March 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ASoC: remove a card from the list, if instantiation failed
by Guennadi Liakhovetski 19 Mar '10

19 Mar '10

If instantiation of a card failed, we still have to remove it from the card list on unregistration. This fixes an Oops on Migo-R, triggering, when after a failed firmware load attempt the driver modules are removed and re-inserted again. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- Mark, is this more like what you mean? diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c8b0556..f63b59a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1232,26 +1232,25 @@ static int soc_remove(struct platform_device *pdev) struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - if (!card->instantiated) - return 0; + if (card->instantiated) { + run_delayed_work(&card->delayed_work); - run_delayed_work(&card->delayed_work); + if (platform->remove) + platform->remove(pdev); - if (platform->remove) - platform->remove(pdev); + if (codec_dev->remove) + codec_dev->remove(pdev); - if (codec_dev->remove) - codec_dev->remove(pdev); + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + if (cpu_dai->remove) + cpu_dai->remove(pdev, cpu_dai); + } - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; - if (cpu_dai->remove) - cpu_dai->remove(pdev, cpu_dai); + if (card->remove) + card->remove(pdev); } - if (card->remove) - card->remove(pdev); - snd_soc_unregister_card(card); return 0;

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[alsa-devel] Are ALSA io plugins implemented as threads or does ALSA use fork() ?
by Stefan Schoenleitner 19 Mar '10

19 Mar '10

Hello, from what I have learned so far ALSA io plugins are implemented as shared libraries with their entry point defined with the SND_PCM_PLUGIN_DEFINE_FUNC() macro. However, each plugin instance can only work for either the SND_PCM_STREAM_PLAYBACK or SND_PCM_STREAM_CAPTURE stream. Hence this is why io plugins supporting playback and capture usually set up their callback functions depending on what type of stream is used. For example in the pulseaudio io plugin, this is done with: pcm->io.callback = stream == SND_PCM_STREAM_PLAYBACK ? &pulse_playback_callback : &pulse_capture_callback; * If such a plugin is used for capture AND playback at the same time, it seems that two instances of the plugin need to be running, right ? * Are these instances running as separate processes (i.e. fork()) or are they implemented as threads ? How is this implemented internally in the ALSA code ? In my plugin I'm planning that both the playback and the capture plugin instances talk to a common data processing thread over shared buffers. The reason for this is that the data processing threads needs to have access to the capture and playback samples *at the same time*. I'm guessing that the plugin instances run with different PIDs so that I would need to use common IPC mechanisms in order to let them communicate with each other, right ? Hence my first approach would be to set up a common shared memory together with a semaphore so that only one process can access the shared memory at a time. Any better ideas ? cheers, Stefan

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[alsa-devel] [PATCH v2 2/4] change bf5xx-ad1938 machine driver to bf5xx-ad193x machine driver
by Barry Song 19 Mar '10

19 Mar '10

Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/Kconfig | 9 ++- sound/soc/blackfin/Makefile | 4 +- .../blackfin/{bf5xx-ad1938.c => bf5xx-ad193x.c} | 66 ++++++++++---------- 3 files changed, 40 insertions(+), 39 deletions(-) rename sound/soc/blackfin/{bf5xx-ad1938.c => bf5xx-ad193x.c} (65%) diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 97f1a25..8ef2502 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -49,13 +49,14 @@ config SND_BF5XX_SOC_AD1836 help Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. -config SND_BF5XX_SOC_AD1938 - tristate "SoC AD1938 Audio support for Blackfin" +config SND_BF5XX_SOC_AD193X + tristate "SoC AD193X Audio support for Blackfin" depends on SND_BF5XX_TDM select SND_BF5XX_SOC_TDM - select SND_SOC_AD1938 + select SND_SOC_AD193X help - Say Y if you want to add support for AD1938 codec on Blackfin. + Say Y if you want to add support for AD193X codec on Blackfin. + This driver supports AD1936, AD1937, AD1938 and AD1939. config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 87e3042..49af3f3 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -20,10 +20,10 @@ snd-ad1836-objs := bf5xx-ad1836.o snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o snd-ad73311-objs := bf5xx-ad73311.o -snd-ad1938-objs := bf5xx-ad1938.o +snd-ad193x-objs := bf5xx-ad193x.o obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o -obj-$(CONFIG_SND_BF5XX_SOC_AD1938) += snd-ad1938.o +obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad193x.c similarity index 65% rename from sound/soc/blackfin/bf5xx-ad1938.c rename to sound/soc/blackfin/bf5xx-ad193x.c index 2ef1e50..b8c9060 100644 --- a/sound/soc/blackfin/bf5xx-ad1938.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -1,9 +1,9 @@ /* - * File: sound/soc/blackfin/bf5xx-ad1938.c + * File: sound/soc/blackfin/bf5xx-ad193x.c * Author: Barry Song <Barry.Song(a)analog.com> * * Created: Thur June 4 2009 - * Description: Board driver for ad1938 sound chip + * Description: Board driver for ad193x sound chip * * Bugs: Enter bugs at http://blackfin.uclinux.org/ * @@ -38,15 +38,15 @@ #include <asm/dma.h> #include <asm/portmux.h> -#include "../codecs/ad1938.h" +#include "../codecs/ad193x.h" #include "bf5xx-sport.h" #include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" -static struct snd_soc_card bf5xx_ad1938; +static struct snd_soc_card bf5xx_ad193x; -static int bf5xx_ad1938_startup(struct snd_pcm_substream *substream) +static int bf5xx_ad193x_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -55,7 +55,7 @@ static int bf5xx_ad1938_startup(struct snd_pcm_substream *substream) return 0; } -static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, +static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -89,61 +89,61 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops bf5xx_ad1938_ops = { - .startup = bf5xx_ad1938_startup, - .hw_params = bf5xx_ad1938_hw_params, +static struct snd_soc_ops bf5xx_ad193x_ops = { + .startup = bf5xx_ad193x_startup, + .hw_params = bf5xx_ad193x_hw_params, }; -static struct snd_soc_dai_link bf5xx_ad1938_dai = { - .name = "ad1938", - .stream_name = "AD1938", +static struct snd_soc_dai_link bf5xx_ad193x_dai = { + .name = "ad193x", + .stream_name = "AD193X", .cpu_dai = &bf5xx_tdm_dai, - .codec_dai = &ad1938_dai, - .ops = &bf5xx_ad1938_ops, + .codec_dai = &ad193x_dai, + .ops = &bf5xx_ad193x_ops, }; -static struct snd_soc_card bf5xx_ad1938 = { - .name = "bf5xx_ad1938", +static struct snd_soc_card bf5xx_ad193x = { + .name = "bf5xx_ad193x", .platform = &bf5xx_tdm_soc_platform, - .dai_link = &bf5xx_ad1938_dai, + .dai_link = &bf5xx_ad193x_dai, .num_links = 1, }; -static struct snd_soc_device bf5xx_ad1938_snd_devdata = { - .card = &bf5xx_ad1938, - .codec_dev = &soc_codec_dev_ad1938, +static struct snd_soc_device bf5xx_ad193x_snd_devdata = { + .card = &bf5xx_ad193x, + .codec_dev = &soc_codec_dev_ad193x, }; -static struct platform_device *bfxx_ad1938_snd_device; +static struct platform_device *bfxx_ad193x_snd_device; -static int __init bf5xx_ad1938_init(void) +static int __init bf5xx_ad193x_init(void) { int ret; - bfxx_ad1938_snd_device = platform_device_alloc("soc-audio", -1); - if (!bfxx_ad1938_snd_device) + bfxx_ad193x_snd_device = platform_device_alloc("soc-audio", -1); + if (!bfxx_ad193x_snd_device) return -ENOMEM; - platform_set_drvdata(bfxx_ad1938_snd_device, &bf5xx_ad1938_snd_devdata); - bf5xx_ad1938_snd_devdata.dev = &bfxx_ad1938_snd_device->dev; - ret = platform_device_add(bfxx_ad1938_snd_device); + platform_set_drvdata(bfxx_ad193x_snd_device, &bf5xx_ad193x_snd_devdata); + bf5xx_ad193x_snd_devdata.dev = &bfxx_ad193x_snd_device->dev; + ret = platform_device_add(bfxx_ad193x_snd_device); if (ret) - platform_device_put(bfxx_ad1938_snd_device); + platform_device_put(bfxx_ad193x_snd_device); return ret; } -static void __exit bf5xx_ad1938_exit(void) +static void __exit bf5xx_ad193x_exit(void) { - platform_device_unregister(bfxx_ad1938_snd_device); + platform_device_unregister(bfxx_ad193x_snd_device); } -module_init(bf5xx_ad1938_init); -module_exit(bf5xx_ad1938_exit); +module_init(bf5xx_ad193x_init); +module_exit(bf5xx_ad193x_exit); /* Module information */ MODULE_AUTHOR("Barry Song"); -MODULE_DESCRIPTION("ALSA SoC AD1938 board driver"); +MODULE_DESCRIPTION("ALSA SoC AD193X board driver"); MODULE_LICENSE("GPL"); -- 1.5.6.3

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[alsa-devel] [PATCHv6 2/2][RESEND] ASoC: TWL6040: Add twl6040 codec driver
by Olaya, Margarita 19 Mar '10

