16); + fifoctrl_a |= DAC33_WIDTH; + break; + default: + dev_err(codec->dev, "unsupported format %d\n", + substream->runtime->format); + return -EINVAL; + } + + mutex_lock(&dac33->mutex); + dac33_soft_power(codec, 1); + + reg_tmp = dac33_read_reg_cache(codec, DAC33_INT_OSC_CTRL); + dac33_write(codec, DAC33_INT_OSC_CTRL, reg_tmp); + + /* Write registers 0x08 and 0x09 (MSB, LSB) */ + dac33_write16(codec, DAC33_INT_OSC_FREQ_RAT_A, oscset); + + /* calib time: 128 is a nice number ;) */ + dac33_write(codec, DAC33_CALIB_TIME, 128); + + /* adjustment treshold & step */ + dac33_write(codec, DAC33_INT_OSC_CTRL_B, DAC33_ADJTHRSHLD(2) | + DAC33_ADJSTEP(1)); + + /* div=4 / gain=1 / div */ + dac33_write(codec, DAC33_INT_OSC_CTRL_C, DAC33_REFDIV(4)); + + pwr_ctrl = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + pwr_ctrl |= DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB; + dac33_write(codec, DAC33_PWR_CTRL, pwr_ctrl); + + dac33_oscwait(codec); + + if (dac33->nsample_switch) { + /* 50-51 : ASRC Control registers */ + dac33_write(codec, DAC33_ASRC_CTRL_A, (1 << 4)); /* div=2 */ + dac33_write(codec, DAC33_ASRC_CTRL_B, 1); /* ??? */ + + /* Write registers 0x34 and 0x35 (MSB, LSB) */ + dac33_write16(codec, DAC33_SRC_REF_CLK_RATIO_A, ratioset); + + /* Set interrupts to high active */ + dac33_write(codec, DAC33_INTP_CTRL_A, DAC33_INTPM_AHIGH); + + dac33_write(codec, DAC33_FIFO_IRQ_MODE_B, + DAC33_ATM(DAC33_FIFO_IRQ_MODE_LEVEL)); + dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT); + } else { + /* 50-51 : ASRC Control registers */ + dac33_write(codec, DAC33_ASRC_CTRL_A, DAC33_SRCBYP); + dac33_write(codec, DAC33_ASRC_CTRL_B, 0); /* ??? */ + } + + if (dac33->nsample_switch) + fifoctrl_a &= ~DAC33_FBYPAS; + else + fifoctrl_a |= DAC33_FBYPAS; + dac33_write(codec, DAC33_FIFO_CTRL_A, fifoctrl_a); + + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); + reg_tmp = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + if (dac33->nsample_switch) + reg_tmp &= ~DAC33_BCLKON; + else + reg_tmp |= DAC33_BCLKON; + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, reg_tmp); + + if (dac33->nsample_switch) { + /* 20: BCLK divide ratio */ + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3); + + dac33_write16(codec, DAC33_ATHR_MSB, + DAC33_THRREG(dac33->alarm_threshold)); + } else { + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32); + } + + mutex_unlock(&dac33->mutex); + + return 0; +} + +static void dac33_calculate_times(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + unsigned int nsample_limit; + + /* Number of samples (16bit, stereo) in one period */ + dac33->nsample_min = snd_pcm_lib_period_bytes(substream) / 4; + + /* Number of samples (16bit, stereo) in ALSA buffer */ + dac33->nsample_max = snd_pcm_lib_buffer_bytes(substream) / 4; + /* Subtract one period from the total */ + dac33->nsample_max -= dac33->nsample_min; + + /* Number of samples for LATENCY_TIME_MS / 2 */ + dac33->alarm_threshold = substream->runtime->rate / + (1000 / (LATENCY_TIME_MS / 2)); + + /* Find and fix up the lowest nsmaple limit */ + nsample_limit = substream->runtime->rate / (1000 / LATENCY_TIME_MS); + + if (dac33->nsample_min < nsample_limit) + dac33->nsample_min = nsample_limit; + + if (dac33->nsample < dac33->nsample_min) + dac33->nsample = dac33->nsample_min; + + /* + * Find and fix up the highest nsmaple limit + * In order to not overflow the DAC33 buffer substract the + * alarm_threshold value from the size of the DAC33 buffer + */ + nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - dac33->alarm_threshold; + + if (dac33->nsample_max > nsample_limit) + dac33->nsample_max = nsample_limit; + + if (dac33->nsample > dac33->nsample_max) + dac33->nsample = dac33->nsample_max; +} + +static int dac33_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + dac33_calculate_times(substream); + dac33_prepare_chip(substream); + + return 0; +} + +static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (dac33->nsample_switch) { + dac33->state = DAC33_PREFILL; + queue_work(dac33->dac33_wq, &dac33->work); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (dac33->nsample_switch) { + dac33->state = DAC33_FLUSH; + queue_work(dac33->dac33_wq, &dac33->work); + } + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int dac33_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 tlv320dac33_priv *dac33 = codec->private_data; + u8 ioc_reg, asrcb_reg; + + ioc_reg = dac33_read_reg_cache(codec, DAC33_INT_OSC_CTRL); + asrcb_reg = dac33_read_reg_cache(codec, DAC33_ASRC_CTRL_B); + switch (clk_id) { + case TLV320DAC33_MCLK: + ioc_reg |= DAC33_REFSEL; + asrcb_reg |= DAC33_SRCREFSEL; + break; + case TLV320DAC33_SLEEPCLK: + ioc_reg &= ~DAC33_REFSEL; + asrcb_reg &= ~DAC33_SRCREFSEL; + break; + default: + dev_err(codec->dev, "Invalid clock ID (%d)\n", clk_id); + break; + } + dac33->refclk = freq; + + dac33_write_reg_cache(codec, DAC33_INT_OSC_CTRL, ioc_reg); + dac33_write_reg_cache(codec, DAC33_ASRC_CTRL_B, asrcb_reg); + + return 0; +} + +static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 aictrl_a, aictrl_b; + + aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A); + aictrl_b = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* Codec Master */ + aictrl_a |= (DAC33_MSBCLK | DAC33_MSWCLK); + break; + case SND_SOC_DAIFMT_CBS_CFS: + /* Codec Slave */ + aictrl_a &= ~(DAC33_MSBCLK | DAC33_MSWCLK); + break; + default: + return -EINVAL; + } + + aictrl_a &= ~DAC33_AFMT_MASK; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aictrl_a |= DAC33_AFMT_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + aictrl_a |= DAC33_AFMT_DSP; + aictrl_b &= ~DAC33_DATA_DELAY_MASK; + aictrl_b |= DAC33_DATA_DELAY(1); /* 1 bit delay */ + break; + case SND_SOC_DAIFMT_DSP_B: + aictrl_a |= DAC33_AFMT_DSP; + aictrl_b &= ~DAC33_DATA_DELAY_MASK; /* No delay */ + break; + case SND_SOC_DAIFMT_RIGHT_J: + aictrl_a |= DAC33_AFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + aictrl_a |= DAC33_AFMT_LEFT_J; + break; + default: + dev_err(codec->dev, "Unsupported format (%u)\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + dac33_write_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); + dac33_write_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B, aictrl_b); + + return 0; +} + +static void dac33_init_chip(struct snd_soc_codec *codec) +{ + /* 44-46: DAC Control Registers */ + /* A : DAC sample rate Fsref/1.5 */ + dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(1)); + /* B : DAC src=normal, not muted */ + dac33_write(codec, DAC33_DAC_CTRL_B, DAC33_DACSRCR_RIGHT | + DAC33_DACSRCL_LEFT); + /* C : (defaults) */ + dac33_write(codec, DAC33_DAC_CTRL_C, 0x00); + + /* 64-65 : L&R DAC power control + Line In -> OUT 1V/V Gain, DAC -> OUT 4V/V Gain*/ + dac33_write(codec, DAC33_LDAC_PWR_CTRL, DAC33_LROUT_GAIN(2)); + dac33_write(codec, DAC33_RDAC_PWR_CTRL, DAC33_LROUT_GAIN(2)); + + /* 73 : volume soft stepping control, + clock source = internal osc (?) */ + dac33_write(codec, DAC33_ANA_VOL_SOFT_STEP_CTRL, DAC33_VOLCLKEN); + + /* 66 : LOP/LOM Modes */ + dac33_write(codec, DAC33_OUT_AMP_CM_CTRL, 0xff); + + /* 68 : LOM inverted from LOP */ + dac33_write(codec, DAC33_OUT_AMP_CTRL, (3<<2)); + + dac33_write(codec, DAC33_PWR_CTRL, DAC33_PDNALLB); +} + +static int dac33_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct tlv320dac33_priv *dac33; + int ret = 0; + + BUG_ON(!tlv320dac33_codec); + + codec = tlv320dac33_codec; + socdev->card->codec = codec; + dac33 = codec->private_data; + + /* Power up the codec */ + dac33_hard_power(codec, 1); + /* Set default configuration */ + dac33_init_chip(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms\n"); + goto pcm_err; + } + + snd_soc_add_controls(codec, dac33_snd_controls, + ARRAY_SIZE(dac33_snd_controls)); + dac33_add_widgets(codec); + + /* power on device */ + dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card\n"); + goto card_err; + } + + return 0; +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + dac33_hard_power(codec, 0); + return ret; +} + +static int dac33_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +static int dac33_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->card->codec; + + dac33_hard_power(codec, 0); + dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int dac33_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dac33_hard_power(codec, 1); + + dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + dac33_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_tlv320dac33 = { + .probe = dac33_soc_probe, + .remove = dac33_soc_remove, + .suspend = dac33_soc_suspend, + .resume = dac33_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320dac33); + +#define DAC33_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) +#define