19 Mar '10

From: Misael Lopez Cruz <x0052729(a)ti.com> Initial version of TWL6040 codec driver. The TWL6040 codec uses a proprietary PDM-based digital audio interface. Audio paths supported are: - Input: Main Mic, Sub Mic, Headset Mic, Auxiliary-FM Left/Right - Output: Headset Left/Right, Handsfree Left/Right TWL6040 codec supports power-up/down manual and automatic sequence. Manual sequence is done through a specific register writes sequence. Automatic sequence is done when the codec is powered-up through the external AUDPWRON line. The completion of the sequence is signaled through the audio interrupt. TWL6040 codec sysclk can be provided by: low-power or high performance PLL: - The low-power PLL takes a low-frequency input at 32,768 Hz and generates an approximate of 17.64 or 19.2 MHz (for 44.1 KHz and 48 KHz respectively) - The high-performance PLL generates an exact 19.2 MHz clock signal from high-frequency input at 12/19.2/26/38.4 MHz. Low-power playback mode is a special scenario where only headset path (headset DAC and driver) is active. For the particular case of headset path, PLL being used defines the headset power mode: low-power, high-performance. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6040.c | 1227 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 141 +++++ 4 files changed, 1374 insertions(+), 0 deletions(-) create mode 100755 sound/soc/codecs/twl6040.c create mode 100644 sound/soc/codecs/twl6040.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 16c47ed..398cbb0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,6 +35,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_TWL6040 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM2000 if I2C @@ -168,6 +169,9 @@ config SND_SOC_TWL4030 select TWL4030_CODEC tristate +config SND_SOC_TWL6040 + tristate + config SND_SOC_UDA134X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6981777..98bd10c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,7 @@ 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-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o @@ -85,6 +86,7 @@ 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-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c new file mode 100755 index 0000000..0e4dce7 --- /dev/null +++ b/sound/soc/codecs/twl6040.c @@ -0,0 +1,1227 @@ +/* + * ALSA SoC TWL6040 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * 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/gpio.h> +#include <linux/platform_device.h> +#include <linux/i2c/twl.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 "twl6040.h" + +#define TWL6040_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct twl6040_data { + struct snd_soc_codec codec; + int audpwron; + int naudint; + int codec_powered; + int pll; + int non_lp; + unsigned int sysclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct completion ready; +}; + +/* + * twl6040 register cache & default register settings + */ +static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { + 0x00, /* not used 0x00 */ + 0x4B, /* TWL6040_ASICID (ro) 0x01 */ + 0x00, /* TWL6040_ASICREV (ro) 0x02 */ + 0x00, /* TWL6040_INTID 0x03 */ + 0x00, /* TWL6040_INTMR 0x04 */ + 0x00, /* TWL6040_NCPCTRL 0x05 */ + 0x00, /* TWL6040_LDOCTL 0x06 */ + 0x60, /* TWL6040_HPPLLCTL 0x07 */ + 0x00, /* TWL6040_LPPLLCTL 0x08 */ + 0x4A, /* TWL6040_LPPLLDIV 0x09 */ + 0x00, /* TWL6040_AMICBCTL 0x0A */ + 0x00, /* TWL6040_DMICBCTL 0x0B */ + 0x18, /* TWL6040_MICLCTL 0x0C - No input selected on Left Mic */ + 0x18, /* TWL6040_MICRCTL 0x0D - No input selected on Right Mic */ + 0x00, /* TWL6040_MICGAIN 0x0E */ + 0x1B, /* TWL6040_LINEGAIN 0x0F */ + 0x00, /* TWL6040_HSLCTL 0x10 */ + 0x00, /* TWL6040_HSRCTL 0x11 */ + 0x00, /* TWL6040_HSGAIN 0x12 */ + 0x00, /* TWL6040_EARCTL 0x13 */ + 0x00, /* TWL6040_HFLCTL 0x14 */ + 0x00, /* TWL6040_HFLGAIN 0x15 */ + 0x00, /* TWL6040_HFRCTL 0x16 */ + 0x00, /* TWL6040_HFRGAIN 0x17 */ + 0x00, /* TWL6040_VIBCTLL 0x18 */ + 0x00, /* TWL6040_VIBDATL 0x19 */ + 0x00, /* TWL6040_VIBCTLR 0x1A */ + 0x00, /* TWL6040_VIBDATR 0x1B */ + 0x00, /* TWL6040_HKCTL1 0x1C */ + 0x00, /* TWL6040_HKCTL2 0x1D */ + 0x00, /* TWL6040_GPOCTL 0x1E */ + 0x00, /* TWL6040_ALB 0x1F */ + 0x00, /* TWL6040_DLB 0x20 */ + 0x00, /* not used 0x21 */ + 0x00, /* not used 0x22 */ + 0x00, /* not used 0x23 */ + 0x00, /* not used 0x24 */ + 0x00, /* not used 0x25 */ + 0x00, /* not used 0x26 */ + 0x00, /* not used 0x27 */ + 0x00, /* TWL6040_TRIM1 0x28 */ + 0x00, /* TWL6040_TRIM2 0x29 */ + 0x00, /* TWL6040_TRIM3 0x2A */ + 0x00, /* TWL6040_HSOTRIM 0x2B */ + 0x00, /* TWL6040_HFOTRIM 0x2C */ + 0x09, /* TWL6040_ACCCTL 0x2D */ + 0x00, /* TWL6040_STATUS (ro) 0x2E */ +}; + +/* + * twl6040 vio/gnd registers: + * registers under vio/gnd supply can be accessed + * before the power-up sequence, after NRESPWRON goes high + */ +static const int twl6040_vio_reg[TWL6040_VIOREGNUM] = { + TWL6040_REG_ASICID, + TWL6040_REG_ASICREV, + TWL6040_REG_INTID, + TWL6040_REG_INTMR, + TWL6040_REG_NCPCTL, + TWL6040_REG_LDOCTL, + TWL6040_REG_AMICBCTL, + TWL6040_REG_DMICBCTL, + TWL6040_REG_HKCTL1, + TWL6040_REG_HKCTL2, + TWL6040_REG_GPOCTL, + TWL6040_REG_TRIM1, + TWL6040_REG_TRIM2, + TWL6040_REG_TRIM3, + TWL6040_REG_HSOTRIM, + TWL6040_REG_HFOTRIM, + TWL6040_REG_ACCCTL, + TWL6040_REG_STATUS, +}; + +/* + * twl6040 vdd/vss registers: + * registers under vdd/vss supplies can only be accessed + * after the power-up sequence + */ +static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = { + TWL6040_REG_HPPLLCTL, + TWL6040_REG_LPPLLCTL, + TWL6040_REG_LPPLLDIV, + TWL6040_REG_MICLCTL, + TWL6040_REG_MICRCTL, + TWL6040_REG_MICGAIN, + TWL6040_REG_LINEGAIN, + TWL6040_REG_HSLCTL, + TWL6040_REG_HSRCTL, + TWL6040_REG_HSGAIN, + TWL6040_REG_EARCTL, + TWL6040_REG_HFLCTL, + TWL6040_REG_HFLGAIN, + TWL6040_REG_HFRCTL, + TWL6040_REG_HFRGAIN, + TWL6040_REG_VIBCTLL, + TWL6040_REG_VIBDATL, + TWL6040_REG_VIBCTLR, + TWL6040_REG_VIBDATR, + TWL6040_REG_ALB, + TWL6040_REG_DLB, +}; + +/* + * read twl6040 register cache + */ +static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + return cache[reg]; +} + +/* + * write twl6040 register cache + */ +static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6040_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * read from twl6040 hardware register + */ +static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &value, reg); + twl6040_write_reg_cache(codec, reg, value); + + return value; +} + +/* + * write to the twl6040 register space + */ +static int twl6040_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + twl6040_write_reg_cache(codec, reg, value); + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); +} + +static void twl6040_init_vio_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + /* allow registers to be accessed by i2c */ + twl6040_write(codec, TWL6040_REG_ACCCTL, cache[TWL6040_REG_ACCCTL]); + + for (i = 0; i < TWL6040_VIOREGNUM; i++) { + reg = twl6040_vio_reg[i]; + /* skip read-only registers (ASICID, ASICREV, STATUS) */ + switch (reg) { + case TWL6040_REG_ASICID: + case TWL6040_REG_ASICREV: + case TWL6040_REG_STATUS: + continue; + default: + break; + } + twl6040_write(codec, reg, cache[reg]); + } +} + +static void twl6040_init_vdd_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + for (i = 0; i < TWL6040_VDDREGNUM; i++) { + reg = twl6040_vdd_reg[i]; + twl6040_write(codec, reg, cache[reg]); + } +} + +/* twl6040 codec manual power-up sequence */ +static void twl6040_power_up(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL); + ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL); + + /* enable reference system */ + ldoctl |= TWL6040_REFENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + msleep(10); + /* enable internal oscillator */ + ldoctl |= TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(10); + /* enable high-side ldo */ + ldoctl |= TWL6040_HSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* enable negative charge pump */ + ncpctl |= TWL6040_NCPENA | TWL6040_NCPOPEN; + twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl); + udelay(488); + /* enable low-side ldo */ + ldoctl |= TWL6040_LSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* enable low-power pll */ + lppllctl |= TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + /* reset state machine */ + accctl |= TWL6040_RESETSPLIT; + twl6040_write(codec, TWL6040_REG_ACCCTL, accctl); + mdelay(5); + accctl &= ~TWL6040_RESETSPLIT; + twl6040_write(codec, TWL6040_REG_ACCCTL, accctl); + /* disable internal oscillator */ + ldoctl &= ~TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); +} + +/* twl6040 codec manual power-down sequence */ +static void twl6040_power_down(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL); + ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL); + + /* enable internal oscillator */ + ldoctl |= TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(10); + /* disable low-power pll */ + lppllctl &= ~TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + /* disable low-side ldo */ + ldoctl &= ~TWL6040_LSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* disable negative charge pump */ + ncpctl &= ~(TWL6040_NCPENA | TWL6040_NCPOPEN); + twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl); + udelay(488); + /* disable high-side ldo */ + ldoctl &= ~TWL6040_HSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* disable internal oscillator */ + ldoctl &= ~TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + /* disable reference system */ + ldoctl &= ~TWL6040_REFENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + msleep(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 = TWL6040_HSDRVMODEL | TWL6040_HSDACMODEL; + + hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + + if (high_perf) { + hslctl &= ~mask; + hsrctl &= ~mask; + } else { + hslctl |= mask; + hsrctl |= mask; + } + + twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl); + twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl); + + return 0; +} + +static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct twl6040_data *priv = codec->private_data; + + if (SND_SOC_DAPM_EVENT_ON(event)) + priv->non_lp++; + else + priv->non_lp--; + + return 0; +} + +/* audio interrupt handler */ +static irqreturn_t twl6040_naudint_handler(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct twl6040_data *priv = codec->private_data; + u8 intid; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); + + switch (intid) { + case TWL6040_THINT: + dev_alert(codec->dev, "die temp over-limit detection\n"); + break; + case TWL6040_PLUGINT: + case TWL6040_UNPLUGINT: + case TWL6040_HOOKINT: + break; + case TWL6040_HFINT: + dev_alert(codec->dev, "hf drivers over current detection\n"); + break; + case TWL6040_VIBINT: + dev_alert(codec->dev, "vib drivers over current detection\n"); + break; + case TWL6040_READYINT: + complete(&priv->ready); + break; + default: + dev_err(codec->dev, "unknown audio interrupt %d\n", intid); + break; + } + + return IRQ_HANDLED; +} + +/* + * MICATT volume control: + * from -6 to 0 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); + +/* + * MICGAIN volume control: + * from 6 to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); + +/* + * HSGAIN volume control: + * from -30 to 0 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0); + +/* + * HFGAIN volume control: + * from -52 to 6 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0); + +/* Left analog microphone selection */ +static const char *twl6040_amicl_texts[] = + {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; + +/* Right analog microphone selection */ +static const char *twl6040_amicr_texts[] = + {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; + +static const struct soc_enum twl6040_enum[] = { + SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 3, twl6040_amicl_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 3, twl6040_amicr_texts), +}; + +static const struct snd_kcontrol_new amicl_control = + SOC_DAPM_ENUM("Route", twl6040_enum[0]); + +static const struct snd_kcontrol_new amicr_control = + SOC_DAPM_ENUM("Route", twl6040_enum[1]); + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new hsdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 5, 1, 0); + +static const struct snd_kcontrol_new hsdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 5, 1, 0); + +/* Handsfree DAC playback switches */ +static const struct snd_kcontrol_new hfdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hfdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0); + +/* Headset driver switches */ +static const struct snd_kcontrol_new hsl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hsr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0); + +/* Handsfree driver switches */ +static const struct snd_kcontrol_new hfl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0); + +static const struct snd_kcontrol_new hfr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0); + +static const struct snd_kcontrol_new twl6040_snd_controls[] = { + /* Capture gains */ + SOC_DOUBLE_TLV("Capture Preamplifier Volume", + TWL6040_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv), + SOC_DOUBLE_TLV("Capture Volume", + TWL6040_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + + /* Playback gains */ + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + +}; + +static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("AFML"), + SND_SOC_DAPM_INPUT("AFMR"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + + /* Analog input muxes for the capture amplifiers */ + SND_SOC_DAPM_MUX("Analog Left Capture Route", + SND_SOC_NOPM, 0, 0, &amicl_control), + SND_SOC_DAPM_MUX("Analog Right Capture Route", + SND_SOC_NOPM, 0, 0, &amicr_control), + + /* Analog capture PGAs */ + SND_SOC_DAPM_PGA("MicAmpL", + TWL6040_REG_MICLCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmpR", + TWL6040_REG_MICRCTL, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", + TWL6040_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", + TWL6040_REG_MICRCTL, 2, 0), + + /* Microphone bias */ + SND_SOC_DAPM_MICBIAS("Headset Mic Bias", + TWL6040_REG_AMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Main Mic Bias", + TWL6040_REG_AMICBCTL, 4, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias", + TWL6040_REG_DMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias", + TWL6040_REG_DMICBCTL, 4, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", + TWL6040_REG_HSLCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", + TWL6040_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", + TWL6040_REG_HFLCTL, 0, 0, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("HFDAC Right", "Handsfree Playback", + TWL6040_REG_HFRCTL, 0, 0, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Analog playback switches */ + SND_SOC_DAPM_SWITCH("HSDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hsdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HSDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hsdacr_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hfdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), + + SND_SOC_DAPM_SWITCH("Headset Left Driver", + SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Headset Right Driver", + SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), + SND_SOC_DAPM_SWITCH_E("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH_E("Handsfree Right Driver", + SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Analog playback PGAs */ + SND_SOC_DAPM_PGA("HFDAC Left PGA", + TWL6040_REG_HFLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HFDAC Right PGA", + TWL6040_REG_HFRCTL, 1, 0, NULL, 0), + +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Capture path */ + {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Left Capture Route", "Main Mic", "MAINMIC"}, + {"Analog Left Capture Route", "Aux/FM Left", "AFML"}, + + {"Analog Right Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Right Capture Route", "Sub Mic", "SUBMIC"}, + {"Analog Right Capture Route", "Aux/FM Right", "AFMR"}, + + {"MicAmpL", NULL, "Analog Left Capture Route"}, + {"MicAmpR", NULL, "Analog Right Capture Route"}, + + {"ADC Left", NULL, "MicAmpL"}, + {"ADC Right", NULL, "MicAmpR"}, + + /* Headset playback path */ + {"HSDAC Left Playback", "Switch", "HSDAC Left"}, + {"HSDAC Right Playback", "Switch", "HSDAC Right"}, + + {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, + {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + + {"HSOL", NULL, "Headset Left Driver"}, + {"HSOR", NULL, "Headset Right Driver"}, + + /* Handsfree playback path */ + {"HFDAC Left Playback", "Switch", "HFDAC Left"}, + {"HFDAC Right Playback", "Switch", "HFDAC Right"}, + + {"HFDAC Left PGA", NULL, "HFDAC Left Playback"}, + {"HFDAC Right PGA", NULL, "HFDAC Right Playback"}, + + {"Handsfree Left Driver", "Switch", "HFDAC Left PGA"}, + {"Handsfree Right Driver", "Switch", "HFDAC Right PGA"}, + + {"HFL", NULL, "Handsfree Left Driver"}, + {"HFR", NULL, "Handsfree Right Driver"}, +}; + +static int twl6040_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl6040_dapm_widgets, + ARRAY_SIZE(twl6040_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int twl6040_power_up_completion(struct snd_soc_codec *codec, + int naudint) +{ + struct twl6040_data *priv = codec->private_data; + int time_left; + u8 intid; + + time_left = wait_for_completion_timeout(&priv->ready, + msecs_to_jiffies(48)); + + if (!time_left) { + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, + TWL6040_REG_INTID); + if (!(intid & TWL6040_READYINT)) { + dev_err(codec->dev, "timeout waiting for READYINT\n"); + return -ETIMEDOUT; + } + } + + priv->codec_powered = 1; + + return 0; +} + +static int twl6040_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct twl6040_data *priv = codec->private_data; + int audpwron = priv->audpwron; + int naudint = priv->naudint; + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 1); + + /* wait for power-up completion */ + ret = twl6040_power_up_completion(codec, naudint); + if (ret) + return ret; + + /* sync registers updated during power-up sequence */ + twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LPPLLCTL); + } else { + /* use manual power-up sequence */ + twl6040_power_up(codec); + priv->codec_powered = 1; + } + + /* initialize vdd/vss registers with reg_cache */ + twl6040_init_vdd_regs(codec); + break; + case SND_SOC_BIAS_OFF: + if (!priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 0); + + /* power-down sequence latency */ + udelay(500); + + /* sync registers updated during power-down sequence */ + twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL); + twl6040_write_reg_cache(codec, TWL6040_REG_LPPLLCTL, + 0x00); + } else { + /* use manual power-down sequence */ + twl6040_power_down(codec); + } + + priv->codec_powered = 0; + break; + } + + codec->bias_level = level; + + return 0; +} + +/* set of rates for each pll: low-power and high-performance */ + +static unsigned int lp_rates[] = { + 88200, + 96000, +}; + +static struct snd_pcm_hw_constraint_list lp_constraints = { + .count = ARRAY_SIZE(lp_rates), + .list = lp_rates, +}; + +static unsigned int hp_rates[] = { + 96000, +}; + +static struct snd_pcm_hw_constraint_list hp_constraints = { + .count = ARRAY_SIZE(hp_rates), + .list = hp_rates, +}; + +static int twl6040_startup(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 twl6040_data *priv = codec->private_data; + + if (!priv->sysclk) { + dev_err(codec->dev, + "no mclk configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + /* + * capture is not supported at 17.64 MHz, + * it's reserved for headset low-power playback scenario + */ + if ((priv->sysclk == 17640000) && substream->stream) { + dev_err(codec->dev, + "capture mode is not supported at %dHz\n", + priv->sysclk); + return -EINVAL; + } + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + priv->sysclk_constraints); + + return 0; +} + +static int twl6040_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; + struct twl6040_data *priv = codec->private_data; + u8 lppllctl; + int rate; + + /* nothing to do for high-perf pll, it supports only 48 kHz */ + if (priv->pll == TWL6040_HPPLL_ID) + return 0; + + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + + rate = params_rate(params); + switch (rate) { + case 88200: + lppllctl |= TWL6040_LPLLFIN; + priv->sysclk = 17640000; + break; + case 96000: + lppllctl &= ~TWL6040_LPLLFIN; + priv->sysclk = 19200000; + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", rate); + return -EINVAL; + } + + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + return 0; +} + +static int twl6040_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 twl6040_data *priv = codec->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + /* + * low-power playback mode is restricted + * for headset path only + */ + if ((priv->sysclk == 17640000) && priv->non_lp) { + dev_err(codec->dev, + "some enabled paths aren't supported at %dHz\n", + priv->sysclk); + return -EPERM; + } + break; + default: + break; + } + + return 0; +} + +static int twl6040_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 twl6040_data *priv = codec->private_data; + u8 hppllctl, lppllctl; + + hppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_HPPLLCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + + switch (clk_id) { + case TWL6040_SYSCLK_SEL_LPPLL: + switch (freq) { + case 32768: + /* headset dac and driver must be in low-power mode */ + headset_power_mode(codec, 0); + + /* clk32k input requires low-power pll */ + lppllctl |= TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + mdelay(5); + lppllctl &= ~TWL6040_HPLLSEL; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + hppllctl &= ~TWL6040_HPLLENA; + twl6040_write(codec, TWL6040_REG_HPPLLCTL, hppllctl); + break; + default: + dev_err(codec->dev, "unknown mclk freq %d\n", freq); + return -EINVAL; + } + + /* lppll divider */ + switch (priv->sysclk) { + case 17640000: + lppllctl |= TWL6040_LPLLFIN; + break; + case 19200000: + lppllctl &= ~TWL6040_LPLLFIN; + break; + default: + /* sysclk not yet configured */ + lppllctl &= ~TWL6040_LPLLFIN; + priv->sysclk = 19200000; + break; + } + + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + priv->pll = TWL6040_LPPLL_ID; + priv->sysclk_constraints = &lp_constraints; + break; + case TWL6040_SYSCLK_SEL_HPPLL: + hppllctl &= ~TWL6040_MCLK_MSK; + + switch (freq) { + case 12000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6040_MCLK_12000KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLENA; + break; + case 19200000: + /* mclk input, pll disabled */ + hppllctl |= TWL6040_MCLK_19200KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLBP; + break; + case 26000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6040_MCLK_26000KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLENA; + break; + case 38400000: + /* clk slicer, pll disabled */ + hppllctl |= TWL6040_MCLK_38400KHZ | + TWL6040_HPLLSQRENA | + TWL6040_HPLLBP; + break; + default: + dev_err(codec->dev, "unknown mclk freq %d\n", freq); + return -EINVAL; + } + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + twl6040_write(codec, TWL6040_REG_HPPLLCTL, hppllctl); + udelay(500); + lppllctl |= TWL6040_HPLLSEL; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + lppllctl &= ~TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + /* high-performance pll can provide only 19.2 MHz */ + priv->pll = TWL6040_HPPLL_ID; + priv->sysclk = 19200000; + priv->sysclk_constraints = &hp_constraints; + break; + default: + dev_err(codec->dev, "unknown clk_id %d\n", clk_id); + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops twl6040_dai_ops = { + .startup = twl6040_startup, + .hw_params = twl6040_hw_params, + .trigger = twl6040_trigger, + .set_sysclk = twl6040_set_dai_sysclk, +}; + +struct snd_soc_dai twl6040_dai = { + .name = "twl6040", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}; +EXPORT_SYMBOL_GPL(twl6040_dai); + +#ifdef CONFIG_PM +static int twl6040_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; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl6040_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6040_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} +#else +#define twl6040_suspend NULL +#define twl6040_resume NULL +#endif + +static struct snd_soc_codec *twl6040_codec; + +static int twl6040_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + BUG_ON(!twl6040_codec); + + codec = twl6040_codec; + socdev->card->codec = codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; + } + + snd_soc_add_controls(codec, twl6040_snd_controls, + ARRAY_SIZE(twl6040_snd_controls)); + twl6040_add_widgets(codec); + + if (ret < 0) { + dev_err(&pdev->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + return ret; +} + +static int twl6040_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_twl6040 = { + .probe = twl6040_probe, + .remove = twl6040_remove, + .suspend = twl6040_suspend, + .resume = twl6040_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl6040); + +static int __devinit twl6040_codec_probe(struct platform_device *pdev) +{ + struct twl4030_codec_data *twl_codec = pdev->dev.platform_data; + struct snd_soc_codec *codec; + struct twl6040_data *priv; + int audpwron, naudint; + int ret = 0; + + priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + if (twl_codec) { + audpwron = twl_codec->audpwron_gpio; + naudint = twl_codec->naudint_irq; + } else { + audpwron = -EINVAL; + naudint = 0; + } + + priv->audpwron = audpwron; + priv->naudint = naudint; + + codec = &priv->codec; + codec->dev = &pdev->dev; + twl6040_dai.dev = &pdev->dev; + + codec->name = "twl6040"; + codec->owner = THIS_MODULE; + codec->read = twl6040_read_reg_cache; + codec->write = twl6040_write; + codec->set_bias_level = twl6040_set_bias_level; + codec->private_data = priv; + codec->dai = &twl6040_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(twl6040_reg); + codec->reg_cache = kmemdup(twl6040_reg, sizeof(twl6040_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + init_completion(&priv->ready); + + if (gpio_is_valid(audpwron)) { + ret = gpio_request(audpwron, "audpwron"); + if (ret) + goto gpio1_err; + + ret = gpio_direction_output(audpwron, 0); + if (ret) + goto gpio2_err; + + priv->codec_powered = 0; + } + + if (naudint) { + /* audio interrupt */ + ret = request_threaded_irq(naudint, NULL, + twl6040_naudint_handler, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "twl6040_codec", codec); + if (ret) + goto gpio2_err; + } else { + if (gpio_is_valid(audpwron)) { + /* enable only codec ready interrupt */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + ~TWL6040_READYMSK & TWL6040_ALLINT_MSK); + } else { + /* no interrupts at all */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + TWL6040_ALLINT_MSK); + } + } + + /* init vio registers */ + twl6040_init_vio_regs(codec); + + /* power on device */ + ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto irq_err; + + ret = snd_soc_register_codec(codec); + if (ret) + goto reg_err; + + twl6040_codec = codec; + + ret = snd_soc_register_dai(&twl6040_dai); + if (ret) + goto dai_err; + + return 0; + +dai_err: + snd_soc_unregister_codec(codec); + twl6040_codec = NULL; +reg_err: + twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); +irq_err: + if (naudint) + free_irq(naudint, codec); +gpio2_err: + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); +gpio1_err: + kfree(codec->reg_cache); +cache_err: + kfree(priv); + return ret; +} + +static int __devexit twl6040_codec_remove(struct platform_device *pdev) +{ + struct twl6040_data *priv = twl6040_codec->private_data; + int audpwron = priv->audpwron; + int naudint = priv->naudint; + + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); + + if (naudint) + free_irq(naudint, twl6040_codec); + + snd_soc_unregister_dai(&twl6040_dai); + snd_soc_unregister_codec(twl6040_codec); + + kfree(twl6040_codec); + twl6040_codec = NULL; + + return 0; +} + +static struct platform_driver twl6040_codec_driver = { + .driver = { + .name = "twl6040_codec", + .owner = THIS_MODULE, + }, + .probe = twl6040_codec_probe, + .remove = __devexit_p(twl6040_codec_remove), +}; + +static int __init twl6040_codec_init(void) +{ + return platform_driver_register(&twl6040_codec_driver); +} +module_init(twl6040_codec_init); + +static void __exit twl6040_codec_exit(void) +{ + platform_driver_unregister(&twl6040_codec_driver); +} +module_exit(twl6040_codec_exit); + +MODULE_DESCRIPTION("ASoC TWL6040 codec driver"); +MODULE_AUTHOR("Misael Lopez Cruz"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h new file mode 100644 index 0000000..c472070 --- /dev/null +++ b/sound/soc/codecs/twl6040.h @@ -0,0 +1,141 @@ +/* + * ALSA SoC TWL6040 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * 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 __TWL6040_H__ +#define __TWL6040_H__ + +#define TWL6040_REG_ASICID 0x01 +#define TWL6040_REG_ASICREV 0x02 +#define TWL6040_REG_INTID 0x03 +#define TWL6040_REG_INTMR 0x04 +#define TWL6040_REG_NCPCTL 0x05 +#define TWL6040_REG_LDOCTL 0x06 +#define TWL6040_REG_HPPLLCTL 0x07 +#define TWL6040_REG_LPPLLCTL 0x08 +#define TWL6040_REG_LPPLLDIV 0x09 +#define TWL6040_REG_AMICBCTL 0x0A +#define TWL6040_REG_DMICBCTL 0x0B +#define TWL6040_REG_MICLCTL 0x0C +#define TWL6040_REG_MICRCTL 0x0D +#define TWL6040_REG_MICGAIN 0x0E +#define TWL6040_REG_LINEGAIN 0x0F +#define TWL6040_REG_HSLCTL 0x10 +#define TWL6040_REG_HSRCTL 0x11 +#define TWL6040_REG_HSGAIN 0x12 +#define TWL6040_REG_EARCTL 0x13 +#define TWL6040_REG_HFLCTL 0x14 +#define TWL6040_REG_HFLGAIN 0x15 +#define TWL6040_REG_HFRCTL 0x16 +#define TWL6040_REG_HFRGAIN 0x17 +#define TWL6040_REG_VIBCTLL 0x18 +#define TWL6040_REG_VIBDATL 0x19 +#define TWL6040_REG_VIBCTLR 0x1A +#define TWL6040_REG_VIBDATR 0x1B +#define TWL6040_REG_HKCTL1 0x1C +#define TWL6040_REG_HKCTL2 0x1D +#define TWL6040_REG_GPOCTL 0x1E +#define TWL6040_REG_ALB 0x1F +#define TWL6040_REG_DLB 0x20 +#define TWL6040_REG_TRIM1 0x28 +#define TWL6040_REG_TRIM2 0x29 +#define TWL6040_REG_TRIM3 0x2A +#define TWL6040_REG_HSOTRIM 0x2B +#define TWL6040_REG_HFOTRIM 0x2C +#define TWL6040_REG_ACCCTL 0x2D +#define TWL6040_REG_STATUS 0x2E + +#define TWL6040_CACHEREGNUM (TWL6040_REG_STATUS + 1) + +#define TWL6040_VIOREGNUM 18 +#define TWL6040_VDDREGNUM 21 + +/* INTID (0x03) fields */ + +#define TWL6040_THINT 0x01 +#define TWL6040_PLUGINT 0x02 +#define TWL6040_UNPLUGINT 0x04 +#define TWL6040_HOOKINT 0x08 +#define TWL6040_HFINT 0x10 +#define TWL6040_VIBINT 0x20 +#define TWL6040_READYINT 0x40 + +/* INTMR (0x04) fields */ + +#define TWL6040_READYMSK 0x40 +#define TWL6040_ALLINT_MSK 0x7B + +/* NCPCTL (0x05) fields */ + +#define TWL6040_NCPENA 0x01 +#define TWL6040_NCPOPEN 0x40 + +/* LDOCTL (0x06) fields */ + +#define TWL6040_LSLDOENA 0x01 +#define TWL6040_HSLDOENA 0x04 +#define TWL6040_REFENA 0x40 +#define TWL6040_OSCENA 0x80 + +/* HPPLLCTL (0x07) fields */ + +#define TWL6040_HPLLENA 0x01 +#define TWL6040_HPLLRST 0x02 +#define TWL6040_HPLLBP 0x04 +#define TWL6040_HPLLSQRENA 0x08 +#define TWL6040_HPLLSQRBP 0x10 +#define TWL6040_MCLK_12000KHZ (0 << 5) +#define TWL6040_MCLK_19200KHZ (1 << 5) +#define TWL6040_MCLK_26000KHZ (2 << 5) +#define TWL6040_MCLK_38400KHZ (3 << 5) +#define TWL6040_MCLK_MSK 0x60 + +/* LPPLLCTL (0x08) fields */ + +#define TWL6040_LPLLENA 0x01 +#define TWL6040_LPLLRST 0x02 +#define TWL6040_LPLLSEL 0x04 +#define TWL6040_LPLLFIN 0x08 +#define TWL6040_HPLLSEL 0x10 + +/* HSLCTL (0x10) fields */ + +#define TWL6040_HSDACMODEL 0x02 +#define TWL6040_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6040_HSDACMODER 0x02 +#define TWL6040_HSDRVMODER 0x08 + +/* ACCCTL (0x2D) fields */ + +#define TWL6040_RESETSPLIT 0x04 + +#define TWL6040_SYSCLK_SEL_LPPLL 1 +#define TWL6040_SYSCLK_SEL_HPPLL 2 + +#define TWL6040_HPPLL_ID 1 +#define TWL6040_LPPLL_ID 2 + +extern struct snd_soc_dai twl6040_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl6040; + +#endif /* End of __TWL6040_H__ */ -- 1.6.1.3

3 5

[alsa-devel] [PATCH 0/2] ASoC: tlv320dac33: DSP mode and clocking updates
by Peter Ujfalusi 19 Mar '10

19 Mar '10

Hello, tlv320dac33 can not support DSP_B mode since the data delay can not be set to be actually 0. Only DSP_A mode can be supported, so fix the DSP_A mode, and remove the DSP_B. The original internal clock distribution config has been found to be not optimal, so fix that as well. Peter --- Peter Ujfalusi (2): ASoC: tlv320dac33: Fix DSP modes ASoC: tlv320dac33: Internal clocking changes sound/soc/codecs/tlv320dac33.c | 10 +++------- 1 files changed, 3 insertions(+), 7 deletions(-)

3 4

[alsa-devel] [PATCH] ASoC: McPDM: Use tabs for indentation
by Jorge Eduardo Candelaria 19 Mar '10

19 Mar '10

Indentation in initial support for McPDM driver was converted to spaces. Use tabs to comply with open source coding-style. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> --- sound/soc/omap/mcpdm.c | 548 ++++++++++++++++++++++++------------------------ 1 files changed, 274 insertions(+), 274 deletions(-) diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c index ad8df6c..479d3b6 100644 --- a/sound/soc/omap/mcpdm.c +++ b/sound/soc/omap/mcpdm.c @@ -1,5 +1,5 @@ /* - * mcpdm.c -- McPDM interface driver + * mcpdm.c -- McPDM interface driver * * Author: Jorge Eduardo Candelaria <x0107209(a)ti.com> * Copyright (C) 2009 - Texas Instruments, Inc. @@ -38,46 +38,46 @@ static struct omap_mcpdm *mcpdm; static inline void omap_mcpdm_write(u16 reg, u32 val) { - __raw_writel(val, mcpdm->io_base + reg); + __raw_writel(val, mcpdm->io_base + reg); } static inline int omap_mcpdm_read(u16 reg) { - return __raw_readl(mcpdm->io_base + reg); + return __raw_readl(mcpdm->io_base + reg); } static void omap_mcpdm_reg_dump(void) { - dev_dbg(mcpdm->dev, "***********************\n"); - dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQSTATUS_RAW)); - dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQSTATUS)); - dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQENABLE_SET)); - dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQENABLE_CLR)); - dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQWAKE_EN)); - dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", - omap_mcpdm_read(MCPDM_DMAENABLE_SET)); - dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", - omap_mcpdm_read(MCPDM_DMAENABLE_CLR)); - dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", - omap_mcpdm_read(MCPDM_DMAWAKEEN)); - dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", - omap_mcpdm_read(MCPDM_CTRL)); - dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", - omap_mcpdm_read(MCPDM_DN_DATA)); - dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", - omap_mcpdm_read(MCPDM_UP_DATA)); - dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", - omap_mcpdm_read(MCPDM_FIFO_CTRL_DN)); - dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", - omap_mcpdm_read(MCPDM_FIFO_CTRL_UP)); - dev_dbg(mcpdm->dev, "DN_OFFSET: 0x%04x\n", - omap_mcpdm_read(MCPDM_DN_OFFSET)); - dev_dbg(mcpdm->dev, "***********************\n"); + dev_dbg(mcpdm->dev, "***********************\n"); + dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS_RAW)); + dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS)); + dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_SET)); + dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_CLR)); + dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQWAKE_EN)); + dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_SET)); + dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_CLR)); + dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAWAKEEN)); + dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", + omap_mcpdm_read(MCPDM_CTRL)); + dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_DATA)); + dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_UP_DATA)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_DN)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_UP)); + dev_dbg(mcpdm->dev, "DN_OFFSET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_OFFSET)); + dev_dbg(mcpdm->dev, "***********************\n"); } /* @@ -86,26 +86,26 @@ static void omap_mcpdm_reg_dump(void) */ static void omap_mcpdm_reset_capture(int reset) { - int ctrl = omap_mcpdm_read(MCPDM_CTRL); + int ctrl = omap_mcpdm_read(MCPDM_CTRL); - if (reset) - ctrl |= SW_UP_RST; - else - ctrl &= ~SW_UP_RST; + if (reset) + ctrl |= SW_UP_RST; + else + ctrl &= ~SW_UP_RST; - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); } static void omap_mcpdm_reset_playback(int reset) { - int ctrl = omap_mcpdm_read(MCPDM_CTRL); + int ctrl = omap_mcpdm_read(MCPDM_CTRL); - if (reset) - ctrl |= SW_DN_RST; - else - ctrl &= ~SW_DN_RST; + if (reset) + ctrl |= SW_DN_RST; + else + ctrl &= ~SW_DN_RST; - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); } /* @@ -114,14 +114,14 @@ static void omap_mcpdm_reset_playback(int reset) */ void omap_mcpdm_start(int stream) { - int ctrl = omap_mcpdm_read(MCPDM_CTRL); + int ctrl = omap_mcpdm_read(MCPDM_CTRL); - if (stream) - ctrl |= mcpdm->up_channels; - else - ctrl |= mcpdm->dn_channels; + if (stream) + ctrl |= mcpdm->up_channels; + else + ctrl |= mcpdm->dn_channels; - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); } /* @@ -130,14 +130,14 @@ void omap_mcpdm_start(int stream) */ void omap_mcpdm_stop(int stream) { - int ctrl = omap_mcpdm_read(MCPDM_CTRL); + int ctrl = omap_mcpdm_read(MCPDM_CTRL); - if (stream) - ctrl &= ~mcpdm->up_channels; - else - ctrl &= ~mcpdm->dn_channels; + if (stream) + ctrl &= ~mcpdm->up_channels; + else + ctrl &= ~mcpdm->dn_channels; - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); } /* @@ -146,38 +146,38 @@ void omap_mcpdm_stop(int stream) */ int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink) { - int irq_mask = 0; - int ctrl; + int irq_mask = 0; + int ctrl; - if (!uplink) - return -EINVAL; + if (!uplink) + return -EINVAL; - mcpdm->uplink = uplink; + mcpdm->uplink = uplink; - /* Enable irq request generation */ - irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + /* Enable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); - /* Configure uplink threshold */ - if (uplink->threshold > UP_THRES_MAX) - uplink->threshold = UP_THRES_MAX; + /* Configure uplink threshold */ + if (uplink->threshold > UP_THRES_MAX) + uplink->threshold = UP_THRES_MAX; - omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); + omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); - /* Configure DMA controller */ - omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE); + /* Configure DMA controller */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE); - /* Set pdm out format */ - ctrl = omap_mcpdm_read(MCPDM_CTRL); - ctrl &= ~PDMOUTFORMAT; - ctrl |= uplink->format & PDMOUTFORMAT; + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= uplink->format & PDMOUTFORMAT; - /* Uplink channels */ - mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK); + /* Uplink channels */ + mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK); - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); - return 0; + return 0; } /* @@ -186,38 +186,38 @@ int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink) */ int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink) { - int irq_mask = 0; - int ctrl; + int irq_mask = 0; + int ctrl; - if (!downlink) - return -EINVAL; + if (!downlink) + return -EINVAL; - mcpdm->downlink = downlink; + mcpdm->downlink = downlink; - /* Enable irq request generation */ - irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + /* Enable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); - /* Configure uplink threshold */ - if (downlink->threshold > DN_THRES_MAX) - downlink->threshold = DN_THRES_MAX; + /* Configure uplink threshold */ + if (downlink->threshold > DN_THRES_MAX) + downlink->threshold = DN_THRES_MAX; - omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); + omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); - /* Enable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE); + /* Enable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE); - /* Set pdm out format */ - ctrl = omap_mcpdm_read(MCPDM_CTRL); - ctrl &= ~PDMOUTFORMAT; - ctrl |= downlink->format & PDMOUTFORMAT; + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= downlink->format & PDMOUTFORMAT; - /* Downlink channels */ - mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK); + /* Downlink channels */ + mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK); - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); - return 0; + return 0; } /* @@ -226,24 +226,24 @@ int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink) */ int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink) { - int irq_mask = 0; + int irq_mask = 0; - if (!uplink) - return -EINVAL; + if (!uplink) + return -EINVAL; - /* Disable irq request generation */ - irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + /* Disable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); - /* Disable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE); + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE); - /* Clear Downlink channels */ - mcpdm->up_channels = 0; + /* Clear Downlink channels */ + mcpdm->up_channels = 0; - mcpdm->uplink = NULL; + mcpdm->uplink = NULL; - return 0; + return 0; } /* @@ -252,124 +252,124 @@ int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink) */ int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink) { - int irq_mask = 0; + int irq_mask = 0; - if (!downlink) - return -EINVAL; + if (!downlink) + return -EINVAL; - /* Disable irq request generation */ - irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + /* Disable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); - /* Disable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE); + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE); - /* clear Downlink channels */ - mcpdm->dn_channels = 0; + /* clear Downlink channels */ + mcpdm->dn_channels = 0; - mcpdm->downlink = NULL; + mcpdm->downlink = NULL; - return 0; + return 0; } static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) { - struct omap_mcpdm *mcpdm_irq = dev_id; - int irq_status; - - irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS); - - /* Acknowledge irq event */ - omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); - - if (irq & MCPDM_DN_IRQ_FULL) { - dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); - omap_mcpdm_reset_playback(1); - omap_mcpdm_playback_open(mcpdm_irq->downlink); - omap_mcpdm_reset_playback(0); - } - - if (irq & MCPDM_DN_IRQ_EMPTY) { - dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); - omap_mcpdm_reset_playback(1); - omap_mcpdm_playback_open(mcpdm_irq->downlink); - omap_mcpdm_reset_playback(0); - } - - if (irq & MCPDM_DN_IRQ) { - dev_dbg(mcpdm_irq->dev, "DN write request\n"); - } - - if (irq & MCPDM_UP_IRQ_FULL) { - dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); - omap_mcpdm_reset_capture(1); - omap_mcpdm_capture_open(mcpdm_irq->uplink); - omap_mcpdm_reset_capture(0); - } - - if (irq & MCPDM_UP_IRQ_EMPTY) { - dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); - omap_mcpdm_reset_capture(1); - omap_mcpdm_capture_open(mcpdm_irq->uplink); - omap_mcpdm_reset_capture(0); - } - - if (irq & MCPDM_UP_IRQ) { - dev_dbg(mcpdm_irq->dev, "UP write request\n"); - } - - return IRQ_HANDLED; + struct omap_mcpdm *mcpdm_irq = dev_id; + int irq_status; + + irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS); + + /* Acknowledge irq event */ + omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); + + if (irq & MCPDM_DN_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset_playback(1); + omap_mcpdm_playback_open(mcpdm_irq->downlink); + omap_mcpdm_reset_playback(0); + } + + if (irq & MCPDM_DN_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset_playback(1); + omap_mcpdm_playback_open(mcpdm_irq->downlink); + omap_mcpdm_reset_playback(0); + } + + if (irq & MCPDM_DN_IRQ) { + dev_dbg(mcpdm_irq->dev, "DN write request\n"); + } + + if (irq & MCPDM_UP_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset_capture(1); + omap_mcpdm_capture_open(mcpdm_irq->uplink); + omap_mcpdm_reset_capture(0); + } + + if (irq & MCPDM_UP_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset_capture(1); + omap_mcpdm_capture_open(mcpdm_irq->uplink); + omap_mcpdm_reset_capture(0); + } + + if (irq & MCPDM_UP_IRQ) { + dev_dbg(mcpdm_irq->dev, "UP write request\n"); + } + + return IRQ_HANDLED; } int omap_mcpdm_request(void) { - int ret; + int ret; - clk_enable(mcpdm->clk); + clk_enable(mcpdm->clk); - spin_lock(&mcpdm->lock); + spin_lock(&mcpdm->lock); - if (!mcpdm->free) { - dev_err(mcpdm->dev, "McPDM interface is in use\n"); - spin_unlock(&mcpdm->lock); - ret = -EBUSY; - goto err; - } - mcpdm->free = 0; + if (!mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is in use\n"); + spin_unlock(&mcpdm->lock); + ret = -EBUSY; + goto err; + } + mcpdm->free = 0; - spin_unlock(&mcpdm->lock); + spin_unlock(&mcpdm->lock); - /* Disable lines while request is ongoing */ - omap_mcpdm_write(MCPDM_CTRL, 0x00); + /* Disable lines while request is ongoing */ + omap_mcpdm_write(MCPDM_CTRL, 0x00); - ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, - 0, "McPDM", (void *)mcpdm); - if (ret) { - dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n"); - goto err; - } + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + 0, "McPDM", (void *)mcpdm); + if (ret) { + dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n"); + goto err; + } - return 0; + return 0; err: - clk_disable(mcpdm->clk); - return ret; + clk_disable(mcpdm->clk); + return ret; } void omap_mcpdm_free(void) { - spin_lock(&mcpdm->lock); - if (mcpdm->free) { - dev_err(mcpdm->dev, "McPDM interface is already free\n"); - spin_unlock(&mcpdm->lock); - return; - } - mcpdm->free = 1; - spin_unlock(&mcpdm->lock); - - clk_disable(mcpdm->clk); - - free_irq(mcpdm->irq, (void *)mcpdm); + spin_lock(&mcpdm->lock); + if (mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is already free\n"); + spin_unlock(&mcpdm->lock); + return; + } + mcpdm->free = 1; + spin_unlock(&mcpdm->lock); + + clk_disable(mcpdm->clk); + + free_irq(mcpdm->irq, (void *)mcpdm); } /* Enable/disable DC offset cancelation for the analog @@ -377,108 +377,108 @@ void omap_mcpdm_free(void) */ int omap_mcpdm_set_offset(int offset1, int offset2) { - int offset; + int offset; - if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) - return -EINVAL; + if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) + return -EINVAL; - offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2); + offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2); - /* offset cancellation for channel 1 */ - if (offset1) - offset |= DN_OFST_RX1_EN; - else - offset &= ~DN_OFST_RX1_EN; + /* offset cancellation for channel 1 */ + if (offset1) + offset |= DN_OFST_RX1_EN; + else + offset &= ~DN_OFST_RX1_EN; - /* offset cancellation for channel 2 */ - if (offset2) - offset |= DN_OFST_RX2_EN; - else - offset &= ~DN_OFST_RX2_EN; + /* offset cancellation for channel 2 */ + if (offset2) + offset |= DN_OFST_RX2_EN; + else + offset &= ~DN_OFST_RX2_EN; - omap_mcpdm_write(MCPDM_DN_OFFSET, offset); + omap_mcpdm_write(MCPDM_DN_OFFSET, offset); - return 0; + return 0; } static int __devinit omap_mcpdm_probe(struct platform_device *pdev) { - struct resource *res; - int ret = 0; - - mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); - if (!mcpdm) { - ret = -ENOMEM; - goto exit; - } - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) { - dev_err(&pdev->dev, "no resource\n"); - goto err_resource; - } - - spin_lock_init(&mcpdm->lock); - mcpdm->free = 1; - mcpdm->io_base = ioremap(res->start, resource_size(res)); - if (!mcpdm->io_base) { - ret = -ENOMEM; - goto err_resource; - } - - mcpdm->irq = platform_get_irq(pdev, 0); - - mcpdm->clk = clk_get(&pdev->dev, "pdm_ck"); - if (IS_ERR(mcpdm->clk)) { - ret = PTR_ERR(mcpdm->clk); - dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret); - goto err_clk; - } - - mcpdm->dev = &pdev->dev; - platform_set_drvdata(pdev, mcpdm); - - return 0; + struct resource *res; + int ret = 0; + + mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + if (!mcpdm) { + ret = -ENOMEM; + goto exit; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + dev_err(&pdev->dev, "no resource\n"); + goto err_resource; + } + + spin_lock_init(&mcpdm->lock); + mcpdm->free = 1; + mcpdm->io_base = ioremap(res->start, resource_size(res)); + if (!mcpdm->io_base) { + ret = -ENOMEM; + goto err_resource; + } + + mcpdm->irq = platform_get_irq(pdev, 0); + + mcpdm->clk = clk_get(&pdev->dev, "pdm_ck"); + if (IS_ERR(mcpdm->clk)) { + ret = PTR_ERR(mcpdm->clk); + dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret); + goto err_clk; + } + + mcpdm->dev = &pdev->dev; + platform_set_drvdata(pdev, mcpdm); + + return 0; err_clk: - iounmap(mcpdm->io_base); + iounmap(mcpdm->io_base); err_resource: - kfree(mcpdm); + kfree(mcpdm); exit: - return ret; + return ret; } static int __devexit omap_mcpdm_remove(struct platform_device *pdev) { - struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); + struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); - platform_set_drvdata(pdev, NULL); + platform_set_drvdata(pdev, NULL); - clk_put(mcpdm_ptr->clk); + clk_put(mcpdm_ptr->clk); - iounmap(mcpdm_ptr->io_base); + iounmap(mcpdm_ptr->io_base); - mcpdm_ptr->clk = NULL; - mcpdm_ptr->free = 0; - mcpdm_ptr->dev = NULL; + mcpdm_ptr->clk = NULL; + mcpdm_ptr->free = 0; + mcpdm_ptr->dev = NULL; - kfree(mcpdm_ptr); + kfree(mcpdm_ptr); - return 0; + return 0; } static struct platform_driver omap_mcpdm_driver = { - .probe = omap_mcpdm_probe, - .remove = __devexit_p(omap_mcpdm_remove), - .driver = { - .name = "omap-mcpdm", - }, + .probe = omap_mcpdm_probe, + .remove = __devexit_p(omap_mcpdm_remove), + .driver = { + .name = "omap-mcpdm", + }, }; static struct platform_device *omap_mcpdm_device; static int __init omap_mcpdm_init(void) { - return platform_driver_register(&omap_mcpdm_driver); + return platform_driver_register(&omap_mcpdm_driver); } arch_initcall(omap_mcpdm_init); -- 1.6.0.4

3 2

[alsa-devel] [PATCH v2] ASoC: omap-mcbsp: Add support for Left Justified format
by Peter Ujfalusi 19 Mar '10

19 Mar '10

Basic support for Left Justified coding for OMAP McBSP. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/omap/omap-mcbsp.c | 12 +++++++++++- 1 files changed, 11 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 2952fb0..0199e95 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -331,7 +331,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; wpf = channels = params_channels(params); - if (channels == 2 && format == SND_SOC_DAIFMT_I2S) { + if (channels == 2 && (format == SND_SOC_DAIFMT_I2S || + format == SND_SOC_DAIFMT_LEFT_J)) { /* Use dual-phase frames */ regs->rcr2 |= RPHASE; regs->xcr2 |= XPHASE; @@ -376,6 +377,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* Set FS period and length in terms of bit clock periods */ switch (format) { case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: regs->srgr2 |= FPER(framesize - 1); regs->srgr1 |= FWID((framesize >> 1) - 1); break; @@ -427,6 +429,14 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; + case SND_SOC_DAIFMT_LEFT_J: + /* 0-bit data delay */ + regs->rcr2 |= RDATDLY(0); + regs->xcr2 |= XDATDLY(0); + regs->spcr1 |= RJUST(2); + /* Invert FS polarity configuration */ + temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + break; case SND_SOC_DAIFMT_DSP_A: /* 1-bit data delay */ regs->rcr2 |= RDATDLY(1); -- 1.7.0.2

4 3

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Add support for Left Justified format
by Peter Ujfalusi 19 Mar '10

19 Mar '10

Basic support for Left Justified coding for OMAP McBSP. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/omap/omap-mcbsp.c | 10 +++++++++- 1 files changed, 9 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 2952fb0..d315cc4 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -331,7 +331,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; wpf = channels = params_channels(params); - if (channels == 2 && format == SND_SOC_DAIFMT_I2S) { + if (channels == 2 && (format == SND_SOC_DAIFMT_I2S || + format == SND_SOC_DAIFMT_LEFT_J)) { /* Use dual-phase frames */ regs->rcr2 |= RPHASE; regs->xcr2 |= XPHASE; @@ -376,6 +377,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* Set FS period and length in terms of bit clock periods */ switch (format) { case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: regs->srgr2 |= FPER(framesize - 1); regs->srgr1 |= FWID((framesize >> 1) - 1); break; @@ -427,6 +429,12 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; + case SND_SOC_DAIFMT_LEFT_J: + /* 0-bit data delay */ + regs->rcr2 |= RDATDLY(0); + regs->xcr2 |= XDATDLY(0); + regs->spcr1 |= RJUST(2); + break; case SND_SOC_DAIFMT_DSP_A: /* 1-bit data delay */ regs->rcr2 |= RDATDLY(1); -- 1.7.0.2

2 4

[alsa-devel] HDMI output with headphone jack sense possible on Conexant CX20561?
by Andrew Udvare 18 Mar '10

18 Mar '10

My audio codec is Conexant CX20561 (Hermosa) (haven't been able to find anything about 'Hermosa' on the Wiki). I'd like to be able to do a few things which I have not been able to do with tutorials I found on the web: - Whenever I plug in headphones, HDMI output is muted and headphones are enabled - Right now, the way I do this is by manually editing /etc/asound.conf to have the following (commented out makes headphones work, not commented makes HDMI work without any headphone/laptop speaker output) /etc/asound.conf: defaults.pcm.card 0 defaults.pcm.device 3 where 3 is my HDMI (IEC958 2; no idea what IEC958 1 is for) output. So I have a switcher script that does the commenting and uncommenting. Would be nice if this could be automatic (even if alsa had to run such a script on detecting headphones). - Route audio output (PCM) to input (for mixer recording) (Intel HDA cannot do this apparently by default (thanks DRM), needs some hacking around?) Information from alsa-info.sh: http://www.alsa- project.org/db/?f=107f3f6012e68f04d3562d5e11aeb052a1454884 Are my model options correct? And if it's important to know, I build ALSA in-kernel. Is this not recommended? I know at the very least it makes it such that I have to reboot to make any new options to take effect. Thanks all -- Andrew Udvare audvare(a)gmail.com 978-870-5908 www.andyudvare.com

1 0

[alsa-devel] [PATCHv6 2/2] ASoC: TWL6040: Add twl6040 codec driver
by Olaya, Margarita 18 Mar '10