DAC33_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static struct snd_soc_dai_ops dac33_dai_ops = { + .shutdown = dac33_shutdown, + .hw_params = dac33_hw_params, + .prepare = dac33_pcm_prepare, + .trigger = dac33_pcm_trigger, + .set_sysclk = dac33_set_dai_sysclk, + .set_fmt = dac33_set_dai_fmt, +}; + +struct snd_soc_dai dac33_dai = { + .name = "tlv320dac33", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = DAC33_RATES, + .formats = DAC33_FORMATS,}, + .ops = &dac33_dai_ops, +}; +EXPORT_SYMBOL_GPL(dac33_dai); + +static int dac33_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct tlv320dac33_platform_data *pdata; + struct tlv320dac33_priv *dac33; + struct snd_soc_codec *codec; + int ret = 0; + + if (client->dev.platform_data == NULL) { + dev_err(&client->dev, "Platform data not set\n"); + return -ENODEV; + } + pdata = (struct tlv320dac33_platform_data *)client->dev.platform_data; + + dac33 = kzalloc(sizeof(struct tlv320dac33_priv), GFP_KERNEL); + if (dac33 == NULL) + return -ENOMEM; + + codec = &dac33->codec; + codec->private_data = dac33; + codec->control_data = client; + + mutex_init(&codec->mutex); + mutex_init(&dac33->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "tlv320dac33"; + codec->owner = THIS_MODULE; + codec->read = dac33_read_reg_cache; + codec->write = dac33_write_locked; + codec->hw_write = (hw_write_t) i2c_master_send; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = dac33_set_bias_level; + codec->dai = &dac33_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(dac33_reg); + codec->reg_cache = kmemdup(dac33_reg, ARRAY_SIZE(dac33_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto error_reg; + } + + i2c_set_clientdata(client, dac33); + + dac33->power_gpio = pdata->power_gpio; + dac33->irq = client->irq; + dac33->nsample = NSAMPLE_MAX; + /* Disable FIFO use by default */ + dac33->nsample_switch = 0; + + tlv320dac33_codec = codec; + + /* Setup work queue */ + dac33->dac33_wq = create_rt_workqueue("tlv320dac33"); + if (dac33->dac33_wq == NULL) { + ret = -ENOMEM; + goto error_wq; + } + + INIT_WORK(&dac33->work, dac33_work); + + codec->dev = &client->dev; + dac33_dai.dev = codec->dev; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto error_codec; + } + + ret = snd_soc_register_dai(&dac33_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + goto error_codec; + } + + /* Check if the reset GPIO number is valid and request it */ + if (dac33->power_gpio >= 0) { + ret = gpio_request(dac33->power_gpio, "tlv320dac33 reset"); + if (ret < 0) { + dev_err(codec->dev, + "Failed to request reset GPIO (%d)\n", + dac33->power_gpio); + snd_soc_unregister_dai(&dac33_dai); + snd_soc_unregister_codec(codec); + goto error_codec; + } + gpio_direction_output(dac33->power_gpio, 0); + } else { + dac33->chip_power = 1; + } + + /* Check if the IRQ number is valid and request it */ + if (dac33->irq >= 0) { + ret = request_irq(dac33->irq, dac33_interrupt_handler, + IRQF_TRIGGER_RISING | IRQF_DISABLED, + codec->name, codec); + if (ret < 0) { + dev_err(codec->dev, "Could not request IRQ%d (%d)\n", + dac33->irq, ret); + dac33->irq = -1; + } + } + + /* Shut down the codec for now */ + dac33_hard_power(codec, 0); + + return ret; + +error_codec: + destroy_workqueue(dac33->dac33_wq); +error_wq: + kfree(codec->reg_cache); +error_reg: + tlv320dac33_codec = NULL; + kfree(dac33); + + return ret; +} + +static int dac33_i2c_remove(struct i2c_client *client) +{ + struct tlv320dac33_priv *dac33; + + dac33 = i2c_get_clientdata(client); + dac33_hard_power(&dac33->codec, 0); + + if (dac33->power_gpio >= 0) + gpio_free(dac33->power_gpio); + if (dac33->irq >= 0) + free_irq(dac33->irq, &dac33->codec); + + destroy_workqueue(dac33->dac33_wq); + snd_soc_unregister_dai(&dac33_dai); + snd_soc_unregister_codec(&dac33->codec); + kfree(dac33->codec.reg_cache); + kfree(dac33); + tlv320dac33_codec = NULL; + + return 0; +} + +static const struct i2c_device_id tlv320dac33_i2c_id[] = { + { + .name = "tlv320dac33", + .driver_data = 0, + }, + { }, +}; + +static struct i2c_driver tlv320dac33_i2c_driver = { + .driver = { + .name = "tlv320dac33", + .owner = THIS_MODULE, + }, + .probe = dac33_i2c_probe, + .remove = __devexit_p(dac33_i2c_remove), + .id_table = tlv320dac33_i2c_id, +}; + +static int __init dac33_module_init(void) +{ + int r; + r = i2c_add_driver(&tlv320dac33_i2c_driver); + if (r < 0) { + printk(KERN_ERR "DAC33: driver registration failed\n"); + return r; + } + return 0; +} +module_init(dac33_module_init); + +static void __exit dac33_module_exit(void) +{ + i2c_del_driver(&tlv320dac33_i2c_driver); +} +module_exit(dac33_module_exit); + + +MODULE_DESCRIPTION("ASoC TLV320DAC33 codec driver"); +MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi(a)nokia.