18 Mar '10

From: Misael Lopez Cruz <x0052729(a)ti.com> Initial version of TWL6040 codec driver. The TWL6040 codec uses a proprietary PDM-based digital audio interface. Audio paths supported are: - Input: Main Mic, Sub Mic, Headset Mic, Auxiliary-FM Left/Right - Output: Headset Left/Right, Handsfree Left/Right TWL6040 codec supports power-up/down manual and automatic sequence. Manual sequence is done through a specific register writes sequence. Automatic sequence is done when the codec is powered-up through the external AUDPWRON line. The completion of the sequence is signaled through the audio interrupt. TWL6040 codec sysclk can be provided by: low-power or high performance PLL: - The low-power PLL takes a low-frequency input at 32,768 Hz and generates an approximate of 17.64 or 19.2 MHz (for 44.1 KHz and 48 KHz respectively) - The high-performance PLL generates an exact 19.2 MHz clock signal from high-frequency input at 12/19.2/26/38.4 MHz. Low-power playback mode is a special scenario where only headset path (headset DAC and driver) is active. For the particular case of headset path, PLL being used defines the headset power mode: low-power, high-performance. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6040.c | 1227 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 141 +++++ 4 files changed, 1374 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl6040.c create mode 100644 sound/soc/codecs/twl6040.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 16c47ed..398cbb0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,6 +35,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_TWL6040 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM2000 if I2C @@ -168,6 +169,9 @@ config SND_SOC_TWL4030 select TWL4030_CODEC tristate +config SND_SOC_TWL6040 + tristate + config SND_SOC_UDA134X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6981777..98bd10c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,7 @@ 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-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o @@ -85,6 +86,7 @@ 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-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c new file mode 100644 index 0000000..b90bc60 --- /dev/null +++ b/sound/soc/codecs/twl6040.c @@ -0,0 +1,1227 @@ +/* + * ALSA SoC TWL6040 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * 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/gpio.h> +#include <linux/platform_device.h> +#include <linux/i2c/twl.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 "twl6040.h" + +#define TWL6040_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct twl6040_data { + struct snd_soc_codec codec; + int audpwron; + int naudint; + int codec_powered; + int pll; + int non_lp; + unsigned int sysclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct completion ready; +}; + +/* + * twl6040 register cache & default register settings + */ +static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { + 0x00, /* not used 0x00 */ + 0x4B, /* TWL6040_ASICID (ro) 0x01 */ + 0x00, /* TWL6040_ASICREV (ro) 0x02 */ + 0x00, /* TWL6040_INTID 0x03 */ + 0x00, /* TWL6040_INTMR 0x04 */ + 0x00, /* TWL6040_NCPCTRL 0x05 */ + 0x00, /* TWL6040_LDOCTL 0x06 */ + 0x60, /* TWL6040_HPPLLCTL 0x07 */ + 0x00, /* TWL6040_LPPLLCTL 0x08 */ + 0x4A, /* TWL6040_LPPLLDIV 0x09 */ + 0x00, /* TWL6040_AMICBCTL 0x0A */ + 0x00, /* TWL6040_DMICBCTL 0x0B */ + 0x18, /* TWL6040_MICLCTL 0x0C - No input selected on Left Mic */ + 0x18, /* TWL6040_MICRCTL 0x0D - No input selected on Right Mic */ + 0x00, /* TWL6040_MICGAIN 0x0E */ + 0x1B, /* TWL6040_LINEGAIN 0x0F */ + 0x00, /* TWL6040_HSLCTL 0x10 */ + 0x00, /* TWL6040_HSRCTL 0x11 */ + 0x00, /* TWL6040_HSGAIN 0x12 */ + 0x00, /* TWL6040_EARCTL 0x13 */ + 0x00, /* TWL6040_HFLCTL 0x14 */ + 0x00, /* TWL6040_HFLGAIN 0x15 */ + 0x00, /* TWL6040_HFRCTL 0x16 */ + 0x00, /* TWL6040_HFRGAIN 0x17 */ + 0x00, /* TWL6040_VIBCTLL 0x18 */ + 0x00, /* TWL6040_VIBDATL 0x19 */ + 0x00, /* TWL6040_VIBCTLR 0x1A */ + 0x00, /* TWL6040_VIBDATR 0x1B */ + 0x00, /* TWL6040_HKCTL1 0x1C */ + 0x00, /* TWL6040_HKCTL2 0x1D */ + 0x00, /* TWL6040_GPOCTL 0x1E */ + 0x00, /* TWL6040_ALB 0x1F */ + 0x00, /* TWL6040_DLB 0x20 */ + 0x00, /* not used 0x21 */ + 0x00, /* not used 0x22 */ + 0x00, /* not used 0x23 */ + 0x00, /* not used 0x24 */ + 0x00, /* not used 0x25 */ + 0x00, /* not used 0x26 */ + 0x00, /* not used 0x27 */ + 0x00, /* TWL6040_TRIM1 0x28 */ + 0x00, /* TWL6040_TRIM2 0x29 */ + 0x00, /* TWL6040_TRIM3 0x2A */ + 0x00, /* TWL6040_HSOTRIM 0x2B */ + 0x00, /* TWL6040_HFOTRIM 0x2C */ + 0x09, /* TWL6040_ACCCTL 0x2D */ + 0x00, /* TWL6040_STATUS (ro) 0x2E */ +}; + +/* + * twl6040 vio/gnd registers: + * registers under vio/gnd supply can be accessed + * before the power-up sequence, after NRESPWRON goes high + */ +static const int twl6040_vio_reg[TWL6040_VIOREGNUM] = { + TWL6040_REG_ASICID, + TWL6040_REG_ASICREV, + TWL6040_REG_INTID, + TWL6040_REG_INTMR, + TWL6040_REG_NCPCTL, + TWL6040_REG_LDOCTL, + TWL6040_REG_AMICBCTL, + TWL6040_REG_DMICBCTL, + TWL6040_REG_HKCTL1, + TWL6040_REG_HKCTL2, + TWL6040_REG_GPOCTL, + TWL6040_REG_TRIM1, + TWL6040_REG_TRIM2, + TWL6040_REG_TRIM3, + TWL6040_REG_HSOTRIM, + TWL6040_REG_HFOTRIM, + TWL6040_REG_ACCCTL, + TWL6040_REG_STATUS, +}; + +/* + * twl6040 vdd/vss registers: + * registers under vdd/vss supplies can only be accessed + * after the power-up sequence + */ +static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = { + TWL6040_REG_HPPLLCTL, + TWL6040_REG_LPPLLCTL, + TWL6040_REG_LPPLLDIV, + TWL6040_REG_MICLCTL, + TWL6040_REG_MICRCTL, + TWL6040_REG_MICGAIN, + TWL6040_REG_LINEGAIN, + TWL6040_REG_HSLCTL, + TWL6040_REG_HSRCTL, + TWL6040_REG_HSGAIN, + TWL6040_REG_EARCTL, + TWL6040_REG_HFLCTL, + TWL6040_REG_HFLGAIN, + TWL6040_REG_HFRCTL, + TWL6040_REG_HFRGAIN, + TWL6040_REG_VIBCTLL, + TWL6040_REG_VIBDATL, + TWL6040_REG_VIBCTLR, + TWL6040_REG_VIBDATR, + TWL6040_REG_ALB, + TWL6040_REG_DLB, +}; + +/* + * read twl6040 register cache + */ +static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + return cache[reg]; +} + +/* + * write twl6040 register cache + */ +static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6040_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * read from twl6040 hardware register + */ +static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &value, reg); + twl6040_write_reg_cache(codec, reg, value); + + return value; +} + +/* + * write to the twl6040 register space + */ +static int twl6040_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + twl6040_write_reg_cache(codec, reg, value); + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); +} + +static void twl6040_init_vio_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + /* allow registers to be accessed by i2c */ + twl6040_write(codec, TWL6040_REG_ACCCTL, cache[TWL6040_REG_ACCCTL]); + + for (i = 0; i < TWL6040_VIOREGNUM; i++) { + reg = twl6040_vio_reg[i]; + /* skip read-only registers (ASICID, ASICREV, STATUS) */ + switch (reg) { + case TWL6040_REG_ASICID: + case TWL6040_REG_ASICREV: + case TWL6040_REG_STATUS: + continue; + default: + break; + } + twl6040_write(codec, reg, cache[reg]); + } +} + +static void twl6040_init_vdd_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + for (i = 0; i < TWL6040_VDDREGNUM; i++) { + reg = twl6040_vdd_reg[i]; + twl6040_write(codec, reg, cache[reg]); + } +} + +/* twl6040 codec manual power-up sequence */ +static void twl6040_power_up(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL); + ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL); + + /* enable reference system */ + ldoctl |= TWL6040_REFENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + msleep(10); + /* enable internal oscillator */ + ldoctl |= TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(10); + /* enable high-side ldo */ + ldoctl |= TWL6040_HSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* enable negative charge pump */ + ncpctl |= TWL6040_NCPENA | TWL6040_NCPOPEN; + twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl); + udelay(488); + /* enable low-side ldo */ + ldoctl |= TWL6040_LSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* enable low-power pll */ + lppllctl |= TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + /* reset state machine */ + accctl |= TWL6040_RESETSPLIT; + twl6040_write(codec, TWL6040_REG_ACCCTL, accctl); + mdelay(5); + accctl &= ~TWL6040_RESETSPLIT; + twl6040_write(codec, TWL6040_REG_ACCCTL, accctl); + /* disable internal oscillator */ + ldoctl &= ~TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); +} + +/* twl6040 codec manual power-down sequence */ +static void twl6040_power_down(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL); + ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL); + + /* enable internal oscillator */ + ldoctl |= TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(10); + /* disable low-power pll */ + lppllctl &= ~TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + /* disable low-side ldo */ + ldoctl &= ~TWL6040_LSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* disable negative charge pump */ + ncpctl &= ~(TWL6040_NCPENA | TWL6040_NCPOPEN); + twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl); + udelay(488); + /* disable high-side ldo */ + ldoctl &= ~TWL6040_HSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* disable internal oscillator */ + ldoctl &= ~TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + /* disable reference system */ + ldoctl &= ~TWL6040_REFENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + msleep(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 = TWL6040_HSDRVMODEL | TWL6040_HSDACMODEL; + + hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + + if (high_perf) { + hslctl &= ~mask; + hsrctl &= ~mask; + } else { + hslctl |= mask; + hsrctl |= mask; + } + + twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl); + twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl); + + return 0; +} + +static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct twl6040_data *priv = codec->private_data; + + if (SND_SOC_DAPM_EVENT_ON(event)) + priv->non_lp++; + else + priv->non_lp--; + + return 0; +} + +/* audio interrupt handler */ +static irqreturn_t twl6040_naudint_handler(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct twl6040_data *priv = codec->private_data; + u8 intid; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); + + switch (intid) { + case TWL6040_THINT: + dev_alert(codec->dev, "die temp over-limit detection\n"); + break; + case TWL6040_PLUGINT: + case TWL6040_UNPLUGINT: + case TWL6040_HOOKINT: + break; + case TWL6040_HFINT: + dev_alert(codec->dev, "hf drivers over current detection\n"); + break; + case TWL6040_VIBINT: + dev_alert(codec->dev, "vib drivers over current detection\n"); + break; + case TWL6040_READYINT: + complete(&priv->ready); + break; + default: + dev_err(codec->dev, "unknown audio interrupt %d\n", intid); + break; + } + + return IRQ_HANDLED; +} + +/* + * MICATT volume control: + * from -6 to 0 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); + +/* + * MICGAIN volume control: + * from 6 to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); + +/* + * HSGAIN volume control: + * from -30 to 0 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0); + +/* + * HFGAIN volume control: + * from -52 to 6 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0); + +/* Left analog microphone selection */ +static const char *twl6040_amicl_texts[] = + {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; + +/* Right analog microphone selection */ +static const char *twl6040_amicr_texts[] = + {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; + +static const struct soc_enum twl6040_enum[] = { + SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 3, twl6040_amicl_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 3, twl6040_amicr_texts), +}; + +static const struct snd_kcontrol_new amicl_control = + SOC_DAPM_ENUM("Route", twl6040_enum[0]); + +static const struct snd_kcontrol_new amicr_control = + SOC_DAPM_ENUM("Route", twl6040_enum[1]); + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new hsdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 5, 1, 0); + +static const struct snd_kcontrol_new hsdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 5, 1, 0); + +/* Handsfree DAC playback switches */ +static const struct snd_kcontrol_new hfdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hfdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0); + +/* Headset driver switches */ +static const struct snd_kcontrol_new hsl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hsr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0); + +/* Handsfree driver switches */ +static const struct snd_kcontrol_new hfl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0); + +static const struct snd_kcontrol_new hfr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0); + +static const struct snd_kcontrol_new twl6040_snd_controls[] = { + /* Capture gains */ + SOC_DOUBLE_TLV("Capture Preamplifier Volume", + TWL6040_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv), + SOC_DOUBLE_TLV("Capture Volume", + TWL6040_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + + /* Playback gains */ + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + +}; + +static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("AFML"), + SND_SOC_DAPM_INPUT("AFMR"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + + /* Analog input muxes for the capture amplifiers */ + SND_SOC_DAPM_MUX("Analog Left Capture Route", + SND_SOC_NOPM, 0, 0, &amicl_control), + SND_SOC_DAPM_MUX("Analog Right Capture Route", + SND_SOC_NOPM, 0, 0, &amicr_control), + + /* Analog capture PGAs */ + SND_SOC_DAPM_PGA("MicAmpL", + TWL6040_REG_MICLCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmpR", + TWL6040_REG_MICRCTL, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", + TWL6040_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", + TWL6040_REG_MICRCTL, 2, 0), + + /* Microphone bias */ + SND_SOC_DAPM_MICBIAS("Headset Mic Bias", + TWL6040_REG_AMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Main Mic Bias", + TWL6040_REG_AMICBCTL, 4, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias", + TWL6040_REG_DMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias", + TWL6040_REG_DMICBCTL, 4, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", + TWL6040_REG_HSLCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", + TWL6040_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", + TWL6040_REG_HFLCTL, 0, 0, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("HFDAC Right", "Handsfree Playback", + TWL6040_REG_HFRCTL, 0, 0, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Analog playback switches */ + SND_SOC_DAPM_SWITCH("HSDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hsdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HSDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hsdacr_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hfdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), + + SND_SOC_DAPM_SWITCH("Headset Left Driver", + SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Headset Right Driver", + SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), + SND_SOC_DAPM_SWITCH_E("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH_E("Handsfree Right Driver", + SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Analog playback PGAs */ + SND_SOC_DAPM_PGA("HFDAC Left PGA", + TWL6040_REG_HFLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HFDAC Right PGA", + TWL6040_REG_HFRCTL, 1, 0, NULL, 0), + +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Capture path */ + {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Left Capture Route", "Main Mic", "MAINMIC"}, + {"Analog Left Capture Route", "Aux/FM Left", "AFML"}, + + {"Analog Right Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Right Capture Route", "Sub Mic", "SUBMIC"}, + {"Analog Right Capture Route", "Aux/FM Right", "AFMR"}, + + {"MicAmpL", NULL, "Analog Left Capture Route"}, + {"MicAmpR", NULL, "Analog Right Capture Route"}, + + {"ADC Left", NULL, "MicAmpL"}, + {"ADC Right", NULL, "MicAmpR"}, + + /* Headset playback path */ + {"HSDAC Left Playback", "Switch", "HSDAC Left"}, + {"HSDAC Right Playback", "Switch", "HSDAC Right"}, + + {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, + {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + + {"HSOL", NULL, "Headset Left Driver"}, + {"HSOR", NULL, "Headset Right Driver"}, + + /* Handsfree playback path */ + {"HFDAC Left Playback", "Switch", "HFDAC Left"}, + {"HFDAC Right Playback", "Switch", "HFDAC Right"}, + + {"HFDAC Left PGA", NULL, "HFDAC Left Playback"}, + {"HFDAC Right PGA", NULL, "HFDAC Right Playback"}, + + {"Handsfree Left Driver", "Switch", "HFDAC Left PGA"}, + {"Handsfree Right Driver", "Switch", "HFDAC Right PGA"}, + + {"HFL", NULL, "Handsfree Left Driver"}, + {"HFR", NULL, "Handsfree Right Driver"}, +}; + +static int twl6040_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl6040_dapm_widgets, + ARRAY_SIZE(twl6040_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int twl6040_power_up_completion(struct snd_soc_codec *codec, + int naudint) +{ + struct twl6040_data *priv = codec->private_data; + int time_left; + u8 intid; + + time_left = wait_for_completion_timeout(&priv->ready, + msecs_to_jiffies(48)); + + if (!time_left) { + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, + TWL6040_REG_INTID); + if (!(intid & TWL6040_READYINT)) { + dev_err(codec->dev, "timeout waiting for READYINT\n"); + return -ETIMEDOUT; + } + } + + priv->codec_powered = 1; + + return 0; +} + +static int twl6040_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct twl6040_data *priv = codec->private_data; + int audpwron = priv->audpwron; + int naudint = priv->naudint; + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 1); + + /* wait for power-up completion */ + ret = twl6040_power_up_completion(codec, naudint); + if (ret) + return ret; + + /* sync registers updated during power-up sequence */ + twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LPPLLCTL); + } else { + /* use manual power-up sequence */ + twl6040_power_up(codec); + priv->codec_powered = 1; + } + + /* initialize vdd/vss registers with reg_cache */ + twl6040_init_vdd_regs(codec); + break; + case SND_SOC_BIAS_OFF: + if (!priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 0); + + /* power-down sequence latency */ + udelay(500); + + /* sync registers updated during power-down sequence */ + twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL); + twl6040_write_reg_cache(codec, TWL6040_REG_LPPLLCTL, + 0x00); + } else { + /* use manual power-down sequence */ + twl6040_power_down(codec); + } + + priv->codec_powered = 0; + break; + } + + codec->bias_level = level; + + return 0; +} + +/* set of rates for each pll: low-power and high-performance */ + +static unsigned int lp_rates[] = { + 88200, + 96000, +}; + +static struct snd_pcm_hw_constraint_list lp_constraints = { + .count = ARRAY_SIZE(lp_rates), + .list = lp_rates, +}; + +static unsigned int hp_rates[] = { + 96000, +}; + +static struct snd_pcm_hw_constraint_list hp_constraints = { + .count = ARRAY_SIZE(hp_rates), + .list = hp_rates, +}; + +static int twl6040_startup(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 twl6040_data *priv = codec->private_data; + + if (!priv->sysclk) { + dev_err(codec->dev, + "no mclk configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + /* + * capture is not supported at 17.64 MHz, + * it's reserved for headset low-power playback scenario + */ + if ((priv->sysclk == 17640000) && substream->stream) { + dev_err(codec->dev, + "capture mode is not supported at %dHz\n", + priv->sysclk); + return -EINVAL; + } + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + priv->sysclk_constraints); + + return 0; +} + +static int twl6040_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; + struct twl6040_data *priv = codec->private_data; + u8 lppllctl; + int rate; + + /* nothing to do for high-perf pll, it supports only 48 kHz */ + if (priv->pll == TWL6040_HPPLL_ID) + return 0; + + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + + rate = params_rate(params); + switch (rate) { + case 88200: + lppllctl |= TWL6040_LPLLFIN; + priv->sysclk = 17640000; + break; + case 96000: + lppllctl &= ~TWL6040_LPLLFIN; + priv->sysclk = 19200000; + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", rate); + return -EINVAL; + } + + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + return 0; +} + +static int twl6040_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 twl6040_data *priv = codec->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + /* + * low-power playback mode is restricted + * for headset path only + */ + if ((priv->sysclk == 17640000) && priv->non_lp) { + dev_err(codec->dev, + "some enabled paths aren't supported at %dHz\n", + priv->sysclk); + return -EPERM; + } + break; + default: + break; + } + + return 0; +} + +static int twl6040_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 twl6040_data *priv = codec->private_data; + u8 hppllctl, lppllctl; + + hppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_HPPLLCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + + switch (clk_id) { + case TWL6040_SYSCLK_SEL_LPPLL: + switch (freq) { + case 32768: + /* headset dac and driver must be in low-power mode */ + headset_power_mode(codec, 0); + + /* clk32k input requires low-power pll */ + lppllctl |= TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + mdelay(5); + lppllctl &= ~TWL6040_HPLLSEL; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + hppllctl &= ~TWL6040_HPLLENA; + twl6040_write(codec, TWL6040_REG_HPPLLCTL, hppllctl); + break; + default: + dev_err(codec->dev, "unknown mclk freq %d\n", freq); + return -EINVAL; + } + + /* lppll divider */ + switch (priv->sysclk) { + case 17640000: + lppllctl |= TWL6040_LPLLFIN; + break; + case 19200000: + lppllctl &= ~TWL6040_LPLLFIN; + break; + default: + /* sysclk not yet configured */ + lppllctl &= ~TWL6040_LPLLFIN; + priv->sysclk = 19200000; + break; + } + + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + priv->pll = TWL6040_LPPLL_ID; + priv->sysclk_constraints = &lp_constraints; + break; + case TWL6040_SYSCLK_SEL_HPPLL: + hppllctl &= ~TWL6040_MCLK_MSK; + + switch (freq) { + case 12000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6040_MCLK_12000KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLENA; + break; + case 19200000: + /* mclk input, pll disabled */ + hppllctl |= TWL6040_MCLK_19200KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLBP; + break; + case 26000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6040_MCLK_26000KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLENA; + break; + case 38400000: + /* clk slicer, pll disabled */ + hppllctl |= TWL6040_MCLK_38400KHZ | + TWL6040_HPLLSQRENA | + TWL6040_HPLLBP; + break; + default: + dev_err(codec->dev, "unknown mclk freq %d\n", freq); + return -EINVAL; + } + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + twl6040_write(codec, TWL6040_REG_HPPLLCTL, hppllctl); + udelay(500); + lppllctl |= TWL6040_HPLLSEL; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + lppllctl &= ~TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + /* high-performance pll can provide only 19.2 MHz */ + priv->pll = TWL6040_HPPLL_ID; + priv->sysclk = 19200000; + priv->sysclk_constraints = &hp_constraints; + break; + default: + dev_err(codec->dev, "unknown clk_id %d\n", clk_id); + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops twl6040_dai_ops = { + .startup = twl6040_startup, + .hw_params = twl6040_hw_params, + .trigger = twl6040_trigger, + .set_sysclk = twl6040_set_dai_sysclk, +}; + +struct snd_soc_dai twl6040_dai = { + .name = "twl6040", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}; +EXPORT_SYMBOL_GPL(twl6040_dai); + +#ifdef CONFIG_PM +static int twl6040_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; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl6040_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6040_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} +#else +#define twl6040_suspend NULL +#define twl6040_resume NULL +#endif + +static struct snd_soc_codec *twl6040_codec; + +static int twl6040_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + BUG_ON(!twl6040_codec); + + codec = twl6040_codec; + socdev->card->codec = codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; + } + + snd_soc_add_controls(codec, twl6040_snd_controls, + ARRAY_SIZE(twl6040_snd_controls)); + twl6040_add_widgets(codec); + + if (ret < 0) { + dev_err(&pdev->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + return ret; +} + +static int twl6040_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_twl6040 = { + .probe = twl6040_probe, + .remove = twl6040_remove, + .suspend = twl6040_suspend, + .resume = twl6040_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl6040); + +static int __devinit twl6040_codec_probe(struct platform_device *pdev) +{ + struct twl4030_codec_data *twl_codec = pdev->dev.platform_data; + struct snd_soc_codec *codec; + struct twl6040_data *priv; + int audpwron, naudint; + int ret = 0; + + priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + if (twl_codec) { + audpwron = twl_codec->audpwron_gpio; + naudint = twl_codec->naudint_irq; + } else { + audpwron = -EINVAL; + naudint = 0; + } + + priv->audpwron = audpwron; + priv->naudint = naudint; + + codec = &priv->codec; + codec->dev = &pdev->dev; + twl6040_dai.dev = &pdev->dev; + + codec->name = "twl6040"; + codec->owner = THIS_MODULE; + codec->read = twl6040_read_reg_cache; + codec->write = twl6040_write; + codec->set_bias_level = twl6040_set_bias_level; + codec->private_data = priv; + codec->dai = &twl6040_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(twl6040_reg); + codec->reg_cache = kmemdup(twl6040_reg, sizeof(twl6040_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + init_completion(&priv->ready); + + if (gpio_is_valid(audpwron)) { + ret = gpio_request(audpwron, "audpwron"); + if (ret) + goto gpio1_err; + + ret = gpio_direction_output(audpwron, 0); + if (ret) + goto gpio2_err; + + priv->codec_powered = 0; + } + + if (naudint) { + /* audio interrupt */ + ret = request_threaded_irq(naudint, NULL, + twl6040_naudint_handler, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "twl6040_codec", codec); + if (ret) + goto gpio2_err; + } else { + if (gpio_is_valid(audpwron)) { + /* enable only codec ready interrupt */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + ~TWL6040_READYMSK & TWL6040_ALLINT_MSK); + } else { + /* no interrupts at all */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + TWL6040_ALLINT_MSK); + } + } + + /* init vio registers */ + twl6040_init_vio_regs(codec); + + /* power on device */ + ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto irq_err; + + ret = snd_soc_register_codec(codec); + if (ret) + goto reg_err; + + twl6040_codec = codec; + + ret = snd_soc_register_dai(&twl6040_dai); + if (ret) + goto dai_err; + + return 0; + +dai_err: + snd_soc_unregister_codec(codec); + twl6040_codec = NULL; +reg_err: + twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); +irq_err: + if (naudint) + free_irq(naudint, codec); +gpio2_err: + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); +gpio1_err: + kfree(codec->reg_cache); +cache_err: + kfree(priv); + return ret; +} + +static int __devexit twl6040_codec_remove(struct platform_device *pdev) +{ + struct twl6040_data *priv = twl6040_codec->private_data; + int audpwron = priv->audpwron; + int naudint = priv->naudint; + + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); + + if (naudint) + free_irq(naudint, twl6040_codec); + + snd_soc_unregister_dai(&twl6040_dai); + snd_soc_unregister_codec(twl6040_codec); + + kfree(twl6040_codec); + twl6040_codec = NULL; + + return 0; +} + +static struct platform_driver twl6040_codec_driver = { + .driver = { + .name = "twl6040_codec", + .owner = THIS_MODULE, + }, + .probe = twl6040_codec_probe, + .remove = __devexit_p(twl6040_codec_remove), +}; + +static int __init twl6040_codec_init(void) +{ + return platform_driver_register(&twl6040_codec_driver); +} +module_init(twl6040_codec_init); + +static void __exit twl6040_codec_exit(void) +{ + platform_driver_unregister(&twl6040_codec_driver); +} +module_exit(twl6040_codec_exit); + +MODULE_DESCRIPTION("ASoC TWL6040 codec driver"); +MODULE_AUTHOR("Misael Lopez Cruz"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h new file mode 100644 index 0000000..c472070 --- /dev/null +++ b/sound/soc/codecs/twl6040.h @@ -0,0 +1,141 @@ +/* + * ALSA SoC TWL6040 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * 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 __TWL6040_H__ +#define __TWL6040_H__ + +#define TWL6040_REG_ASICID 0x01 +#define TWL6040_REG_ASICREV 0x02 +#define TWL6040_REG_INTID 0x03 +#define TWL6040_REG_INTMR 0x04 +#define TWL6040_REG_NCPCTL 0x05 +#define TWL6040_REG_LDOCTL 0x06 +#define TWL6040_REG_HPPLLCTL 0x07 +#define TWL6040_REG_LPPLLCTL 0x08 +#define TWL6040_REG_LPPLLDIV 0x09 +#define TWL6040_REG_AMICBCTL 0x0A +#define TWL6040_REG_DMICBCTL 0x0B +#define TWL6040_REG_MICLCTL 0x0C +#define TWL6040_REG_MICRCTL 0x0D +#define TWL6040_REG_MICGAIN 0x0E +#define TWL6040_REG_LINEGAIN 0x0F +#define TWL6040_REG_HSLCTL 0x10 +#define TWL6040_REG_HSRCTL 0x11 +#define TWL6040_REG_HSGAIN 0x12 +#define TWL6040_REG_EARCTL 0x13 +#define TWL6040_REG_HFLCTL 0x14 +#define TWL6040_REG_HFLGAIN 0x15 +#define TWL6040_REG_HFRCTL 0x16 +#define TWL6040_REG_HFRGAIN 0x17 +#define TWL6040_REG_VIBCTLL 0x18 +#define TWL6040_REG_VIBDATL 0x19 +#define TWL6040_REG_VIBCTLR 0x1A +#define TWL6040_REG_VIBDATR 0x1B +#define TWL6040_REG_HKCTL1 0x1C +#define TWL6040_REG_HKCTL2 0x1D +#define TWL6040_REG_GPOCTL 0x1E +#define TWL6040_REG_ALB 0x1F +#define TWL6040_REG_DLB 0x20 +#define TWL6040_REG_TRIM1 0x28 +#define TWL6040_REG_TRIM2 0x29 +#define TWL6040_REG_TRIM3 0x2A +#define TWL6040_REG_HSOTRIM 0x2B +#define TWL6040_REG_HFOTRIM 0x2C +#define TWL6040_REG_ACCCTL 0x2D +#define TWL6040_REG_STATUS 0x2E + +#define TWL6040_CACHEREGNUM (TWL6040_REG_STATUS + 1) + +#define TWL6040_VIOREGNUM 18 +#define TWL6040_VDDREGNUM 21 + +/* INTID (0x03) fields */ + +#define TWL6040_THINT 0x01 +#define TWL6040_PLUGINT 0x02 +#define TWL6040_UNPLUGINT 0x04 +#define TWL6040_HOOKINT 0x08 +#define TWL6040_HFINT 0x10 +#define TWL6040_VIBINT 0x20 +#define TWL6040_READYINT 0x40 + +/* INTMR (0x04) fields */ + +#define TWL6040_READYMSK 0x40 +#define TWL6040_ALLINT_MSK 0x7B + +/* NCPCTL (0x05) fields */ + +#define TWL6040_NCPENA 0x01 +#define TWL6040_NCPOPEN 0x40 + +/* LDOCTL (0x06) fields */ + +#define TWL6040_LSLDOENA 0x01 +#define TWL6040_HSLDOENA 0x04 +#define TWL6040_REFENA 0x40 +#define TWL6040_OSCENA 0x80 + +/* HPPLLCTL (0x07) fields */ + +#define TWL6040_HPLLENA 0x01 +#define TWL6040_HPLLRST 0x02 +#define TWL6040_HPLLBP 0x04 +#define TWL6040_HPLLSQRENA 0x08 +#define TWL6040_HPLLSQRBP 0x10 +#define TWL6040_MCLK_12000KHZ (0 << 5) +#define TWL6040_MCLK_19200KHZ (1 << 5) +#define TWL6040_MCLK_26000KHZ (2 << 5) +#define TWL6040_MCLK_38400KHZ (3 << 5) +#define TWL6040_MCLK_MSK 0x60 + +/* LPPLLCTL (0x08) fields */ + +#define TWL6040_LPLLENA 0x01 +#define TWL6040_LPLLRST 0x02 +#define TWL6040_LPLLSEL 0x04 +#define TWL6040_LPLLFIN 0x08 +#define TWL6040_HPLLSEL 0x10 + +/* HSLCTL (0x10) fields */ + +#define TWL6040_HSDACMODEL 0x02 +#define TWL6040_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6040_HSDACMODER 0x02 +#define TWL6040_HSDRVMODER 0x08 + +/* ACCCTL (0x2D) fields */ + +#define TWL6040_RESETSPLIT 0x04 + +#define TWL6040_SYSCLK_SEL_LPPLL 1 +#define TWL6040_SYSCLK_SEL_HPPLL 2 + +#define TWL6040_HPPLL_ID 1 +#define TWL6040_LPPLL_ID 2 + +extern struct snd_soc_dai twl6040_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl6040; + +#endif /* End of __TWL6040_H__ */ -- 1.6.1.3

3 2

[alsa-devel] [PATCHv2 1/2] OMAP4: PMIC: Rename twl6030_codec as twl6040_codec
by Olaya, Margarita 18 Mar '10

18 Mar '10

Correction for chips: twl6030 is Phoenix Power chip twl6040 is Phoenix Audio chip Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- drivers/mfd/twl-core.c | 4 ++-- include/linux/i2c/twl.h | 6 +++--- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/mfd/twl-core.c b/drivers/mfd/twl-core.c index 562cd49..720e099 100644 --- a/drivers/mfd/twl-core.c +++ b/drivers/mfd/twl-core.c @@ -109,7 +109,7 @@ #endif #if defined(CONFIG_TWL4030_CODEC) || defined(CONFIG_TWL4030_CODEC_MODULE) ||\ - defined(CONFIG_SND_SOC_TWL6030) || defined(CONFIG_SND_SOC_TWL6030_MODULE) + defined(CONFIG_SND_SOC_TWL6040) || defined(CONFIG_SND_SOC_TWL6040_MODULE) #define twl_has_codec() true #else #define twl_has_codec() false @@ -708,7 +708,7 @@ add_children(struct twl4030_platform_data *pdata, unsigned long features) /* Phoenix*/ if (twl_has_codec() && pdata->codec && twl_class_is_6030()) { sub_chip_id = twl_map[TWL_MODULE_AUDIO_VOICE].sid; - child = add_child(sub_chip_id, "twl6030_codec", + child = add_child(sub_chip_id, "twl6040_codec", pdata->codec, sizeof(*pdata->codec), false, 0, 0); if (IS_ERR(child)) diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h index fb6784e..ebd90ce 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -569,9 +569,9 @@ struct twl4030_codec_data { struct twl4030_codec_audio_data *audio; struct twl4030_codec_vibra_data *vibra; - /* twl6030 */ - int audpwron_gpio; /* audio power-on gpio */ - int naudint_irq; /* audio interrupt */ + /* twl6040 */ + int audpwron_gpio; /* audio power-on gpio */ + int naudint_irq; /* audio interrupt */ }; struct twl4030_platform_data { -- 1.6.1.3

4 4

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 18 Mar '10

18 Mar '10

The following changes since commit dd76769dd53bdb804a4b02a3eb256f4740f78720: Mark Brown (1): ASoC: Refresh WM8750 bias management are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Barry Song (1): ASoC: soc-cache: let reg be AND'ed by 0xff instead of data buffer for 8_8 mode Cliff Cai (1): ASoC: soc-cache: add i2c read entry for 8_8 mode sound/soc/soc-cache.c | 39 ++++++++++++++++++++++++++++++++++++++- 1 files changed, 38 insertions(+), 1 deletions(-)

2 1

[alsa-devel] ASoC update for 2.6.34
by Mark Brown 18 Mar '10

18 Mar '10

The following changes since commit 57d54889cd00db2752994b389ba714138652e60c: Linus Torvalds (1): Linux 2.6.34-rc1 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Guennadi Liakhovetski (1): ASoC: SIU driver shall select FW_LOADER sound/soc/sh/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-)

2 1

[alsa-devel] [PATCH 0/2] Support for button reporting on jacks
by Mark Brown 18 Mar '10

18 Mar '10

This patch series adds support for key reporting to the generic ALSA jack API. Currently a series of switches are supported, with the jack implementation assuming that only switches will be used. This isn't ideal for things like the simple buttons implemented in headsets which are used in the same way as keyboard keys and physically resemble them more closely than switches. If this is applied I'd like if possible to merge the topic into the ASoC tree so that I can use it in machine drivers there. The following changes since commit 57d54889cd00db2752994b389ba714138652e60c: Linus Torvalds (1): Linux 2.6.34-rc1 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git topic/jack Mark Brown (2): ALSA: Rename jack switch table in preparation for button support ALSA: Add support for key reporting via the jack interface include/sound/jack.h | 8 +++++ sound/core/jack.c | 71 +++++++++++++++++++++++++++++++++++++++++++++---- 2 files changed, 73 insertions(+), 6 deletions(-)

2 3

[alsa-devel] HDMI output with headphone jack sense possible on Conexant CX20561?
by Andrew Udvare 18 Mar '10

18 Mar '10

My audio codec is Conexant CX20561 (Hermosa) (haven't been able to find anything about 'Hermosa' on the Wiki). I'd like to be able to do a few things which I have not been able to do with tutorials I found on the web: - Whenever I plug in headphones, HDMI output is muted and headphones are enabled - Right now, the way I do this is by manually editing /etc/asound.conf to have the following (commented out makes headphones work, not commented makes HDMI work without any headphone/laptop speaker output) /etc/asound.conf: defaults.pcm.card 0 defaults.pcm.device 3 where 3 is my HDMI (IEC958 2; no idea what IEC958 1 is for) output. So I have a switcher script that does the commenting and uncommenting. Would be nice if this could be automatic (even if alsa had to run such a script on detecting headphones). - Route audio output (PCM) to input (for mixer recording) (Intel HDA cannot do this apparently by default (thanks DRM), needs some hacking around?) Information from alsa-info.sh: http://www.alsa- project.org/db/?f=107f3f6012e68f04d3562d5e11aeb052a1454884 Are my model options correct? And if it's important to know, I build ALSA in-kernel. Is this not recommended? I know at the very least it makes it such that I have to reboot to make any new options to take effect. Thanks all -- Andrew Udvare audvare(a)gmail.com 978-870-5908 www.andyudvare.com

1 0

[alsa-devel] [PATCH] ASoC: SIU driver shall select FW_LOADER
by Guennadi Liakhovetski 18 Mar '10

18 Mar '10

The SIU ASoC driver must load firmware to program the DSP, therefore it has to select FW_LOADER in its Kconfig entry. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 1066749..f07f6d8 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_SH4_SIU select DMA_ENGINE select DMADEVICES select SH_DMAE + select FW_LOADER ## ## Boards

3 2

[alsa-devel] [PATCH] ASoC: remove a card from the list, if instantiation failed
by Guennadi Liakhovetski 18 Mar '10

18 Mar '10

If instantiation of a card failed, we have to remove it from the card list. This fixes an Oops on Migo-R, triggering, when after a failed firmware load attempt the driver modules are removed and re-inserted again. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- Should also be good for stable. diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c8b0556..307eea0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2361,6 +2361,8 @@ static int snd_soc_register_card(struct snd_soc_card *card) mutex_lock(&client_mutex); list_add(&card->list, &card_list); snd_soc_instantiate_cards(); + if (!card->instantiated) + list_del(&card->list); mutex_unlock(&client_mutex); dev_dbg(card->dev, "Registered card '%s'\n", card->name);

2 3

[alsa-devel] [PATCHv6 0/2] TWL6040 audio codec initial support
by Olaya, Margarita 17 Mar '10

17 Mar '10

Following patch series add initial support for TWL6040 codec driver. The patch set is based on the series sent from Misael Lopez in October 2009. Changes from v5: - Add CONFIG_PM for twl6040_suspend and twl6030_resume functions - Add description for patch 1 Regards, Margarita --- Misael Lopez Cruz (1): ASoC: TWL6040: Add twl6040 codec driver Margarita Olaya Cabrera (1): OMAP4: PMIC: Rename twl6030_codec as twl6040_codec drivers/mfd/twl-core.c | 4 +- include/linux/i2c/twl.h | 6 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6040.c | 1227 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 141 +++++ 6 files changed, 1379 insertions(+), 5 deletions(-) create mode 100644 sound/soc/codecs/twl6040.c create mode 100644 sound/soc/codecs/twl6040.h

1 0

[PATCH] ASoC: support pxa168 ssp in ASoC
by Haojian Zhuang 17 Mar '10

17 Mar '10

Support pxa168 ssp in ASoC. The SSP bit clock mechanism is totally different from pxa2xx series. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- arch/arm/mach-mmp/include/mach/regs-mpmu.h | 48 ++++++ sound/soc/pxa/Kconfig | 6 +- sound/soc/pxa/Makefile | 2 + sound/soc/pxa/pxa-ssp.h | 1 + sound/soc/pxa/pxa168-ssp.c | 233 ++++++++++++++++++++++++++++ sound/soc/pxa/pxa168-ssp.h | 27 ++++ 6 files changed, 316 insertions(+), 1 deletions(-) create mode 100644 arch/arm/mach-mmp/include/mach/regs-mpmu.h create mode 100644 sound/soc/pxa/pxa168-ssp.c create mode 100644 sound/soc/pxa/pxa168-ssp.h diff --git a/arch/arm/mach-mmp/include/mach/regs-mpmu.h b/arch/arm/mach-mmp/include/mach/regs-mpmu.h new file mode 100644 index 0000000..0d57236 --- /dev/null +++ b/arch/arm/mach-mmp/include/mach/regs-mpmu.h @@ -0,0 +1,48 @@ +/* + * linux/arch/arm/mach-mmp/include/mach/regs-mpmu.h + * + * Main Power Management Unit + * + * 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 __ASM_MACH_REGS_MPMU_H +#define __ASM_MACH_REGS_MPMU_H + +#include <mach/addr-map.h> + +#define MPMU_VIRT_BASE (APB_VIRT_BASE + 0x50000) +#define MPMU_REG(off) (MPMU_VIRT_BASE + (off)) + +#define MPMU_CPCR MPMU_REG(0x0000) +#define MPMU_FCCR MPMU_REG(0x0008) +#define MPMU_POCR MPMU_REG(0x000c) +#define MPMU_POSR MPMU_REG(0x0010) +#define MPMU_SUCCR MPMU_REG(0x0014) +#define MPMU_VRCR MPMU_REG(0x0018) +#define MPMU_OHCR MPMU_REG(0x001c) +#define MPMU_GPCR MPMU_REG(0x0030) +#define MPMU_PLL2CR MPMU_REG(0x0034) +#define MPMU_SCCR MPMU_REG(0x0038) +#define MPMU_CWUCRM MPMU_REG(0x004c) +#define MPMU_PLL1_REG1 MPMU_REG(0x0050) +#define MPMU_PLL1_REG2 MPMU_REG(0x0054) +#define MPMU_PLL1_SSC MPMU_REG(0x0058) +#define MPMU_PLL2_REG1 MPMU_REG(0x0060) +#define MPMU_PLL2_REG2 MPMU_REG(0x0064) +#define MPMU_PLL2_SSC MPMU_REG(0x0068) +#define MPMU_TS MPMU_REG(0x0080) +#define MPMU_WDTPCR MPMU_REG(0x0200) +#define MPMU_APCR MPMU_REG(0x1000) +#define MPMU_APSR MPMU_REG(0x1004) +#define MPMU_APRR MPMU_REG(0x1020) +#define MPMU_ACGR MPMU_REG(0x1024) +#define MPMU_ARSR MPMU_REG(0x1028) +#define MPMU_AWUCRS MPMU_REG(0x1048) +#define MPMU_AWUCRM MPMU_REG(0x104c) +#define MPMU_ASYSDR MPMU_REG(0x1050) +#define MPMU_ASSPDR MPMU_REG(0x1054) + +#endif /* __ASM_MACH_REGS_APMU_H */ diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 7be1d5f..286d52a 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,6 +1,6 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" - depends on ARCH_PXA + depends on ARCH_PXA || ARCH_MMP select SND_PXA2XX_LIB help Say Y or M if you want to add support for codecs attached to @@ -25,6 +25,10 @@ config SND_PXA2XX_SOC_SSP tristate select PXA_SSP +config SND_PXA168_SOC_SSP + tristate + select PXA_SSP + config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 33c1579..a74e6c9 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -3,11 +3,13 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o snd-soc-pxa2xx-ssp-objs := pxa-ssp.o pxa2xx-ssp.o +snd-soc-pxa168-ssp-objs := pxa-ssp.o pxa168-ssp.o obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o obj-$(CONFIG_SND_PXA2XX_SOC_SSP) += snd-soc-pxa2xx-ssp.o +obj-$(CONFIG_SND_PXA168_SOC_SSP) += snd-soc-pxa168-ssp.o # PXA Machine Support snd-soc-corgi-objs := corgi.o diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h index af4a09b..852660a 100644 --- a/sound/soc/pxa/pxa-ssp.h +++ b/sound/soc/pxa/pxa-ssp.h @@ -15,6 +15,7 @@ struct ssp_priv { struct ssp_device *ssp; unsigned int sysclk; + unsigned int sysclk_idx; int dai_fmt; #ifdef CONFIG_PM uint32_t cr0; diff --git a/sound/soc/pxa/pxa168-ssp.c b/sound/soc/pxa/pxa168-ssp.c new file mode 100644 index 0000000..487e899 --- /dev/null +++ b/sound/soc/pxa/pxa168-ssp.c @@ -0,0 +1,233 @@ +/* + * pxa168-ssp.c -- ALSA Soc Audio Layer + * + * Copyright 2009-2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * TODO: + * o Test network mode + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/pxa2xx-lib.h> + +#include <mach/hardware.h> +#include <mach/dma.h> +#include <mach/regs-apbc.h> +#include <mach/regs-apmu.h> +#include <mach/regs-mpmu.h> +#include <plat/ssp.h> + +#include "pxa2xx-pcm.h" +#include "pxa168-ssp.h" +#include "pxa-ssp.h" + +struct pxa168_ssp_mclk { + unsigned int rate; + unsigned int format; + unsigned int channel; + unsigned int mclk; + unsigned int mclk_denom; + unsigned int mclk_num; + unsigned int bclk; + unsigned int bclk_denom; + unsigned int bclk_num; +}; + +/* + * This table is used while CPU is clock master. + * MCLK = 312MHz * ASYSCLK_DENOM / ASYSCLK_NUM + * BCLK = MCLK * SSPSCLK_DENOM / SSPSCLK_NUM + */ +static const struct pxa168_ssp_mclk mclk_conf[] = { + /* rate, fmt, chn, mclk, den, num, bclk, den, num */ + {96000, 16, 2, 12288000, 64, 1625, 6144000, 1, 2}, + {96000, 16, 1, 12288000, 64, 1625, 1536000, 1, 8}, + {88200, 16, 2, 11289600, 294, 8125, 5644800, 1, 2}, + {88200, 16, 1, 11289600, 294, 8125, 1411200, 1, 8}, + {48000, 16, 2, 12288000, 64, 1625, 3072000, 1, 4}, + {48000, 16, 1, 12288000, 64, 1625, 768000, 1, 16}, + {44100, 16, 2, 11289600, 294, 8125, 2822400, 1, 4}, + {44100, 16, 1, 11289600, 294, 8125, 705600, 1, 16}, + {32000, 16, 2, 12288000, 64, 1625, 2048000, 1, 6}, + {32000, 16, 1, 12288000, 64, 1625, 512000, 1, 24}, + {22050, 16, 2, 11289600, 294, 8125, 1411200, 1, 8}, + {22050, 16, 1, 11289600, 294, 8125, 352800, 1, 32}, + {16000, 16, 2, 12288000, 64, 1625, 1024000, 1, 12}, + {16000, 16, 1, 12288000, 64, 1625, 256000, 1, 48}, + {11025, 16, 2, 11289600, 294, 8125, 705600, 1, 16}, + {11025, 16, 1, 11289600, 294, 8125, 176400, 1, 64}, + { 8000, 16, 2, 12288000, 64, 1625, 512000, 1, 24}, + { 8000, 16, 1, 12288000, 64, 1625, 128000, 1, 96}, +}; + +int pxa168_query_mclk(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_priv *priv = cpu_dai->private_data; + unsigned int rate, width, channel; + int i, ret = -EINVAL; + + rate = params_rate(params); + width = snd_pcm_format_physical_width(params_format(params)); + channel = params_channels(params); + + for (i = 0; i < ARRAY_SIZE(mclk_conf); i++) { + if ((mclk_conf[i].rate == rate) + && (mclk_conf[i].format == width) + && (mclk_conf[i].channel == channel)) { + /* save both mclk and mclk index into ssp_priv */ + priv->sysclk = mclk_conf[i].mclk; + priv->sysclk_idx = i; + ret = priv->sysclk; + break; + } + } + return ret; +} +EXPORT_SYMBOL_GPL(pxa168_query_mclk); + +/* + * Set the SSP ports SYSCLK only from Audio SYSCLK. + */ +static int pxa168_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + unsigned int sscr0, data, asysdr, asspdr; + + dev_dbg(&ssp->pdev->dev, "%s id: %d, clk_id %d, freq %u\n", + __func__, cpu_dai->id, clk_id, freq); + + /* freq is the index of mclk_conf table */ + if ((freq < 0) || (priv->sysclk_idx >= ARRAY_SIZE(mclk_conf))) { + dev_warn(&ssp->pdev->dev, "Wrong frequency on SYSCLK\n"); + return -EINVAL; + } + asysdr = (mclk_conf[priv->sysclk_idx].mclk_num << 16) + | mclk_conf[priv->sysclk_idx].mclk_denom; + asspdr = 0; + /* If ASYSCLK is supplied by pxa168, ASSPDR should be configured. */ + if (dir == SND_SOC_CLOCK_OUT) + asspdr = (mclk_conf[priv->sysclk_idx].bclk_num << 16) + | mclk_conf[priv->sysclk_idx].bclk_denom; + + pxa_ssp_disable(ssp); + clk_disable(ssp->clk); /* SSP port internal clock */ + + /* clear ECS, NCS, MOD, ACS */ + sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + data = sscr0 & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + if (sscr0 != data) + pxa_ssp_write_reg(ssp, SSCR0, data); + + /* update divider register in MPMU */ + __raw_writel(asysdr, MPMU_ASYSDR); + __raw_writel(asspdr, MPMU_ASSPDR); + + clk_enable(ssp->clk); /* SSP port internal clock */ + pxa_ssp_enable(ssp); + return 0; +} + +static int pxa168_ssp_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + + pxa_ssp_disable(ssp); + /* update divider register in MPMU */ + __raw_writel(0, MPMU_ASYSDR); + __raw_writel(0, MPMU_ASSPDR); + return 0; +} + +static struct snd_soc_dai_ops pxa168_ssp_dai_ops = { + .hw_free = pxa168_ssp_hw_free, + .set_sysclk = pxa168_ssp_set_dai_sysclk, +}; + +#define PXA168_SSP_RATES SNDRV_PCM_RATE_8000_96000 +#define PXA168_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define PXA168_SSP_DAI(_id) \ +{ \ + .name = "pxa168-ssp", \ + .id = _id, \ + .playback = { \ + .channels_min = 1, \ + .channels_max = 2, \ + .rates = PXA168_SSP_RATES, \ + .formats = PXA168_SSP_FORMATS, \ + }, \ + .capture = { \ + .channels_min = 1, \ + .channels_max = 2, \ + .rates = PXA168_SSP_RATES, \ + .formats = PXA168_SSP_FORMATS, \ + }, \ + .ops = &pxa168_ssp_dai_ops, \ +} + +struct snd_soc_dai pxa168_ssp_dai[] = { + PXA168_SSP_DAI(PXA168_DAI_SSP1), + PXA168_SSP_DAI(PXA168_DAI_SSP2), + PXA168_SSP_DAI(PXA168_DAI_SSP3), + PXA168_SSP_DAI(PXA168_DAI_SSP4), + PXA168_SSP_DAI(PXA168_DAI_SSP5), +}; +EXPORT_SYMBOL_GPL(pxa168_ssp_dai); + +static int __init pxa168_ssp_init(void) +{ + struct snd_soc_dai *dai; + int i, ret; + + for (i = 0; i < ARRAY_SIZE(pxa168_ssp_dai); i++) { + dai = &pxa168_ssp_dai[i]; + ret = pxa_ssp_register_dai(dai); + if (ret) + return ret; + } + return ret; +} +module_init(pxa168_ssp_init); + +static void __exit pxa168_ssp_exit(void) +{ + struct snd_soc_dai *dai = NULL; + int i; + + for (i = 0; i < ARRAY_SIZE(pxa168_ssp_dai); i++) { + dai = &pxa168_ssp_dai[i]; + snd_soc_unregister_dai(dai); + } +} +module_exit(pxa168_ssp_exit); + +/* Module information */ +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_DESCRIPTION("PXA168 SSP SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/pxa/pxa168-ssp.h b/sound/soc/pxa/pxa168-ssp.h new file mode 100644 index 0000000..c7e23c7 --- /dev/null +++ b/sound/soc/pxa/pxa168-ssp.h @@ -0,0 +1,27 @@ +/* + * ASoC PXA168 SSP port support + * + * 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 _PXA168_SSP_H +#define _PXA168_SSP_H + +/* pxa DAI SSP IDs */ +#define PXA168_DAI_SSP1 0 +#define PXA168_DAI_SSP2 1 +#define PXA168_DAI_SSP3 2 +#define PXA168_DAI_SSP4 3 +#define PXA168_DAI_SSP5 4 + +/* PXA168 SSP SYSCLK source */ +#define PXA168_ASYSCLK_MASTER 0 /* ASYSCLK master -- pxa168 */ +#define PXA168_ASYSCLK_SLAVE 1 /* ASYSCLK slave -- pxa168 */ + +extern struct snd_soc_dai pxa168_ssp_dai[]; + +extern int pxa168_query_mclk(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params); +#endif -- 1.5.6.5