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320dac33.h b/sound/soc/codecs/tlv320dac33.h new file mode 100644 index 0000000..0fedd70 --- /dev/null +++ b/sound/soc/codecs/tlv320dac33.h @@ -0,0 +1,267 @@ +/* + * ALSA SoC Texas Instruments TLV320DAC33 codec driver + * + * Author: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> + * + * Copyright: (C) 2009 Nokia Corporation + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TLV320DAC33_H +#define __TLV320DAC33_H + +#define DAC33_PAGE_SELECT 0x00 +#define DAC33_PWR_CTRL 0x01 +#define DAC33_PLL_CTRL_A 0x02 +#define DAC33_PLL_CTRL_B 0x03 +#define DAC33_PLL_CTRL_C 0x04 +#define DAC33_PLL_CTRL_D 0x05 +#define DAC33_PLL_CTRL_E 0x06 +#define DAC33_INT_OSC_CTRL 0x07 +#define DAC33_INT_OSC_FREQ_RAT_A 0x08 +#define DAC33_INT_OSC_FREQ_RAT_B 0x09 +#define DAC33_INT_OSC_DAC_RATIO_SET 0x0A +#define DAC33_CALIB_TIME 0x0B +#define DAC33_INT_OSC_CTRL_B 0x0C +#define DAC33_INT_OSC_CTRL_C 0x0D +#define DAC33_INT_OSC_STATUS 0x0E +#define DAC33_INT_OSC_DAC_RATIO_READ 0x0F +#define DAC33_INT_OSC_FREQ_RAT_READ_A 0x10 +#define DAC33_INT_OSC_FREQ_RAT_READ_B 0x11 +#define DAC33_SER_AUDIOIF_CTRL_A 0x12 +#define DAC33_SER_AUDIOIF_CTRL_B 0x13 +#define DAC33_SER_AUDIOIF_CTRL_C 0x14 +#define DAC33_FIFO_CTRL_A 0x15 +#define DAC33_UTHR_MSB 0x16 +#define DAC33_UTHR_LSB 0x17 +#define DAC33_ATHR_MSB 0x18 +#define DAC33_ATHR_LSB 0x19 +#define DAC33_LTHR_MSB 0x1A +#define DAC33_LTHR_LSB 0x1B +#define DAC33_PREFILL_MSB 0x1C +#define DAC33_PREFILL_LSB 0x1D +#define DAC33_NSAMPLE_MSB 0x1E +#define DAC33_NSAMPLE_LSB 0x1F +#define DAC33_FIFO_WPTR_MSB 0x20 +#define DAC33_FIFO_WPTR_LSB 0x21 +#define DAC33_FIFO_RPTR_MSB 0x22 +#define DAC33_FIFO_RPTR_LSB 0x23 +#define DAC33_FIFO_DEPTH_MSB 0x24 +#define DAC33_FIFO_DEPTH_LSB 0x25 +#define DAC33_SAMPLES_REMAINING_MSB 0x26 +#define DAC33_SAMPLES_REMAINING_LSB 0x27 +#define DAC33_FIFO_IRQ_FLAG 0x28 +#define DAC33_FIFO_IRQ_MASK 0x29 +#define DAC33_FIFO_IRQ_MODE_A 0x2A +#define DAC33_FIFO_IRQ_MODE_B 0x2B +#define DAC33_DAC_CTRL_A 0x2C +#define DAC33_DAC_CTRL_B 0x2D +#define DAC33_DAC_CTRL_C 0x2E +#define DAC33_LDAC_DIG_VOL_CTRL 0x2F +#define DAC33_RDAC_DIG_VOL_CTRL 0x30 +#define DAC33_DAC_STATUS_FLAGS 0x31 +#define DAC33_ASRC_CTRL_A 0x32 +#define DAC33_ASRC_CTRL_B 0x33 +#define DAC33_SRC_REF_CLK_RATIO_A 0x34 +#define DAC33_SRC_REF_CLK_RATIO_B 0x35 +#define DAC33_SRC_EST_REF_CLK_RATIO_A 0x36 +#define DAC33_SRC_EST_REF_CLK_RATIO_B 0x37 +#define DAC33_INTP_CTRL_A 0x38 +#define DAC33_INTP_CTRL_B 0x39 +/* Registers 0x3A - 0x3F Reserved */ +#define DAC33_LDAC_PWR_CTRL 0x40 +#define DAC33_RDAC_PWR_CTRL 0x41 +#define DAC33_OUT_AMP_CM_CTRL 0x42 +#define DAC33_OUT_AMP_PWR_CTRL 0x43 +#define DAC33_OUT_AMP_CTRL 0x44 +#define DAC33_LINEL_TO_LLO_VOL 0x45 +/* Registers 0x45 - 0x47 Reserved */ +#define DAC33_LINER_TO_RLO_VOL 0x48 +#define DAC33_ANA_VOL_SOFT_STEP_CTRL 0x49 +#define DAC33_OSC_TRIM 0x4A +/* Registers 0x4B - 0x7C Reserved */ +#define DAC33_DEVICE_ID_MSB 0x7D +#define DAC33_DEVICE_ID_LSB 0x7E +#define DAC33_DEVICE_REV_ID 0x7F + +#define DAC33_CACHEREGNUM 128 + +/* Bit definitions */ + +/* DAC33_PWR_CTRL (0x01) */ +#define DAC33_DACRPDNB (0x01 << 0) +#define DAC33_DACLPDNB (0x01 << 1) +#define DAC33_OSCPDNB (0x01 << 2) +#define DAC33_PLLPDNB (0x01 << 3) +#define DAC33_PDNALLB (0x01 << 4) +#define DAC33_SOFT_RESET (0x01 << 7) + +/* DAC33_INT_OSC_CTRL (0x07) */ +#define DAC33_REFSEL (0x01 << 1) + +/* DAC33_INT_OSC_CTRL_B (0x0C) */ +#define DAC33_ADJSTEP(x) (x << 0) +#define DAC33_ADJTHRSHLD(x) (x << 4) + +/* DAC33_INT_OSC_CTRL_C (0x0D) */ +#define DAC33_REFDIV(x) (x << 4) + +/* DAC33_INT_OSC_STATUS (0x0E) */ +#define DAC33_OSCSTATUS_IDLE_CALIB (0x00) +#define DAC33_OSCSTATUS_NORMAL (0x01) +#define DAC33_OSCSTATUS_ADJUSTMENT (0x03) +#define DAC33_OSCSTATUS_NOT_USED (0x02) + +/* DAC33_SER_AUDIOIF_CTRL_A (0x12) */ +#define DAC33_MSWCLK (0x01 << 0) +#define DAC33_MSBCLK (0x01 << 1) +#define DAC33_AFMT_MASK (0x03 << 2) +#define