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[PATCH] ASoC: support pxa168 ssp in ASoC
by Haojian Zhuang 17 Mar '10

17 Mar '10

Support pxa168 ssp in ASoC. The SSP bit clock mechanism is totally different from pxa2xx series. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- arch/arm/mach-mmp/include/mach/regs-mpmu.h | 48 ++++++++ sound/soc/pxa/Kconfig | 6 +- sound/soc/pxa/Makefile | 2 + sound/soc/pxa/pxa168-ssp.c | 166 ++++++++++++++++++++++++++++ sound/soc/pxa/pxa168-ssp.h | 64 +++++++++++ 5 files changed, 285 insertions(+), 1 deletions(-) create mode 100644 arch/arm/mach-mmp/include/mach/regs-mpmu.h create mode 100644 sound/soc/pxa/pxa168-ssp.c create mode 100644 sound/soc/pxa/pxa168-ssp.h diff --git a/arch/arm/mach-mmp/include/mach/regs-mpmu.h b/arch/arm/mach-mmp/include/mach/regs-mpmu.h new file mode 100644 index 0000000..0d57236 --- /dev/null +++ b/arch/arm/mach-mmp/include/mach/regs-mpmu.h @@ -0,0 +1,48 @@ +/* + * linux/arch/arm/mach-mmp/include/mach/regs-mpmu.h + * + * Main Power Management Unit + * + * 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 __ASM_MACH_REGS_MPMU_H +#define __ASM_MACH_REGS_MPMU_H + +#include <mach/addr-map.h> + +#define MPMU_VIRT_BASE (APB_VIRT_BASE + 0x50000) +#define MPMU_REG(off) (MPMU_VIRT_BASE + (off)) + +#define MPMU_CPCR MPMU_REG(0x0000) +#define MPMU_FCCR MPMU_REG(0x0008) +#define MPMU_POCR MPMU_REG(0x000c) +#define MPMU_POSR MPMU_REG(0x0010) +#define MPMU_SUCCR MPMU_REG(0x0014) +#define MPMU_VRCR MPMU_REG(0x0018) +#define MPMU_OHCR MPMU_REG(0x001c) +#define MPMU_GPCR MPMU_REG(0x0030) +#define MPMU_PLL2CR MPMU_REG(0x0034) +#define MPMU_SCCR MPMU_REG(0x0038) +#define MPMU_CWUCRM MPMU_REG(0x004c) +#define MPMU_PLL1_REG1 MPMU_REG(0x0050) +#define MPMU_PLL1_REG2 MPMU_REG(0x0054) +#define MPMU_PLL1_SSC MPMU_REG(0x0058) +#define MPMU_PLL2_REG1 MPMU_REG(0x0060) +#define MPMU_PLL2_REG2 MPMU_REG(0x0064) +#define MPMU_PLL2_SSC MPMU_REG(0x0068) +#define MPMU_TS MPMU_REG(0x0080) +#define MPMU_WDTPCR MPMU_REG(0x0200) +#define MPMU_APCR MPMU_REG(0x1000) +#define MPMU_APSR MPMU_REG(0x1004) +#define MPMU_APRR MPMU_REG(0x1020) +#define MPMU_ACGR MPMU_REG(0x1024) +#define MPMU_ARSR MPMU_REG(0x1028) +#define MPMU_AWUCRS MPMU_REG(0x1048) +#define MPMU_AWUCRM MPMU_REG(0x104c) +#define MPMU_ASYSDR MPMU_REG(0x1050) +#define MPMU_ASSPDR MPMU_REG(0x1054) + +#endif /* __ASM_MACH_REGS_APMU_H */ diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 7be1d5f..286d52a 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,6 +1,6 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" - depends on ARCH_PXA + depends on ARCH_PXA || ARCH_MMP select SND_PXA2XX_LIB help Say Y or M if you want to add support for codecs attached to @@ -25,6 +25,10 @@ config SND_PXA2XX_SOC_SSP tristate select PXA_SSP +config SND_PXA168_SOC_SSP + tristate + select PXA_SSP + config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 33c1579..a74e6c9 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -3,11 +3,13 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o snd-soc-pxa2xx-ssp-objs := pxa-ssp.o pxa2xx-ssp.o +snd-soc-pxa168-ssp-objs := pxa-ssp.o pxa168-ssp.o obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o obj-$(CONFIG_SND_PXA2XX_SOC_SSP) += snd-soc-pxa2xx-ssp.o +obj-$(CONFIG_SND_PXA168_SOC_SSP) += snd-soc-pxa168-ssp.o # PXA Machine Support snd-soc-corgi-objs := corgi.o diff --git a/sound/soc/pxa/pxa168-ssp.c b/sound/soc/pxa/pxa168-ssp.c new file mode 100644 index 0000000..d28d18a --- /dev/null +++ b/sound/soc/pxa/pxa168-ssp.c @@ -0,0 +1,166 @@ +/* + * pxa168-ssp.c -- ALSA Soc Audio Layer + * + * Copyright 2009-2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * TODO: + * o Test network mode + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/pxa2xx-lib.h> + +#include <mach/hardware.h> +#include <mach/dma.h> +#include <mach/regs-apbc.h> +#include <mach/regs-apmu.h> +#include <mach/regs-mpmu.h> +#include <plat/ssp.h> + +#include "pxa2xx-pcm.h" +#include "pxa168-ssp.h" +#include "pxa-ssp.h" + +/* + * Set the SSP ports SYSCLK only from Audio SYSCLK. + */ +static int pxa168_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + unsigned int sscr0, data, asysdr, asspdr; + + dev_dbg(&ssp->pdev->dev, "%s id: %d, clk_id %d, freq %u\n", + __func__, cpu_dai->id, clk_id, freq); + + /* freq is the index of mclk_conf table */ + if ((freq < 0) || (freq >= ARRAY_SIZE(mclk_conf))) { + dev_warn(&ssp->pdev->dev, "Wrong frequency index:%d\n", freq); + return -EINVAL; + } + asysdr = (mclk_conf[freq].mclk_num << 16) + | mclk_conf[freq].mclk_denom; + asspdr = 0; + /* If ASYSCLK is supplied by pxa168, ASSPDR should be configured. */ + if (dir == SND_SOC_CLOCK_OUT) + asspdr = (mclk_conf[freq].bclk_num << 16) + | mclk_conf[freq].bclk_denom; + + pxa_ssp_disable(ssp); + clk_disable(ssp->clk); /* SSP port internal clock */ + + /* clear ECS, NCS, MOD, ACS */ + sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + data = sscr0 & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + if (sscr0 != data) + pxa_ssp_write_reg(ssp, SSCR0, data); + + /* update divider register in MPMU */ + __raw_writel(asysdr, MPMU_ASYSDR); + __raw_writel(asspdr, MPMU_ASSPDR); + + clk_enable(ssp->clk); /* SSP port internal clock */ + pxa_ssp_enable(ssp); + return 0; +} + +static int pxa168_ssp_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + + pxa_ssp_disable(ssp); + /* update divider register in MPMU */ + __raw_writel(0, MPMU_ASYSDR); + __raw_writel(0, MPMU_ASSPDR); + return 0; +} + +static struct snd_soc_dai_ops pxa168_ssp_dai_ops = { + .hw_free = pxa168_ssp_hw_free, + .set_sysclk = pxa168_ssp_set_dai_sysclk, +}; + +#define PXA168_SSP_RATES SNDRV_PCM_RATE_8000_96000 +#define PXA168_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define PXA168_SSP_DAI(_id) \ +{ \ + .name = "pxa168-ssp", \ + .id = _id, \ + .playback = { \ + .channels_min = 1, \ + .channels_max = 2, \ + .rates = PXA168_SSP_RATES, \ + .formats = PXA168_SSP_FORMATS, \ + }, \ + .capture = { \ + .channels_min = 1, \ + .channels_max = 2, \ + .rates = PXA168_SSP_RATES, \ + .formats = PXA168_SSP_FORMATS, \ + }, \ + .ops = &pxa168_ssp_dai_ops, \ +} + +struct snd_soc_dai pxa168_ssp_dai[] = { + PXA168_SSP_DAI(PXA168_DAI_SSP1), + PXA168_SSP_DAI(PXA168_DAI_SSP2), + PXA168_SSP_DAI(PXA168_DAI_SSP3), + PXA168_SSP_DAI(PXA168_DAI_SSP4), + PXA168_SSP_DAI(PXA168_DAI_SSP5), +}; +EXPORT_SYMBOL_GPL(pxa168_ssp_dai); + +static int __init pxa168_ssp_init(void) +{ + struct snd_soc_dai *dai; + int i, ret; + + for (i = 0; i < ARRAY_SIZE(pxa168_ssp_dai); i++) { + dai = &pxa168_ssp_dai[i]; + ret = pxa_ssp_register_dai(dai); + if (ret) + return ret; + } + return ret; +} +module_init(pxa168_ssp_init); + +static void __exit pxa168_ssp_exit(void) +{ + struct snd_soc_dai *dai = NULL; + int i; + + for (i = 0; i < ARRAY_SIZE(pxa168_ssp_dai); i++) { + dai = &pxa168_ssp_dai[i]; + snd_soc_unregister_dai(dai); + } +} +module_exit(pxa168_ssp_exit); + +/* Module information */ +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_DESCRIPTION("PXA168 SSP SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/pxa/pxa168-ssp.h b/sound/soc/pxa/pxa168-ssp.h new file mode 100644 index 0000000..bf73137 --- /dev/null +++ b/sound/soc/pxa/pxa168-ssp.h @@ -0,0 +1,64 @@ +/* + * ASoC PXA168 SSP port support + * + * 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 _PXA168_SSP_H +#define _PXA168_SSP_H + +/* pxa DAI SSP IDs */ +#define PXA168_DAI_SSP1 0 +#define PXA168_DAI_SSP2 1 +#define PXA168_DAI_SSP3 2 +#define PXA168_DAI_SSP4 3 +#define PXA168_DAI_SSP5 4 + +/* PXA168 SSP SYSCLK source */ +#define PXA168_ASYSCLK_MASTER 0 /* ASYSCLK master -- pxa168 */ +#define PXA168_ASYSCLK_SLAVE 1 /* ASYSCLK slave -- pxa168 */ + +struct pxa168_ssp_mclk { + unsigned int rate; + unsigned int format; + unsigned int channel; + unsigned int mclk; + unsigned int mclk_denom; + unsigned int mclk_num; + unsigned int bclk; + unsigned int bclk_denom; + unsigned int bclk_num; +}; + +/* + * This table is used while CPU is clock master. + * MCLK = 312MHz * ASYSCLK_DENOM / ASYSCLK_NUM + * BCLK = MCLK * SSPSCLK_DENOM / SSPSCLK_NUM + */ +static const struct pxa168_ssp_mclk mclk_conf[] = { + /* rate, fmt, chn, mclk, den, num, bclk, den, num */ + {96000, 16, 2, 12288000, 64, 1625, 6144000, 1, 2}, + {96000, 16, 1, 12288000, 64, 1625, 1536000, 1, 8}, + {88200, 16, 2, 11289600, 294, 8125, 5644800, 1, 2}, + {88200, 16, 1, 11289600, 294, 8125, 1411200, 1, 8}, + {48000, 16, 2, 12288000, 64, 1625, 3072000, 1, 4}, + {48000, 16, 1, 12288000, 64, 1625, 768000, 1, 16}, + {44100, 16, 2, 11289600, 294, 8125, 2822400, 1, 4}, + {44100, 16, 1, 11289600, 294, 8125, 705600, 1, 16}, + {32000, 16, 2, 12288000, 64, 1625, 2048000, 1, 6}, + {32000, 16, 1, 12288000, 64, 1625, 512000, 1, 24}, + {22050, 16, 2, 11289600, 294, 8125, 1411200, 1, 8}, + {22050, 16, 1, 11289600, 294, 8125, 352800, 1, 32}, + {16000, 16, 2, 12288000, 64, 1625, 1024000, 1, 12}, + {16000, 16, 1, 12288000, 64, 1625, 256000, 1, 48}, + {11025, 16, 2, 11289600, 294, 8125, 705600, 1, 16}, + {11025, 16, 1, 11289600, 294, 8125, 176400, 1, 64}, + { 8000, 16, 2, 12288000, 64, 1625, 512000, 1, 24}, + { 8000, 16, 1, 12288000, 64, 1625, 128000, 1, 96}, +}; + +extern struct snd_soc_dai pxa168_ssp_dai[]; + +#endif -- 1.5.6.5

1 0

[alsa-devel] How to create an Alsa mixer for a USB device w/o mixer descriptor?
by Felix Homann 17 Mar '10

17 Mar '10

Hi, as posted earlier, I've got a USB device (Fast Track Ultra 8R) wich *has* a mixer unit but doesn't have a descriptor for it. It's operated by standard control messages. Although I'm specifically interested in my device, I guess there are many more USB interfaces out there with similar issues. So in general: What's the best way to build a Alsa mixer for such devices. Should one bypass all the descriptor parsing and build a mixer specifically for the device in interest? Or should one think about passing a "fake" descriptor to the mixer build process? Any ideas? Kind regards, Felix

2 3

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 17 Mar '10

17 Mar '10

The following changes since commit fbc2dae854da5fc44e2ce2f97446c8bd283c7e17: Mark Brown (1): ASoC: Support GPIO based microphone detection for WM8904 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Mark Brown (3): ASoC: Implement WM835x microphone jack detection support ASoC: Remove current PGA control handling ASoC: Refresh WM8750 bias management The PGA control handling is the build issue fix. include/linux/mfd/wm8350/audio.h | 2 + sound/soc/codecs/wm8350.c | 58 +++++++++++++++++++++++ sound/soc/codecs/wm8350.h | 3 + sound/soc/codecs/wm8750.c | 50 ++++----------------- sound/soc/soc-dapm.c | 93 +------------------------------------ 5 files changed, 75 insertions(+), 131 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 17 Mar '10

17 Mar '10

The following changes since commit 75581d2459d3e6e5e796a42b362e4abb612bc958: Peter Ujfalusi (1): ASoC: OMAP3: Report delay caused by the internal FIFO are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Pretty much all of which is the first bit of some improvements to ASoC jack detection, including implementation in some drivers. Mark Brown (9): ASoC: Remove version display from WM8750 ASoC: Remove unused 'muted' flag from DAPM widgets ASoC: Allow pins to be force enabled ASoC: Add GPIO configuration support for WM8903 ASoC: Initial WM8903 microphone bias and short detection ASoC: Add WM8903 interrupt support ASoC: Implement interrupt driven microphone detection for WM8903 ASoC: Allow configuration of WM8904 GPIO pin functions ASoC: Support GPIO based microphone detection for WM8904 include/sound/soc-dapm.h | 4 +- include/sound/wm8903.h | 249 +++++++++++++++++++++++++++++++++++++++++++++ include/sound/wm8904.h | 110 +++++++++++++++++++- sound/soc/codecs/wm8750.c | 3 - sound/soc/codecs/wm8903.c | 201 ++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm8903.h | 221 +--------------------------------------- sound/soc/codecs/wm8904.c | 17 +++ sound/soc/codecs/wm8904.h | 97 ------------------ sound/soc/soc-dapm.c | 35 ++++++- 9 files changed, 603 insertions(+), 334 deletions(-) create mode 100644 include/sound/wm8903.h

2 4

[alsa-devel] snd-hda-intel: no data when recording from mic, "IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj" logged in kern.log
by Aleksander Adamowski 17 Mar '10

17 Mar '10

Hi! I think I have some interesting problems to analyse for Takashi. Maybe my sound card model is a non typical one, but I have a continuous stream of capture-related problems with it. It's an integrated sound card on my Gigabyte GA-MA74GM-S2H Rev. 1.1 motherboard. "grep ^Codec /proc/asound/card0/codec#0" shows: Codec: Realtek ALC888 I'm attaching the full contents of /proc/asound/card0/codec#0 for identification. But first, some background. With this particular card, I've been continuously hit by the "azx_get_response timeout, switching to single_cmd mode" problem, described here: http://www.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.html#_cod… However, in my case setting probe_mask=1 (experimentally determined) didn't make the problem go away, although it seemed to reduce its frequency (might be just a coincidence). Whenever the dreaded "azx_get_response timeout, switching to single_cmd mode" message appeared in the kernel log, recording stopped working. The card got into the bad state where it would not record sound, and arecord would be giving a stream of "overrun" error messages: arecord -f cd output.wav Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo overrun!!! (at least 27.876 ms long) overrun!!! (at least 26.965 ms long) overrun!!! (at least 16.169 ms long) overrun!!! (at least 11.749 ms long) overrun!!! (at least 11.298 ms long) overrun!!! (at least 11.814 ms long) overrun!!! (at least 11.190 ms long) overrun!!! (at least 11.190 ms long) overrun!!! (at least 11.747 ms long) overrun!!! (at least 11.819 ms long) overrun!!! (at least 11.149 ms long) overrun!!! (at least 11.257 ms long) overrun!!! (at least 11.315 ms long) overrun!!! (at least 11.651 ms long) overrun!!! (at least 11.745 ms long) overrun!!! (at least 11.235 ms long) overrun!!! (at least 11.169 ms long) overrun!!! (at least 11.776 ms long) overrun!!! (at least 11.391 ms long) overrun!!! (at least 11.669 ms long) overrun!!! (at least 11.714 ms long) overrun!!! (at least 11.671 ms long) overrun!!! (at least 11.190 ms long) What's notable, sound playback would continue to work OK. After being fed up with interrupted Skype conversations (where I weren't immediately aware that the other side stopped hearing me, since I was hearing them), I've decided to upgrade do kernel 2.6.29 (citing the http://www.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.html#_cod… document: "Since 2.6.29 kernel, the driver has a more robust probing method, so this error might happen rarely, though"). After upgrading to 2.6.29 I've removed any extra snd-hda-intel options from modprobe config files. Now the azx_get_response timeout seems to be gone, but I'm experiencing another problem: after some powered up time, recording starts giving empty streams - there's no error, but audio data has zero lengh. e.g. when I arecord, I get a .wav file that is only 44 bytes long (which seems to be the length of the WAV header). When I make a Skype test call to echo123, and it starts playing back my message, it's zero length too - the playback end signal can be heard immediately after the playback start signal. This problem seems to concide with the following error message in the kernel log: hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. Do you have any idea what could be the cause and how to debug this issue further? -- Best Regards, Aleksander Adamowski http://olo.org.pl

2 10

[alsa-devel] [PATCHv5 2/2] ASoC: TWL6040: Add twl6040 codec driver
by Olaya, Margarita 16 Mar '10