DAC33_AFMT_I2S (0x00 << 2) +#define DAC33_AFMT_DSP (0x01 << 2) +#define DAC33_AFMT_RIGHT_J (0x02 << 2) +#define DAC33_AFMT_LEFT_J (0x03 << 2) +#define DAC33_WLEN_MASK (0x03 << 4) +#define DAC33_WLEN_16 (0x00 << 4) +#define DAC33_WLEN_20 (0x01 << 4) +#define DAC33_WLEN_24 (0x02 << 4) +#define DAC33_WLEN_32 (0x03 << 4) +#define DAC33_NCYCL_MASK (0x03 << 6) +#define DAC33_NCYCL_16 (0x00 << 6) +#define DAC33_NCYCL_20 (0x01 << 6) +#define DAC33_NCYCL_24 (0x02 << 6) +#define DAC33_NCYCL_32 (0x03 << 6) + +/* DAC33_SER_AUDIOIF_CTRL_B (0x13) */ +#define DAC33_DATA_DELAY_MASK (0x03 << 2) +#define DAC33_DATA_DELAY(x) (x << 2) +#define DAC33_BCLKON (0x01 << 5) + +/* DAC33_FIFO_CTRL_A (0x15) */ +#define DAC33_WIDTH (0x01 << 0) +#define DAC33_FBYPAS (0x01 << 1) +#define DAC33_FAUTO (0x01 << 2) +#define DAC33_FIFOFLUSH (0x01 << 3) + +/* + * UTHR, ATHR, LTHR, PREFILL, NSAMPLE (0x16 - 0x1F) + * 13-bit values +*/ +#define DAC33_THRREG(x) (((x) & 0x1FFF) << 3) + +/* DAC33_FIFO_IRQ_MASK (0x29) */ +#define DAC33_MNS (0x01 << 0) +#define DAC33_MPS (0x01 << 1) +#define DAC33_MAT (0x01 << 2) +#define DAC33_MLT (0x01 << 3) +#define DAC33_MUT (0x01 << 4) +#define DAC33_MUF (0x01 << 5) +#define DAC33_MOF (0x01 << 6) + +#define DAC33_FIFO_IRQ_MODE_MASK (0x03) +#define DAC33_FIFO_IRQ_MODE_RISING (0x00) +#define DAC33_FIFO_IRQ_MODE_FALLING (0x01) +#define DAC33_FIFO_IRQ_MODE_LEVEL (0x02) +#define DAC33_FIFO_IRQ_MODE_EDGE (0x03) + +/* DAC33_FIFO_IRQ_MODE_A (0x2A) */ +#define DAC33_UTM(x) (x << 0) +#define DAC33_UFM(x) (x << 2) +#define DAC33_OFM(x) (x << 4) + +/* DAC33_FIFO_IRQ_MODE_B (0x2B) */ +#define DAC33_NSM(x) (x << 0) +#define DAC33_PSM(x) (x << 2) +#define DAC33_ATM(x) (x << 4) +#define DAC33_LTM(x) (x << 4) + +/* DAC33_DAC_CTRL_A (0x2C) */ +#define DAC33_DACRATE(x) (x << 0) +#define DAC33_DACDUAL (0x01 << 4) +#define DAC33_DACLKSEL_MASK (0x03 << 5) +#define DAC33_DACLKSEL_INTSOC (0x00 << 5) +#define DAC33_DACLKSEL_PLL (0x01 << 5) +#define DAC33_DACLKSEL_MCLK (0x02 << 5) +#define DAC33_DACLKSEL_BCLK (0x03 << 5) + +/* DAC33_DAC_CTRL_B (0x2D) */ +#define DAC33_DACSRCR_MASK (0x03 << 0) +#define DAC33_DACSRCR_MUTE (0x00 << 0) +#define DAC33_DACSRCR_RIGHT (0x01 << 0) +#define DAC33_DACSRCR_LEFT (0x02 << 0) +#define DAC33_DACSRCR_MONOMIX (0x03 << 0) +#define DAC33_DACSRCL_MASK (0x03 << 2) +#define DAC33_DACSRCL_MUTE (0x00 << 2) +#define DAC33_DACSRCL_LEFT (0x01 << 2) +#define DAC33_DACSRCL_RIGHT (0x02 << 2) +#define DAC33_DACSRCL_MONOMIX (0x03 << 2) +#define DAC33_DVOLSTEP_MASK (0x03 << 4) +#define DAC33_DVOLSTEP_SS_PERFS (0x00 << 4) +#define DAC33_DVOLSTEP_SS_PER2FS (0x01 << 4) +#define DAC33_DVOLSTEP_SS_DISABLED (0x02 << 4) +#define DAC33_DVOLCTRL_MASK (0x03 << 6) +#define DAC33_DVOLCTRL_LR_INDEPENDENT1 (0x00 << 6) +#define DAC33_DVOLCTRL_LR_RIGHT_CONTROL (0x01 << 6) +#define DAC33_DVOLCTRL_LR_LEFT_CONTROL (0x02 << 6) +#define DAC33_DVOLCTRL_LR_INDEPENDENT2 (0x03 << 6) + +/* DAC33_DAC_CTRL_C (0x2E) */ +#define DAC33_DEEMENR (0x01 << 0) +#define DAC33_EFFENR (0x01 << 1) +#define DAC33_DEEMENL (0x01 << 2) +#define DAC33_EFFENL (0x01 << 3) +#define DAC33_EN3D (0x01 << 4) +#define DAC33_RESYNMUTE (0x01 << 5) +#define DAC33_RESYNEN (0x01 << 6) + +/* DAC33_ASRC_CTRL_A (0x32) */ +#define DAC33_SRCBYP (0x01 << 0) +#define DAC33_SRCLKSEL_MASK (0x03 << 1) +#define DAC33_SRCLKSEL_INTSOC (0x00 << 1) +#define DAC33_SRCLKSEL_PLL (0x01 << 1) +#define DAC33_SRCLKSEL_MCLK (0x02 << 1) +#define DAC33_SRCLKSEL_BCLK (0x03 << 1) +#define DAC33_SRCLKDIV(x) (x << 3) + +/* DAC33_ASRC_CTRL_B (0x33) */ +#define DAC33_SRCSETUP(x) (x << 0) +#define DAC33_SRCREFSEL (0x01 << 4) +#define DAC33_SRCREFDIV(x) (x << 5) + +/* DAC33_INTP_CTRL_A (0x38) */ +#define DAC33_INTPSEL (0x01 << 0) +#define DAC33_INTPM_MASK (0x03 << 1) +#define DAC33_INTPM_ALOW_OPENDRAIN (0x00 << 1) +#define DAC33_INTPM_ALOW (0x01 << 1) +#define DAC33_INTPM_AHIGH (0x02 << 1) + +/* DAC33_LDAC_PWR_CTRL (0x40) */ +/* DAC33_RDAC_PWR_CTRL (0x41) */ +#define DAC33_DACLRNUM (0x01 << 2) +#define DAC33_LROUT_GAIN(x) (x << 0) + +/* DAC33_ANA_VOL_SOFT_STEP_CTRL (0x49) */ +#define DAC33_VOLCLKSEL (0x01 << 0) +#define DAC33_VOLCLKEN (0x01 << 1) +#define DAC33_VOLBYPASS (0x01 << 2) + +#define TLV320DAC33_MCLK 0 +#define TLV320DAC33_SLEEPCLK 1 + +extern struct snd_soc_dai dac33_dai; +extern struct snd_soc_codec_device soc_codec_dev_tlv320dac33; + +#endif /* __TLV320DAC33_H */ -- 1.6.5

3 8

[alsa-devel] [PATCHv2 5/7] ASoC: TWL6030: Add support for low-power mode
by Lopez Cruz, Misael 14 Oct '09

14 Oct '09

TWL6030 codec supports two power modes: low-power and high performance. In low-power mode, headset downlink must be the only path enabled and components in that path (headset DAC and driver) should be in that mode too. In this mode, codec can stream audio at 44.1 and 48 kHz if sys clock from CLK32K is configured to 17.64 and 19.2 MHz using low-power PLL, respectively. In high-performance mode, codec can only work at 19.2 MHz from high-performance PLL. All paths supported in the codec can be used but audio can be streamed only at 48 kHz. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 214 ++++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/twl6030.h | 17 ++++ 2 files changed, 202 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index f1e333f..032619d 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -46,6 +46,7 @@ struct twl6030_data { struct snd_soc_codec codec; int codec_powered; + unsigned int sysclk; }; /* @@ -305,6 +306,29 @@ static void twl6030_power_down(struct snd_soc_codec *codec) mdelay(10); } +/* set headset dac and driver power mode */ +static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) +{ + int hslctl, hsrctl; + int mask = TWL6030_HSDRVMODEL | TWL6030_HSDACMODEL; + + hslctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSLCTL); + hsrctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSRCTL); + + if (high_perf) { + hslctl &= ~mask; + hsrctl &= ~mask; + } else { + hslctl |= mask; + hsrctl |= mask; + } + + twl6030_write(codec, TWL6030_REG_HSLCTL, hslctl); + twl6030_write(codec, TWL6030_REG_HSRCTL, hsrctl); + + return 0; +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -598,6 +622,7 @@ static int twl6030_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct twl6030_data *priv = codec->private_data; int rate, format; /* hardware dai (McPDM) requires bit stream of twice @@ -607,7 +632,20 @@ static int twl6030_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); switch (rate) { case 44100: + if (priv->sysclk != 17640000) { + dev_err(codec->dev, + "rate %d not supported at current sysclk %d\n", + rate, priv->sysclk); + return -EINVAL; + } + break; case 48000: + if (priv->sysclk != 19200000) { + dev_err(codec->dev, + "rate %d not supported at current sysclk %d\n", + rate, priv->sysclk); + return -EINVAL; + } break; default: dev_err(codec->dev, "unknown rate %d\n", rate); @@ -630,46 +668,163 @@ static int twl6030_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 twl6030_data *priv = codec->private_data; u8 hppll, lppll; + lppll = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); hppll = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); - hppll &= TWL6030_HPLLRST; - - switch (freq) { - case 12000000: - /* MCLK input, PLL enabled */ - hppll = TWL6030_MCLK_12000KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLENA; - break; - case 19200000: - /* MCLK input, PLL disabled */ - hppll = TWL6030_MCLK_19200KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLBP; + + switch (clk_id) { + case TWL6030_SYSCLK_SEL_LPPLL: + if (freq != 32768) { + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* CLK32K input requires low-power PLL */ + lppll |= TWL6030_LPLLENA | TWL6030_LPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + mdelay(5); + lppll &= ~TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* headset dac and driver must be in low-power mode */ + headset_power_mode(codec, 0); + + hppll &= ~TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); break; - case 26000000: - /* MCLK input, PLL enabled */ - hppll = TWL6030_MCLK_26000KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLENA; + case TWL6030_SYSCLK_SEL_HPPLL: + switch (freq) { + case 12000000: + hppll = TWL6030_MCLK_12000KHZ; + break; + case 26000000: + hppll = TWL6030_MCLK_26000KHZ; + break; + default: + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* 12 and 26 MHz freqs require high-performance PLL */ + hppll |= TWL6030_HPLLSQRBP | TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + udelay(500); + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + lppll |= TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + lppll &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* only 19.2 MHz can be generated by HPPLL */ + priv->sysclk = 19200000; break; - case 38400000: - /* clk slicer input, PLL disabled */ - hppll = TWL6030_MCLK_38400KHZ - | TWL6030_HPLLSQRENA - | TWL6030_HPLLBP; + case TWL6030_SYSCLK_SEL_MCLK: + switch (freq) { + case 19200000: + hppll = TWL6030_MCLK_19200KHZ | TWL6030_HPLLSQRBP; + break; + case 38400000: + hppll = TWL6030_MCLK_38400KHZ; + break; + default: + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* 19.2 and 38.4 MHz freqs don't require PLL */ + hppll |= TWL6030_HPLLBP; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + udelay(500); + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + lppll |= TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + lppll &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* only 19.2 MHz can be generated by MCLK */ + priv->sysclk = 19200000; break; default: - dev_err(codec->dev, "unknown sysclk rate %d\n", freq); + dev_err(codec->dev, "unknown clk_id %d\n", clk_id); return -EINVAL; } - twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + return 0; +} + +static int twl6030_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct twl6030_data *priv = codec->private_data; + int div; + u8 ldoctl, lppllctl, hppllctl; + + ldoctl = twl6030_read_reg_cache(codec, TWL6030_REG_LDOCTL); + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + hppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); + + if (freq_in && freq_out) { + /* div = round(freq_out / freq_in) */ + div = (freq_out + (freq_in >> 2)) / freq_in; + + switch (pll_id) { + case TWL6030_LPPLL_ID: + if (div < 512) { + dev_err(codec->dev, "invalid pll div %d\n", + div); + return -EINVAL; + } + twl6030_write(codec, TWL6030_REG_LPPLLDIV, div - 512); + lppllctl |= TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + + priv->sysclk = freq_out; + break; + case TWL6030_HPPLL_ID: + hppllctl |= TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); + break; + default: + dev_err(codec->dev, "unknown pll id %d\n", pll_id); + return -EINVAL; + } - /* Disable LPPLL and select HPPLL */ - lppll = TWL6030_HPLLSEL; - twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + /* switch to pll */ + ldoctl &= ~TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + } else { + /* switch to internal oscillator */ + ldoctl |= TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + + /* turn-off pll */ + switch (pll_id) { + case TWL6030_HPPLL_ID: + hppllctl &= ~TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); + break; + case TWL6030_LPPLL_ID: + lppllctl &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + break; + default: + dev_err(codec->dev, "unknown pll id %d\n", pll_id); + return -EINVAL; + } + + priv->sysclk = 0; + } return 0; } @@ -689,6 +844,7 @@ static int twl6030_set_dai_fmt(struct snd_soc_dai *codec_dai, static struct snd_soc_dai_ops twl6030_dai_ops = { .hw_params = twl6030_hw_params, .set_sysclk = twl6030_set_dai_sysclk, + .set_pll = twl6030_set_dai_pll, .set_fmt = twl6030_set_dai_fmt, }; diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 7375ae8..15d3e1b 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -92,6 +92,10 @@ #define TWL6030_MCLK_38400KHZ (3 << 5) #define TWL6030_MCLK_MSK 0x60 +#define TWL6030_SYSCLK_SEL_LPPLL 1 +#define TWL6030_SYSCLK_SEL_HPPLL 2 +#define TWL6030_SYSCLK_SEL_MCLK 3 + /* LPPLLCTL (0x08) fields */ #define TWL6030_LPLLENA 0x01 @@ -100,6 +104,19 @@ #define TWL6030_LPLLFIN 0x08 #define TWL6030_HPLLSEL 0x10 +#define TWL6030_HPPLL_ID 1 +#define TWL6030_LPPLL_ID 2 + +/* HSLCTL (0x10) fields */ + +#define TWL6030_HSDACMODEL 0x02 +#define TWL6030_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6030_HSDACMODER 0x02 +#define TWL6030_HSDRVMODER 0x08 + /* ACCCTL (0x2D) fields */ #define TWL6030_RESETSPLIT 0x04 -- 1.5.4.3