16 Mar '10

From: Misael Lopez Cruz <x0052729(a)ti.com> Initial version of TWL6040 codec driver. The TWL6040 codec uses a proprietary PDM-based digital audio interface. Audio paths supported are: - Input: Main Mic, Sub Mic, Headset Mic, Auxiliary-FM Left/Right - Output: Headset Left/Right, Handsfree Left/Right TWL6040 codec supports power-up/down manual and automatic sequence. Manual sequence is done through a specific register writes sequence. Automatic sequence is done when the codec is powered-up through the external AUDPWRON line. The completion of the sequence is signaled through the audio interrupt. TWL6040 codec sysclk can be provided by: low-power or high performance PLL: - The low-power PLL takes a low-frequency input at 32,768 Hz and generates an approximate of 17.64 or 19.2 MHz (for 44.1 KHz and 48 KHz respectively) - The high-performance PLL generates an exact 19.2 MHz clock signal from high-frequency input at 12/19.2/26/38.4 MHz. Low-power playback mode is a special scenario where only headset path (headset DAC and driver) is active. For the particular case of headset path, PLL being used defines the headset power mode: low-power, high-performance. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6040.c | 1223 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 141 +++++ 4 files changed, 1370 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl6040.c create mode 100644 sound/soc/codecs/twl6040.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 16c47ed..398cbb0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,6 +35,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_TWL6040 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM2000 if I2C @@ -168,6 +169,9 @@ config SND_SOC_TWL4030 select TWL4030_CODEC tristate +config SND_SOC_TWL6040 + tristate + config SND_SOC_UDA134X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6981777..98bd10c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,7 @@ 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-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o @@ -85,6 +86,7 @@ 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-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c new file mode 100644 index 0000000..54835cc --- /dev/null +++ b/sound/soc/codecs/twl6040.c @@ -0,0 +1,1223 @@ +/* + * ALSA SoC TWL6040 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * 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/gpio.h> +#include <linux/platform_device.h> +#include <linux/i2c/twl.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 "twl6040.h" + +#define TWL6040_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct twl6040_data { + struct snd_soc_codec codec; + int audpwron; + int naudint; + int codec_powered; + int pll; + int non_lp; + unsigned int sysclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct completion ready; +}; + +/* + * twl6040 register cache & default register settings + */ +static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { + 0x00, /* not used 0x00 */ + 0x4B, /* TWL6040_ASICID (ro) 0x01 */ + 0x00, /* TWL6040_ASICREV (ro) 0x02 */ + 0x00, /* TWL6040_INTID 0x03 */ + 0x00, /* TWL6040_INTMR 0x04 */ + 0x00, /* TWL6040_NCPCTRL 0x05 */ + 0x00, /* TWL6040_LDOCTL 0x06 */ + 0x00, /* TWL6040_HPPLLCTL 0x07 */ + 0x00, /* TWL6040_LPPLLCTL 0x08 */ + 0x00, /* TWL6040_LPPLLDIV 0x09 */ + 0x00, /* TWL6040_AMICBCTL 0x0A */ + 0x00, /* TWL6040_DMICBCTL 0x0B */ + 0x18, /* TWL6040_MICLCTL 0x0C */ + 0x18, /* TWL6040_MICRCTL 0x0D */ + 0x00, /* TWL6040_MICGAIN 0x0E */ + 0x1B, /* TWL6040_LINEGAIN 0x0F */ + 0x00, /* TWL6040_HSLCTL 0x10 */ + 0x00, /* TWL6040_HSRCTL 0x11 */ + 0x00, /* TWL6040_HSGAIN 0x12 */ + 0x06, /* TWL6040_EARCTL 0x13 */ + 0x00, /* TWL6040_HFLCTL 0x14 */ + 0x03, /* TWL6040_HFLGAIN 0x15 */ + 0x00, /* TWL6040_HFRCTL 0x16 */ + 0x03, /* TWL6040_HFRGAIN 0x17 */ + 0x00, /* TWL6040_VIBCTLL 0x18 */ + 0x00, /* TWL6040_VIBDATL 0x19 */ + 0x00, /* TWL6040_VIBCTLR 0x1A */ + 0x00, /* TWL6040_VIBDATR 0x1B */ + 0x00, /* TWL6040_HKCTL1 0x1C */ + 0x00, /* TWL6040_HKCTL2 0x1D */ + 0x00, /* TWL6040_GPOCTL 0x1E */ + 0x00, /* TWL6040_ALB 0x1F */ + 0x00, /* TWL6040_DLB 0x20 */ + 0x00, /* not used 0x21 */ + 0x00, /* not used 0x22 */ + 0x00, /* not used 0x23 */ + 0x00, /* not used 0x24 */ + 0x00, /* not used 0x25 */ + 0x00, /* not used 0x26 */ + 0x00, /* not used 0x27 */ + 0x00, /* TWL6040_TRIM1 0x28 */ + 0x00, /* TWL6040_TRIM2 0x29 */ + 0x00, /* TWL6040_TRIM3 0x2A */ + 0x00, /* TWL6040_HSOTRIM 0x2B */ + 0x00, /* TWL6040_HFOTRIM 0x2C */ + 0x09, /* TWL6040_ACCCTL 0x2D */ + 0x00, /* TWL6040_STATUS (ro) 0x2E */ +}; + +/* + * twl6040 vio/gnd registers: + * registers under vio/gnd supply can be accessed + * before the power-up sequence, after NRESPWRON goes high + */ +static const int twl6040_vio_reg[TWL6040_VIOREGNUM] = { + TWL6040_REG_ASICID, + TWL6040_REG_ASICREV, + TWL6040_REG_INTID, + TWL6040_REG_INTMR, + TWL6040_REG_NCPCTL, + TWL6040_REG_LDOCTL, + TWL6040_REG_AMICBCTL, + TWL6040_REG_DMICBCTL, + TWL6040_REG_HKCTL1, + TWL6040_REG_HKCTL2, + TWL6040_REG_GPOCTL, + TWL6040_REG_TRIM1, + TWL6040_REG_TRIM2, + TWL6040_REG_TRIM3, + TWL6040_REG_HSOTRIM, + TWL6040_REG_HFOTRIM, + TWL6040_REG_ACCCTL, + TWL6040_REG_STATUS, +}; + +/* + * twl6040 vdd/vss registers: + * registers under vdd/vss supplies can only be accessed + * after the power-up sequence + */ +static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = { + TWL6040_REG_HPPLLCTL, + TWL6040_REG_LPPLLCTL, + TWL6040_REG_LPPLLDIV, + TWL6040_REG_MICLCTL, + TWL6040_REG_MICRCTL, + TWL6040_REG_MICGAIN, + TWL6040_REG_LINEGAIN, + TWL6040_REG_HSLCTL, + TWL6040_REG_HSRCTL, + TWL6040_REG_HSGAIN, + TWL6040_REG_EARCTL, + TWL6040_REG_HFLCTL, + TWL6040_REG_HFLGAIN, + TWL6040_REG_HFRCTL, + TWL6040_REG_HFRGAIN, + TWL6040_REG_VIBCTLL, + TWL6040_REG_VIBDATL, + TWL6040_REG_VIBCTLR, + TWL6040_REG_VIBDATR, + TWL6040_REG_ALB, + TWL6040_REG_DLB, +}; + +/* + * read twl6040 register cache + */ +static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + return cache[reg]; +} + +/* + * write twl6040 register cache + */ +static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6040_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * read from twl6040 hardware register + */ +static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &value, reg); + twl6040_write_reg_cache(codec, reg, value); + + return value; +} + +/* + * write to the twl6040 register space + */ +static int twl6040_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + twl6040_write_reg_cache(codec, reg, value); + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); +} + +static void twl6040_init_vio_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + /* allow registers to be accessed by i2c */ + twl6040_write(codec, TWL6040_REG_ACCCTL, cache[TWL6040_REG_ACCCTL]); + + for (i = 0; i < TWL6040_VIOREGNUM; i++) { + reg = twl6040_vio_reg[i]; + /* skip read-only registers (ASICID, ASICREV, STATUS) */ + switch (reg) { + case TWL6040_REG_ASICID: + case TWL6040_REG_ASICREV: + case TWL6040_REG_STATUS: + continue; + default: + break; + } + twl6040_write(codec, reg, cache[reg]); + } +} + +static void twl6040_init_vdd_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + for (i = 0; i < TWL6040_VDDREGNUM; i++) { + reg = twl6040_vdd_reg[i]; + twl6040_write(codec, reg, cache[reg]); + } +} + +/* twl6040 codec manual power-up sequence */ +static void twl6040_power_up(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL); + ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL); + + /* enable reference system */ + ldoctl |= TWL6040_REFENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + msleep(10); + /* enable internal oscillator */ + ldoctl |= TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(10); + /* enable high-side ldo */ + ldoctl |= TWL6040_HSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* enable negative charge pump */ + ncpctl |= TWL6040_NCPENA | TWL6040_NCPOPEN; + twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl); + udelay(488); + /* enable low-side ldo */ + ldoctl |= TWL6040_LSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* enable low-power pll */ + lppllctl |= TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + /* reset state machine */ + accctl |= TWL6040_RESETSPLIT; + twl6040_write(codec, TWL6040_REG_ACCCTL, accctl); + mdelay(5); + accctl &= ~TWL6040_RESETSPLIT; + twl6040_write(codec, TWL6040_REG_ACCCTL, accctl); + /* disable internal oscillator */ + ldoctl &= ~TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); +} + +/* twl6040 codec manual power-down sequence */ +static void twl6040_power_down(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL); + ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL); + + /* enable internal oscillator */ + ldoctl |= TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(10); + /* disable low-power pll */ + lppllctl &= ~TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + /* disable low-side ldo */ + ldoctl &= ~TWL6040_LSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* disable negative charge pump */ + ncpctl &= ~(TWL6040_NCPENA | TWL6040_NCPOPEN); + twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl); + udelay(488); + /* disable high-side ldo */ + ldoctl &= ~TWL6040_HSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* disable internal oscillator */ + ldoctl &= ~TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + /* disable reference system */ + ldoctl &= ~TWL6040_REFENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + msleep(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 = TWL6040_HSDRVMODEL | TWL6040_HSDACMODEL; + + hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + + if (high_perf) { + hslctl &= ~mask; + hsrctl &= ~mask; + } else { + hslctl |= mask; + hsrctl |= mask; + } + + twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl); + twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl); + + return 0; +} + +static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct twl6040_data *priv = codec->private_data; + + if (SND_SOC_DAPM_EVENT_ON(event)) + priv->non_lp++; + else + priv->non_lp--; + + return 0; +} + +/* audio interrupt handler */ +static irqreturn_t twl6040_naudint_handler(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct twl6040_data *priv = codec->private_data; + u8 intid; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); + + switch (intid) { + case TWL6040_THINT: + dev_alert(codec->dev, "die temp over-limit detection\n"); + break; + case TWL6040_PLUGINT: + case TWL6040_UNPLUGINT: + case TWL6040_HOOKINT: + break; + case TWL6040_HFINT: + dev_alert(codec->dev, "hf drivers over current detection\n"); + break; + case TWL6040_VIBINT: + dev_alert(codec->dev, "vib drivers over current detection\n"); + break; + case TWL6040_READYINT: + complete(&priv->ready); + break; + default: + dev_err(codec->dev, "unknown audio interrupt %d\n", intid); + break; + } + + return IRQ_HANDLED; +} + +/* + * MICATT volume control: + * from -6 to 0 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); + +/* + * MICGAIN volume control: + * from 6 to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); + +/* + * HSGAIN volume control: + * from -30 to 0 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0); + +/* + * HFGAIN volume control: + * from -52 to 6 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0); + +/* Left analog microphone selection */ +static const char *twl6040_amicl_texts[] = + {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; + +/* Right analog microphone selection */ +static const char *twl6040_amicr_texts[] = + {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; + +static const struct soc_enum twl6040_enum[] = { + SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 3, twl6040_amicl_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 3, twl6040_amicr_texts), +}; + +static const struct snd_kcontrol_new amicl_control = + SOC_DAPM_ENUM("Route", twl6040_enum[0]); + +static const struct snd_kcontrol_new amicr_control = + SOC_DAPM_ENUM("Route", twl6040_enum[1]); + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new hsdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 5, 1, 0); + +static const struct snd_kcontrol_new hsdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 5, 1, 0); + +/* Handsfree DAC playback switches */ +static const struct snd_kcontrol_new hfdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hfdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0); + +/* Headset driver switches */ +static const struct snd_kcontrol_new hsl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hsr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0); + +/* Handsfree driver switches */ +static const struct snd_kcontrol_new hfl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0); + +static const struct snd_kcontrol_new hfr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0); + +static const struct snd_kcontrol_new twl6040_snd_controls[] = { + /* Capture gains */ + SOC_DOUBLE_TLV("Capture Preamplifier Volume", + TWL6040_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv), + SOC_DOUBLE_TLV("Capture Volume", + TWL6040_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + + /* Playback gains */ + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + +}; + +static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("AFML"), + SND_SOC_DAPM_INPUT("AFMR"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + + /* Analog input muxes for the capture amplifiers */ + SND_SOC_DAPM_MUX("Analog Left Capture Route", + SND_SOC_NOPM, 0, 0, &amicl_control), + SND_SOC_DAPM_MUX("Analog Right Capture Route", + SND_SOC_NOPM, 0, 0, &amicr_control), + + /* Analog capture PGAs */ + SND_SOC_DAPM_PGA("MicAmpL", + TWL6040_REG_MICLCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmpR", + TWL6040_REG_MICRCTL, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", + TWL6040_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", + TWL6040_REG_MICRCTL, 2, 0), + + /* Microphone bias */ + SND_SOC_DAPM_MICBIAS("Headset Mic Bias", + TWL6040_REG_AMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Main Mic Bias", + TWL6040_REG_AMICBCTL, 4, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias", + TWL6040_REG_DMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias", + TWL6040_REG_DMICBCTL, 4, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", + TWL6040_REG_HSLCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", + TWL6040_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", + TWL6040_REG_HFLCTL, 0, 0, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("HFDAC Right", "Handsfree Playback", + TWL6040_REG_HFRCTL, 0, 0, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Analog playback switches */ + SND_SOC_DAPM_SWITCH("HSDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hsdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HSDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hsdacr_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hfdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), + + SND_SOC_DAPM_SWITCH("Headset Left Driver", + SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Headset Right Driver", + SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), + SND_SOC_DAPM_SWITCH_E("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH_E("Handsfree Right Driver", + SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Analog playback PGAs */ + SND_SOC_DAPM_PGA("HFDAC Left PGA", + TWL6040_REG_HFLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HFDAC Right PGA", + TWL6040_REG_HFRCTL, 1, 0, NULL, 0), + +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Capture path */ + {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Left Capture Route", "Main Mic", "MAINMIC"}, + {"Analog Left Capture Route", "Aux/FM Left", "AFML"}, + + {"Analog Right Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Right Capture Route", "Sub Mic", "SUBMIC"}, + {"Analog Right Capture Route", "Aux/FM Right", "AFMR"}, + + {"MicAmpL", NULL, "Analog Left Capture Route"}, + {"MicAmpR", NULL, "Analog Right Capture Route"}, + + {"ADC Left", NULL, "MicAmpL"}, + {"ADC Right", NULL, "MicAmpR"}, + + /* Headset playback path */ + {"HSDAC Left Playback", "Switch", "HSDAC Left"}, + {"HSDAC Right Playback", "Switch", "HSDAC Right"}, + + {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, + {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + + {"HSOL", NULL, "Headset Left Driver"}, + {"HSOR", NULL, "Headset Right Driver"}, + + /* Handsfree playback path */ + {"HFDAC Left Playback", "Switch", "HFDAC Left"}, + {"HFDAC Right Playback", "Switch", "HFDAC Right"}, + + {"HFDAC Left PGA", NULL, "HFDAC Left Playback"}, + {"HFDAC Right PGA", NULL, "HFDAC Right Playback"}, + + {"Handsfree Left Driver", "Switch", "HFDAC Left PGA"}, + {"Handsfree Right Driver", "Switch", "HFDAC Right PGA"}, + + {"HFL", NULL, "Handsfree Left Driver"}, + {"HFR", NULL, "Handsfree Right Driver"}, +}; + +static int twl6040_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl6040_dapm_widgets, + ARRAY_SIZE(twl6040_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int twl6040_power_up_completion(struct snd_soc_codec *codec, + int naudint) +{ + struct twl6040_data *priv = codec->private_data; + int time_left; + u8 intid; + + time_left = wait_for_completion_timeout(&priv->ready, + msecs_to_jiffies(48)); + + if (!time_left) { + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, + TWL6040_REG_INTID); + if (!(intid & TWL6040_READYINT)) { + dev_err(codec->dev, "timeout waiting for READYINT\n"); + return -ETIMEDOUT; + } + } + + priv->codec_powered = 1; + + return 0; +} + +static int twl6040_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct twl6040_data *priv = codec->private_data; + int audpwron = priv->audpwron; + int naudint = priv->naudint; + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 1); + + /* wait for power-up completion */ + ret = twl6040_power_up_completion(codec, naudint); + if (ret) + return ret; + + /* sync registers updated during power-up sequence */ + twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LPPLLCTL); + } else { + /* use manual power-up sequence */ + twl6040_power_up(codec); + priv->codec_powered = 1; + } + + /* initialize vdd/vss registers with reg_cache */ + twl6040_init_vdd_regs(codec); + break; + case SND_SOC_BIAS_OFF: + if (!priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 0); + + /* power-down sequence latency */ + udelay(500); + + /* sync registers updated during power-down sequence */ + twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL); + twl6040_write_reg_cache(codec, TWL6040_REG_LPPLLCTL, + 0x00); + } else { + /* use manual power-down sequence */ + twl6040_power_down(codec); + } + + priv->codec_powered = 0; + break; + } + + codec->bias_level = level; + + return 0; +} + +/* set of rates for each pll: low-power and high-performance */ + +static unsigned int lp_rates[] = { + 88200, + 96000, +}; + +static struct snd_pcm_hw_constraint_list lp_constraints = { + .count = ARRAY_SIZE(lp_rates), + .list = lp_rates, +}; + +static unsigned int hp_rates[] = { + 96000, +}; + +static struct snd_pcm_hw_constraint_list hp_constraints = { + .count = ARRAY_SIZE(hp_rates), + .list = hp_rates, +}; + +static int twl6040_startup(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 twl6040_data *priv = codec->private_data; + + if (!priv->sysclk) { + dev_err(codec->dev, + "no mclk configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + /* + * capture is not supported at 17.64 MHz, + * it's reserved for headset low-power playback scenario + */ + if ((priv->sysclk == 17640000) && substream->stream) { + dev_err(codec->dev, + "capture mode is not supported at %dHz\n", + priv->sysclk); + return -EINVAL; + } + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + priv->sysclk_constraints); + + return 0; +} + +static int twl6040_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; + struct twl6040_data *priv = codec->private_data; + u8 lppllctl; + int rate; + + /* nothing to do for high-perf pll, it supports only 48 kHz */ + if (priv->pll == TWL6040_HPPLL_ID) + return 0; + + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + + rate = params_rate(params); + switch (rate) { + case 88200: + lppllctl |= TWL6040_LPLLFIN; + priv->sysclk = 17640000; + break; + case 96000: + lppllctl &= ~TWL6040_LPLLFIN; + priv->sysclk = 19200000; + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", rate); + return -EINVAL; + } + + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + return 0; +} + +static int twl6040_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 twl6040_data *priv = codec->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * low-power playback mode is restricted + * for headset path only + */ + if ((priv->sysclk == 17640000) && priv->non_lp) { + dev_err(codec->dev, + "some enabled paths aren't supported at %dHz\n", + priv->sysclk); + return -EPERM; + } + break; + default: + break; + } + + return 0; +} + +static int twl6040_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 twl6040_data *priv = codec->private_data; + u8 hppllctl, lppllctl; + + hppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_HPPLLCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + + switch (clk_id) { + case TWL6040_SYSCLK_SEL_LPPLL: + switch (freq) { + case 32768: + /* headset dac and driver must be in low-power mode */ + headset_power_mode(codec, 0); + + /* clk32k input requires low-power pll */ + lppllctl |= TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + mdelay(5); + lppllctl &= ~TWL6040_HPLLSEL; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + hppllctl &= ~TWL6040_HPLLENA; + twl6040_write(codec, TWL6040_REG_HPPLLCTL, hppllctl); + break; + default: + dev_err(codec->dev, "unknown mclk freq %d\n", freq); + return -EINVAL; + } + + /* lppll divider */ + switch (priv->sysclk) { + case 17640000: + lppllctl |= TWL6040_LPLLFIN; + break; + case 19200000: + lppllctl &= ~TWL6040_LPLLFIN; + break; + default: + /* sysclk not yet configured */ + lppllctl &= ~TWL6040_LPLLFIN; + priv->sysclk = 19200000; + break; + } + + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + priv->pll = TWL6040_LPPLL_ID; + priv->sysclk_constraints = &lp_constraints; + break; + case TWL6040_SYSCLK_SEL_HPPLL: + hppllctl &= ~TWL6040_MCLK_MSK; + + switch (freq) { + case 12000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6040_MCLK_12000KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLENA; + break; + case 19200000: + /* mclk input, pll disabled */ + hppllctl |= TWL6040_MCLK_19200KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLBP; + break; + case 26000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6040_MCLK_26000KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLENA; + break; + case 38400000: + /* clk slicer, pll disabled */ + hppllctl |= TWL6040_MCLK_38400KHZ | + TWL6040_HPLLSQRENA | + TWL6040_HPLLBP; + break; + default: + dev_err(codec->dev, "unknown mclk freq %d\n", freq); + return -EINVAL; + } + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + twl6040_write(codec, TWL6040_REG_HPPLLCTL, hppllctl); + udelay(500); + lppllctl |= TWL6040_HPLLSEL; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + lppllctl &= ~TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + /* high-performance pll can provide only 19.2 MHz */ + priv->pll = TWL6040_HPPLL_ID; + priv->sysclk = 19200000; + priv->sysclk_constraints = &hp_constraints; + break; + default: + dev_err(codec->dev, "unknown clk_id %d\n", clk_id); + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops twl6040_dai_ops = { + .startup = twl6040_startup, + .hw_params = twl6040_hw_params, + .trigger = twl6040_trigger, + .set_sysclk = twl6040_set_dai_sysclk, +}; + +struct snd_soc_dai twl6040_dai = { + .name = "twl6040", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}; +EXPORT_SYMBOL_GPL(twl6040_dai); + +static int twl6040_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; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl6040_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6040_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *twl6040_codec; + +static int twl6040_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + BUG_ON(!twl6040_codec); + + codec = twl6040_codec; + socdev->card->codec = codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; + } + + snd_soc_add_controls(codec, twl6040_snd_controls, + ARRAY_SIZE(twl6040_snd_controls)); + twl6040_add_widgets(codec); + + if (ret < 0) { + dev_err(&pdev->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + return ret; +} + +static int twl6040_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_twl6040 = { + .probe = twl6040_probe, + .remove = twl6040_remove, + .suspend = twl6040_suspend, + .resume = twl6040_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl6040); + +static int __devinit twl6040_codec_probe(struct platform_device *pdev) +{ + struct twl4030_codec_data *twl_codec = pdev->dev.platform_data; + struct snd_soc_codec *codec; + struct twl6040_data *priv; + int audpwron, naudint; + int ret = 0; + + priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + if (twl_codec) { + audpwron = twl_codec->audpwron_gpio; + naudint = twl_codec->naudint_irq; + } else { + audpwron = -EINVAL; + naudint = 0; + } + + priv->audpwron = audpwron; + priv->naudint = naudint; + + codec = &priv->codec; + codec->dev = &pdev->dev; + twl6040_dai.dev = &pdev->dev; + + codec->name = "twl6040"; + codec->owner = THIS_MODULE; + codec->read = twl6040_read_reg_cache; + codec->write = twl6040_write; + codec->set_bias_level = twl6040_set_bias_level; + codec->private_data = priv; + codec->dai = &twl6040_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(twl6040_reg); + codec->reg_cache = kmemdup(twl6040_reg, sizeof(twl6040_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + init_completion(&priv->ready); + + if (gpio_is_valid(audpwron)) { + ret = gpio_request(audpwron, "audpwron"); + if (ret) + goto gpio1_err; + + ret = gpio_direction_output(audpwron, 0); + if (ret) + goto gpio2_err; + + priv->codec_powered = 0; + } + + if (naudint) { + /* audio interrupt */ + ret = request_threaded_irq(naudint, NULL, + twl6040_naudint_handler, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "twl6040_codec", codec); + if (ret) + goto gpio2_err; + } else { + if (gpio_is_valid(audpwron)) { + /* enable only codec ready interrupt */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + ~TWL6040_READYMSK & TWL6040_ALLINT_MSK); + } else { + /* no interrupts at all */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + TWL6040_ALLINT_MSK); + } + } + + /* init vio registers */ + twl6040_init_vio_regs(codec); + + /* power on device */ + ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto irq_err; + + ret = snd_soc_register_codec(codec); + if (ret) + goto reg_err; + + twl6040_codec = codec; + + ret = snd_soc_register_dai(&twl6040_dai); + if (ret) + goto dai_err; + + return 0; + +dai_err: + snd_soc_unregister_codec(codec); + twl6040_codec = NULL; +reg_err: + twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); +irq_err: + if (naudint) + free_irq(naudint, codec); +gpio2_err: + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); +gpio1_err: + kfree(codec->reg_cache); +cache_err: + kfree(priv); + return ret; +} + +static int __devexit twl6040_codec_remove(struct platform_device *pdev) +{ + struct twl6040_data *priv = twl6040_codec->private_data; + int audpwron = priv->audpwron; + int naudint = priv->naudint; + + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); + + if (naudint) + free_irq(naudint, twl6040_codec); + + snd_soc_unregister_dai(&twl6040_dai); + snd_soc_unregister_codec(twl6040_codec); + + kfree(twl6040_codec); + twl6040_codec = NULL; + + return 0; +} + +static struct platform_driver twl6040_codec_driver = { + .driver = { + .name = "twl6040_codec", + .owner = THIS_MODULE, + }, + .probe = twl6040_codec_probe, + .remove = __devexit_p(twl6040_codec_remove), +}; + +static int __init twl6040_codec_init(void) +{ + return platform_driver_register(&twl6040_codec_driver); +} +module_init(twl6040_codec_init); + +static void __exit twl6040_codec_exit(void) +{ + platform_driver_unregister(&twl6040_codec_driver); +} +module_exit(twl6040_codec_exit); + +MODULE_DESCRIPTION("ASoC TWL6040 codec driver"); +MODULE_AUTHOR("Misael Lopez Cruz"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h new file mode 100644 index 0000000..c472070 --- /dev/null +++ b/sound/soc/codecs/twl6040.h @@ -0,0 +1,141 @@ +/* + * ALSA SoC TWL6040 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * 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 __TWL6040_H__ +#define __TWL6040_H__ + +#define TWL6040_REG_ASICID 0x01 +#define TWL6040_REG_ASICREV 0x02 +#define TWL6040_REG_INTID 0x03 +#define TWL6040_REG_INTMR 0x04 +#define TWL6040_REG_NCPCTL 0x05 +#define TWL6040_REG_LDOCTL 0x06 +#define TWL6040_REG_HPPLLCTL 0x07 +#define TWL6040_REG_LPPLLCTL 0x08 +#define TWL6040_REG_LPPLLDIV 0x09 +#define TWL6040_REG_AMICBCTL 0x0A +#define TWL6040_REG_DMICBCTL 0x0B +#define TWL6040_REG_MICLCTL 0x0C +#define TWL6040_REG_MICRCTL 0x0D +#define TWL6040_REG_MICGAIN 0x0E +#define TWL6040_REG_LINEGAIN 0x0F +#define TWL6040_REG_HSLCTL 0x10 +#define TWL6040_REG_HSRCTL 0x11 +#define TWL6040_REG_HSGAIN 0x12 +#define TWL6040_REG_EARCTL 0x13 +#define TWL6040_REG_HFLCTL 0x14 +#define TWL6040_REG_HFLGAIN 0x15 +#define TWL6040_REG_HFRCTL 0x16 +#define TWL6040_REG_HFRGAIN 0x17 +#define TWL6040_REG_VIBCTLL 0x18 +#define TWL6040_REG_VIBDATL 0x19 +#define TWL6040_REG_VIBCTLR 0x1A +#define TWL6040_REG_VIBDATR 0x1B +#define TWL6040_REG_HKCTL1 0x1C +#define TWL6040_REG_HKCTL2 0x1D +#define TWL6040_REG_GPOCTL 0x1E +#define TWL6040_REG_ALB 0x1F +#define TWL6040_REG_DLB 0x20 +#define TWL6040_REG_TRIM1 0x28 +#define TWL6040_REG_TRIM2 0x29 +#define TWL6040_REG_TRIM3 0x2A +#define TWL6040_REG_HSOTRIM 0x2B +#define TWL6040_REG_HFOTRIM 0x2C +#define TWL6040_REG_ACCCTL 0x2D +#define TWL6040_REG_STATUS 0x2E + +#define TWL6040_CACHEREGNUM (TWL6040_REG_STATUS + 1) + +#define TWL6040_VIOREGNUM 18 +#define TWL6040_VDDREGNUM 21 + +/* INTID (0x03) fields */ + +#define TWL6040_THINT 0x01 +#define TWL6040_PLUGINT 0x02 +#define TWL6040_UNPLUGINT 0x04 +#define TWL6040_HOOKINT 0x08 +#define TWL6040_HFINT 0x10 +#define TWL6040_VIBINT 0x20 +#define TWL6040_READYINT 0x40 + +/* INTMR (0x04) fields */ + +#define TWL6040_READYMSK 0x40 +#define TWL6040_ALLINT_MSK 0x7B + +/* NCPCTL (0x05) fields */ + +#define TWL6040_NCPENA 0x01 +#define TWL6040_NCPOPEN 0x40 + +/* LDOCTL (0x06) fields */ + +#define TWL6040_LSLDOENA 0x01 +#define TWL6040_HSLDOENA 0x04 +#define TWL6040_REFENA 0x40 +#define TWL6040_OSCENA 0x80 + +/* HPPLLCTL (0x07) fields */ + +#define TWL6040_HPLLENA 0x01 +#define TWL6040_HPLLRST 0x02 +#define TWL6040_HPLLBP 0x04 +#define TWL6040_HPLLSQRENA 0x08 +#define TWL6040_HPLLSQRBP 0x10 +#define TWL6040_MCLK_12000KHZ (0 << 5) +#define TWL6040_MCLK_19200KHZ (1 << 5) +#define TWL6040_MCLK_26000KHZ (2 << 5) +#define TWL6040_MCLK_38400KHZ (3 << 5) +#define TWL6040_MCLK_MSK 0x60 + +/* LPPLLCTL (0x08) fields */ + +#define TWL6040_LPLLENA 0x01 +#define TWL6040_LPLLRST 0x02 +#define TWL6040_LPLLSEL 0x04 +#define TWL6040_LPLLFIN 0x08 +#define TWL6040_HPLLSEL 0x10 + +/* HSLCTL (0x10) fields */ + +#define TWL6040_HSDACMODEL 0x02 +#define TWL6040_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6040_HSDACMODER 0x02 +#define TWL6040_HSDRVMODER 0x08 + +/* ACCCTL (0x2D) fields */ + +#define TWL6040_RESETSPLIT 0x04 + +#define TWL6040_SYSCLK_SEL_LPPLL 1 +#define TWL6040_SYSCLK_SEL_HPPLL 2 + +#define TWL6040_HPPLL_ID 1 +#define TWL6040_LPPLL_ID 2 + +extern struct snd_soc_dai twl6040_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl6040; + +#endif /* End of __TWL6040_H__ */ -- 1.6.1.3