3 6

[alsa-devel] [PATCH 09/10] [PATCH 04/10] ASoC: finally enable support for eXeda and CM-X300
by Mike Rapoport 14 Oct '09

14 Oct '09

From: Igor Grinberg <grinberg(a)compulab.co.il> Signed-off-by: Igor Grinberg <grinberg(a)compulab.co.il> Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> CC: Mark Brown <broonie(a)opensource.wolfsonmicro.com> CC: alsa-devel(a)alsa-project.org --- sound/soc/pxa/Kconfig | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index dcb3181..d4f4031 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -90,7 +90,8 @@ config SND_PXA2XX_SOC_E800 config SND_PXA2XX_SOC_EM_X270 tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" - depends on SND_PXA2XX_SOC && MACH_EM_X270 + depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ + MACH_CM_X300) select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help -- 1.6.0.6

2 1

[alsa-devel] [PATCH 1/1] ALSA: HDA VIA: Remove 48k sample rate limit for S/PDIF
by Logan Li 14 Oct '09

14 Oct '09

From: Logan Li <loganli(a)viatech.com.cn> Subject: ALSA: HDA VIA: Remove 48k sample rate limit for S/PDIF 48 kHz limit is for slightly better stability, and sample rates other than 48k (like 96k/192k) are for better sound quality. We choose better quality, so remove the 48k limit. Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 4 ---- 1 file changed, 4 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -4626,7 +4626,6 @@ .substreams = 2, .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -5124,7 +5123,6 @@ .substreams = 2, .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -5561,7 +5559,6 @@ .substreams = 1, .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -5914,7 +5911,6 @@ .substreams = 1, .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open,

2 1

[alsa-devel] Which transfer mode to use
by Till Crueger 14 Oct '09

14 Oct '09

Hi, First of all, I am fairly new to this list, as well as sound development. So if this list is the wrong place for this question, it would be nice if you could point me in the right direction. I am trying to use ALSA in my OCaml [1] programms. As far as I could see there are no well documented bindings to ALSA for this language, so I am stuck with writing my own bindings. I have gotten to the point where I can open a pcm device with a minimal configuration space on my desktop as well as on my laptop. However I am not sure which transfer mode to use, since it is hard to see for me, what the advantages and disadvantages would be. Here is what I have found out so far. 1. Asynchronous callback: This method seems to be the best explained by all the tutorials I could find, so I am guessing this is the best choice for audio applications. However for my purpose this seems hard to do, since the thread which contains the callback must never enter the OCaml side of the wrapper. This constraint is due to the OCaml garbage collector, which AFAIK would be messed up severely. So to use this method I would have to write an extra buffer in the C side, to which I write the data from OCaml. Depending on the way this buffer is set up -- I was thinking of using a linked list to contain all the single packets -- I would have to keep track what part of the usage of this buffer to be able to synchronise anything to the sound. Not the best choice I as far as I can see. 2. read/write: So far this seems the best choice. This way I could probably rewrite the asynchronous callback method on the OCaml side. For this however I would have to figure out after each polling, how much data can be written. I had a look at the API documentation and it seems ALSA provides function to get this information. However when I had a look at the example sine-wave generator [2], those functions were only used by the mmap method. So the question would be if this information could still be provided when using the read/write method. 3. mmap: During the research I did for this project I found some posts on the portaudio list, that said this method could not be used with all soundcards. Also I would have to figure out if the memory layout of the data passed from the OCaml side is at all compatible to this method. If it is incompatible the use of mmap does not seem to make sense at all. So far this are the points I have gathered to help me make a choice. As you can see I already have a clear preference, but at this point I am still not quite sure. I also found some links that hinted at other issues that would have to be taken into consideration. For example I found some concerns about differences in lattency between these three methods. So what is your opinion on this. Do you think reintroducing the async. callback on the OCaml side would be a good idea, or would you rather use the second buffer that is accesible from the OCaml side. As I said, I haven't done any real sound development, so I am really new to this. Thanks for your help, Till [1] OCaml is a functional language developed by Inria. http://caml.inria.fr/ [2] http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html

2 1

[alsa-devel] Mic/Line-in record is not working with wm9713 AC97
by Sreekanth W 14 Oct '09

14 Oct '09

Hi, I am trying to record a wav file using mic/line-in with wm9713 driver. The Wav file is created, only header is created but almost there is no data. This is seen using CoolEdit/Vim editor. If I try to play the recorded file, there is no sound. But, playing works for other existing files, able to using aplay. I am issuing the following command to record # ./arecord –t wav –c 1 –r 8000 –d 10 –f S16_LE test.wav The below is the dump of /sys/devices/platform/soc-audio/dapm_widget, while "arecord" is running (using mic): # cat /sys/devices/platform/soc-audio/dapm_widget Mic Bias: Off Mic B Pre Amp: Off Mic A Pre Amp: On Mic B PGA: Off Mic A PGA: Off Mono In: Off Right Line In: Off Left Line In: Off Mono Out: Off Out 4: Off Out 3: Off Right Speaker: Off Left Speaker: Off Right Headphone: Off Left Headphone: Off Right Voice ADC: Off Left Voice ADC: Off Right HiFi ADC: On Left HiFi ADC: On Right ADC: On Left ADC: On Aux DAC: Off Voice DAC: Off Capture Mixer: Off Line Mixer: Off HP Mixer: Off AC97 Mixer: Off Right DAC: Off Left DAC: Off Speaker Mixer: Off Mono Mixer: Off Right HP Mixer: Off Left HP Mixer: Off PM State: On I'm using DMA for data transfer, and getting the callback function after DMA completed. Also I'm printing the number of Bytes transfered. Any idea why record is not working? Anything wrong with these settings? Regards, Sreekanth

2 1

[alsa-devel] Channel conversion in plughw not working for my driver... confused about snd_pcm_plug_hw_params()
by David Huggins-Daines 14 Oct '09

14 Oct '09

Hi, I'm trying to make mono output work on the audio driver for the Cirrus EDB9302 evaluation board. For some reason, plughw is refusing to do any channel conversion. When I try to play a mono file, I simply get: aplay: set_params:1022: Unable to install hw params: The driver's struct snd_pcm_hardware contains: .channels_min = 2, .channels_max = 2, which I believe is the correct way to tell userspace "this driver can only do stereo output" - at least, this is how intel8x0.c does it, and that driver really ought to work by now! I've traced this to the exact point of failure, and what I see is that in snd_pcm_plug_hw_params(), despite figuring out exactly what the set of hardware parameters will work for the slave device, it goes and calls _snd_pcm_hw_params() with the actual parameters requested by the client, which obviously fails, and leads to the error shown above. I added some calls to snd_pcm_hw_params_dump() which give this output. The first is sparams, which is the refined set of parameters which I assume are supposed to be set in the slave device. The second is params, which is the set of parameters requested by the client (aplay in this case). sparams are: ACCESS: RW_INTERLEAVED FORMAT: U8 SUBFORMAT: STD SAMPLE_BITS: 8 FRAME_BITS: 16 CHANNELS: 2 RATE: 11025 PERIOD_TIME: (124988 124989) PERIOD_SIZE: 1378 PERIOD_BYTES: 2756 PERIODS: (4 5) BUFFER_TIME: (500045 500046) BUFFER_SIZE: 5513 BUFFER_BYTES: 11026 TICK_TIME: 0 params are: ACCESS: RW_INTERLEAVED FORMAT: U8 SUBFORMAT: STD SAMPLE_BITS: 8 FRAME_BITS: 8 CHANNELS: 1 RATE: 11025 PERIOD_TIME: (124988 124989) PERIOD_SIZE: 1378 PERIOD_BYTES: 1378 PERIODS: (4 5) BUFFER_TIME: (500045 500046) BUFFER_SIZE: 5513 BUFFER_BYTES: 5513 TICK_TIME: 0 inserting plugins whatever that means in snd_pcm_plug_change_channels! So, yeah, that's great, it figured it all out, and realized that it had to do channel conversion, which presumably is done in snd_pcm_plug_change_channels(). But now, we see that it goes ahead and tries to set these parameters in the slave device: calling _snd_pcm_hw_params with params: ACCESS: RW_INTERLEAVED FORMAT: U8 SUBFORMAT: STD SAMPLE_BITS: 8 FRAME_BITS: 8 CHANNELS: 1 RATE: 11025 PERIOD_TIME: (124988 124989) PERIOD_SIZE: 1378 PERIOD_BYTES: 1378 PERIODS: (4 5) BUFFER_TIME: (500045 500046) BUFFER_SIZE: 5513 BUFFER_BYTES: 5513 TICK_TIME: 0 But that's wrong! The slave device can't do single-channel input! So what gives? This can't possibly be broken, or nobody's Intel on-board audio would ever work at all... so what is my driver doing wrong?

1 0

[alsa-devel] SPDIF with X-Fi
by Daniel L. Miller 14 Oct '09

14 Oct '09

Perhaps I'm going about this the wrong way (if so, please tell me), but... My X-Fi card is now working with the new driver. I am connected to my amplifier via a fiber optic cable. Sending standard two-channel audio to the "front" alsa plug gets directed to the spdif output and sounds great. But I want to have my full surround active - particularly for movies already encoded with dts and/or ac-3. Do I need to specially configure ALSA is any fashion to make this work? Do I need to use the iecset tool? I'm using Kubuntu Karmic, running a 2.6.31 kernel, if that makes any difference. -- Daniel

2 1

[alsa-devel] [PATCH 1/1] Echoaudio: fix broken Mia driver
by Jordan 14 Oct '09

14 Oct '09

2 1