3 3

[alsa-devel] [PATCH 1/2] OMAP4: PMIC: Rename twl6030_codec as twl6040_codec
by Olaya, Margarita 16 Mar '10

16 Mar '10

Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- drivers/mfd/twl-core.c | 4 ++-- include/linux/i2c/twl.h | 6 +++--- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/mfd/twl-core.c b/drivers/mfd/twl-core.c index 562cd49..720e099 100644 --- a/drivers/mfd/twl-core.c +++ b/drivers/mfd/twl-core.c @@ -109,7 +109,7 @@ #endif #if defined(CONFIG_TWL4030_CODEC) || defined(CONFIG_TWL4030_CODEC_MODULE) ||\ - defined(CONFIG_SND_SOC_TWL6030) || defined(CONFIG_SND_SOC_TWL6030_MODULE) + defined(CONFIG_SND_SOC_TWL6040) || defined(CONFIG_SND_SOC_TWL6040_MODULE) #define twl_has_codec() true #else #define twl_has_codec() false @@ -708,7 +708,7 @@ add_children(struct twl4030_platform_data *pdata, unsigned long features) /* Phoenix*/ if (twl_has_codec() && pdata->codec && twl_class_is_6030()) { sub_chip_id = twl_map[TWL_MODULE_AUDIO_VOICE].sid; - child = add_child(sub_chip_id, "twl6030_codec", + child = add_child(sub_chip_id, "twl6040_codec", pdata->codec, sizeof(*pdata->codec), false, 0, 0); if (IS_ERR(child)) diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h index fb6784e..ebd90ce 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -569,9 +569,9 @@ struct twl4030_codec_data { struct twl4030_codec_audio_data *audio; struct twl4030_codec_vibra_data *vibra; - /* twl6030 */ - int audpwron_gpio; /* audio power-on gpio */ - int naudint_irq; /* audio interrupt */ + /* twl6040 */ + int audpwron_gpio; /* audio power-on gpio */ + int naudint_irq; /* audio interrupt */ }; struct twl4030_platform_data { -- 1.6.1.3

2 2

Re: [alsa-devel] Configuration help please?
by Alex Austin 16 Mar '10

16 Mar '10

That is a very complex thing you're asking for. 5.1 sound works by having mythtv pass the raw datastream out the spdif port. Stereo analog is decoded by mythtv, so to handle both, you would need mythtv to output the raw stream to spdif at the same time as it decodes to stereo on analog. You could have the stream split to two outputs, but the spdif would always be PCM stereo. On Mar 16, 2010 9:32 AM, "Dan Armbrust" <daniel.armbrust.list(a)gmail.com> wrote: On Mon, Mar 15, 2010 at 9:11 AM, Dan Armbrust <daniel.armbrust.list(a)gmail.com> wrote: > Hi, > > I'm ... Hello? Is this really a hard thing that I'm asking to do? No one can help me configure alsa? Thanks, Dan _______________________________________________ Alsa-devel mailing list Alsa-devel@als...

2 1

[alsa-devel] Configuration help please?
by Dan Armbrust 16 Mar '10

16 Mar '10

Hi, I'm trying to configure a mythtv box to send sound to my spdif port AND my analog stereo speakers. In order to make the spdif port work, I found I needed this in /etc/asound.conf: #---- pcm.!default { type plug slave.pcm "spdif" } #---- I thought maybe this would work to send the sound to both.... #---- pcm.!default { type plug slave.pcm "front" slave.pcm "spdif" } #---- But that was a little naive... apparently. The sound just goes to whichever item I put last. I know right next to nothing about writing alsa config files... and every example I find is very large, very complicated looking, and doesn't do what I want. Can someone tell me how to configure alsa for this? What I want - exactly - Sound to always go to both the "front" (my 2 channel stereo speakers), and my spdif output, with anywhere between 1 and 5.1 channels... depending on what my source audio has. I also want mythtv to pass the full 5.1 channels from HD to the spdif port, while sending the stereo sound to the analog. Some info: # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC888 Analog [ALC888 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC888 Digital [ALC888 Digital] Subdevices: 0/1 Subdevice #0: subdevice #0 #aplay -L front:CARD=Intel,DEV=0 HDA Intel, ALC888 Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, ALC888 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, ALC888 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, ALC888 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, ALC888 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, ALC888 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Intel,DEV=0 HDA Intel, ALC888 Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) Thanks! Dan

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Re: [alsa-devel] underrun problem when short audio play
by ANISH KUMAR 16 Mar '10

16 Mar '10

2 1

Re: [alsa-devel] underrun problem when short audio play
by ANISH KUMAR 16 Mar '10

16 Mar '10

>Are you sure that the >drivers you are using implement suspend and resume at all? I am using s3c6410 code.I2S driver has suspend/resume call(jassi brar maintained). Audio codec is also having generic code for suspend/resume but don't know if that is the problem or not.However how can i verify if my codec suspend/resume has got some problem?Most likely my codec has got some configuration problems but i am not getting any pointers to debug it.Any pointers in this direction would be very helpful.

1 0

Re: [alsa-devel] underrun problem when short audio play
by ANISH KUMAR 16 Mar '10

16 Mar '10

Thanks Mark, >If your driver is not responding promptly to requests to terminate playback then an underrun doesn't seem surprising. Sometimes in my driver write calls doesn't return causing application to block forever and CPU pegged problem. Situation:Recording going on->go to sleep->come back from sleep->start recording again->try recording and playing back simultaneously(something like recording going on in background and you try to play short audio). This causes write call to get blocked in playback case causing application to hangup. This doesn't happen if we don't go to sleep. So i think somewhere there is some problem with suspend->resume case which is difficult to figure out. I checked driver logs after coming back from sleep.I found for some time interrupts are coming then stopped coming suddenly So now after waking up i am reopening the playback device before opening the record device.Which has fixed the problem but not the root cause.

2 1

[alsa-devel] underrun problem when short audio play
by ANISH KUMAR 16 Mar '10

16 Mar '10

Hi all, I am facing a problem when i try to play short duration audio(problem is not observerd when i am playing tones which is also short duration audio). Somehow i am getting buffer overrun everytime i am playing short duration audio. It is happening because of avail >= stop_threshold. When i tried to increase the stop_threshold but then i got "same audio played again kind of echo" problem. I can't try the solution suggested by jaroslav i.e. checking for DRAINING state in lower level driver and stop queuing because of some design issue as in my code i don't call drain function as soon as all buffers are written. I wait for some time(in miliseconds) and then call drain function. Please suggest any pointers.Should i increase the buffer size or stop_threshold both of which i tried but to no use. I am always getting underrun problem that too always in the function update_hw_ptr_post function. How can i disable the stop_threshold check?? Thanks, anish

2 1

[alsa-devel] [PATCH 2/4] mmp: support ssp in pxa168
by Haojian Zhuang 16 Mar '10

16 Mar '10

4 5

Re: [alsa-devel] [PATCH 0/4] Some typo fixing
by Jiri Kosina 16 Mar '10

16 Mar '10

On Mon, 15 Mar 2010, Randy Dunlap wrote: > > I have fixed some typos. > > Acked-by: Randy Dunlap <rdunlap(a)xenotime.net> > > Jiri, can you merge these, please, unless someone objects (?). Yes, I will take it, thanks. A couple comments though: - [important!] Thomas, it's not necessary to CC zillions of people on such patches. Just submit them to trivial(a)kernel.org (and eventually CC lkml), and that's it. I believe many people might get annoyed by this. - I will not be applying the drivers/staging hunks. Staging patches are moving target, the code is changing quickly (including complete drops/rewrites) so we'll likely be getting conflicts there. I will reroute them to Greg. - I will fold all the patches into one. We don't need one commit per one specific typo. -- Jiri Kosina SUSE Labs, Novell Inc.

1 0

[alsa-devel] [patch] sequencer: clean up remove bogus check
by Dan Carpenter 16 Mar '10

16 Mar '10

A few lines earlier bend is limited to 2399. So semitones is always less than 24 here. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index c798746..e85789e 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -1631,8 +1631,6 @@ unsigned long compute_finetune(unsigned long base_freq, int bend, int range, } semitones = bend / 100; - if (semitones > 99) - semitones = 99; cents = bend % 100; amount = (int) (semitone_tuning[semitones] * multiplier * cent_tuning[cents]) / 10000;

2 1

[alsa-devel] [PATCH] ALSA: hda: Use LPIB and 6stack-dig for eMachines T5212
by Daniel Chen 16 Mar '10

16 Mar '10

Hi, Attached is a patch against current sound-2.6 master HEAD (applicable also to 2.6.32.10 and 2.6.33.1) that corrects the model quirk for the OR's machine and adds the LPIB quirk. Best, -Dan

2 1

[alsa-devel] [PATCH 0/4] Some typo fixing
by Thomas Weber 15 Mar '10

15 Mar '10

1 0

[alsa-devel] ALSA not working on HP Touchsmart 600-1100 with CentOS 5.4
by Robert Krakora 15 Mar '10

15 Mar '10

Hello, I have tried installing ALSA driver, lib and util for every version from 1.21 to the tip and I cannot get alsaconf to identify any audio device on my HP Touchsmart 600-1100 with CentOS 5.4 installed. Please see the attached lspci and lshw dumps. Thanks in advance. Best Regards, -- Rob Krakora Senior Software Engineer MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext. 206 (317)663-0808 Fax

1 0

[Patch v2] aconnect -x: Do not update index after removal of connection. Signed-off-by: lieven moors <lievenmoors@gmail.com>
by lieven moors 15 Mar '10

15 Mar '10

--- seq/aconnect/aconnect.c | 43 ++++++++++++------------------------------- 1 files changed, 12 insertions(+), 31 deletions(-) diff --git a/seq/aconnect/aconnect.c b/seq/aconnect/aconnect.c index 1a50666..6092748 100644 --- a/seq/aconnect/aconnect.c +++ b/seq/aconnect/aconnect.c @@ -192,52 +192,33 @@ static void remove_connection(snd_seq_t *seq, snd_seq_client_info_t *cinfo, snd_seq_port_info_t *pinfo, int count) { snd_seq_query_subscribe_t *query; - snd_seq_query_subscribe_alloca(&query); snd_seq_query_subscribe_set_root(query, snd_seq_port_info_get_addr(pinfo)); - snd_seq_query_subscribe_set_type(query, SND_SEQ_QUERY_SUBS_READ); snd_seq_query_subscribe_set_index(query, 0); - for (; snd_seq_query_port_subscribers(seq, query) >= 0; - snd_seq_query_subscribe_set_index(query, snd_seq_query_subscribe_get_index(query) + 1)) { + + while (snd_seq_query_port_subscribers(seq, query) >= 0) + { snd_seq_port_info_t *port; snd_seq_port_subscribe_t *subs; const snd_seq_addr_t *sender = snd_seq_query_subscribe_get_root(query); const snd_seq_addr_t *dest = snd_seq_query_subscribe_get_addr(query); snd_seq_port_info_alloca(&port); - if (snd_seq_get_any_port_info(seq, dest->client, dest->port, port) < 0) - continue; - if (!(snd_seq_port_info_get_capability(port) & SND_SEQ_PORT_CAP_SUBS_WRITE)) - continue; - if (snd_seq_port_info_get_capability(port) & SND_SEQ_PORT_CAP_NO_EXPORT) - continue; - snd_seq_port_subscribe_alloca(&subs); - snd_seq_port_subscribe_set_queue(subs, snd_seq_query_subscribe_get_queue(query)); - snd_seq_port_subscribe_set_sender(subs, sender); - snd_seq_port_subscribe_set_dest(subs, dest); - snd_seq_unsubscribe_port(seq, subs); - } - snd_seq_query_subscribe_set_type(query, SND_SEQ_QUERY_SUBS_WRITE); - snd_seq_query_subscribe_set_index(query, 0); - for (; snd_seq_query_port_subscribers(seq, query) >= 0; - snd_seq_query_subscribe_set_index(query, snd_seq_query_subscribe_get_index(query) + 1)) { - snd_seq_port_info_t *port; - snd_seq_port_subscribe_t *subs; - const snd_seq_addr_t *dest = snd_seq_query_subscribe_get_root(query); - const snd_seq_addr_t *sender = snd_seq_query_subscribe_get_addr(query); - snd_seq_port_info_alloca(&port); - if (snd_seq_get_any_port_info(seq, sender->client, sender->port, port) < 0) - continue; - if (!(snd_seq_port_info_get_capability(port) & SND_SEQ_PORT_CAP_SUBS_READ)) - continue; - if (snd_seq_port_info_get_capability(port) & SND_SEQ_PORT_CAP_NO_EXPORT) + if ((snd_seq_get_any_port_info(seq, dest->client, dest->port, port) < 0) || + !(snd_seq_port_info_get_capability(port) & SND_SEQ_PORT_CAP_SUBS_WRITE) || + (snd_seq_port_info_get_capability(port) & SND_SEQ_PORT_CAP_NO_EXPORT)) + { + snd_seq_query_subscribe_set_index(query, snd_seq_query_subscribe_get_index(query) + 1); continue; + } snd_seq_port_subscribe_alloca(&subs); snd_seq_port_subscribe_set_queue(subs, snd_seq_query_subscribe_get_queue(query)); snd_seq_port_subscribe_set_sender(subs, sender); snd_seq_port_subscribe_set_dest(subs, dest); - snd_seq_unsubscribe_port(seq, subs); + if(snd_seq_unsubscribe_port(seq, subs) < 0){ + snd_seq_query_subscribe_set_index(query, snd_seq_query_subscribe_get_index(query) + 1); + } } } -- 1.6.3.3 --------------020302040200040109060709--

1 0

[alsa-devel] Driver for a codec with built-in DSP for MP3 playback
by Dajun Chen 15 Mar '10

15 Mar '10

Hi all, I am wondering where to find the guidances of developing a device driver for a codec device which has a built-in DSP for MP3 (and tracks in other format) playback and recording? Regards David

2 5

[alsa-devel] [PATCH] ALSA: hda - Add PCI quirks for MSI all-in-one computers NetOn AP1900 and Wind Top AE2220
by Anisse Astier 15 Mar '10

15 Mar '10

Signed-off-by: Anisse Astier <anisse(a)astier.eu> --- Hi, This should make the speakers and jack detection work on those computers. Regards, Anisse sound/pci/hda/patch_realtek.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e8cbe21..970d0b2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9191,6 +9191,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG), + SND_PCI_QUIRK(0x1462, 0x4570, "MSI Wind Top AE2220", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x6510, "MSI GX620", ALC883_TARGA_8ch_DIG), SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), @@ -9200,6 +9201,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7437, "MSI NetOn AP1900", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG), -- 1.6.5.7

2 4

Re: [alsa-devel] [2.6.34-rc1 REGRESSION] alsa: no mixer controls (and no sound) with 2.6.24-rc1+ on Realtek ALC260
by Takashi Iwai 15 Mar '10

15 Mar '10

At Mon, 15 Mar 2010 13:41:30 +0100, Alessandro Guido wrote: > > 2010/3/15 Takashi Iwai <tiwai(a)suse.de> > > > At Mon, 15 Mar 2010 01:06:17 +0100, > > Alessandro Guido wrote: > > > > > > With 2.6.34-rc1+ my laptop (Sony Vaio VGN-FS215S) can't reproduce any > > > sound. All controls are missing from alsamixer and this message is found > > > in dmesg: > > > > > > hda_codec: cannot build controlsfor #0 (error -22) > > > > > > Kernel config is attached, and the output from alsa-info.sh follows: > > > > Does the patch below fix? > > > > Yes, it does. Thanks for checking. The fix is merged in sound tree. Takashi

1 0

[alsa-devel] [PATCH 2/2] Quick fix for the audio clocks on SAM9-L9261
by Benjamin Valentin 15 Mar '10

15 Mar '10

Signed-off-by: Dimitar Dimitrov <dimitar(a)olimex.com> From 10a85229b2357c89d5170efde5ac0e1cba4c9f9e Mon Sep 17 00:00:00 2001 From: Dimitar Dimitrov <dimitar(a)olimex.com> Date: Wed, 27 May 2009 08:11:37 +0300 Subject: [PATCH] Quick fix for the audio clocks on SAM9-L9261 This patch together with support from SAM9-L9261 hardware fix an issue with audio clocks. AT73C213 needs MCLK and BCLK with equal phase. The MCLK from AT91SAM9XXX clock output and BCLK from the SSC TK clock output are not phase-synchronized. Thus the SAM9-L9261 board provides external hardware divison by 8 from MCLK to BCLK, ensuring zero phase difference. See: http://www.at91.com/phpbb/viewtopic.php?t=2633 --- sound/spi/at73c213.c | 54 ++++++++++++++++++++++++++++++++++++++----------- 1 files changed, 42 insertions(+), 12 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 4c7b051..325e534 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -29,6 +29,8 @@ #include <linux/atmel-ssc.h> +#include <asm/mach-types.h> + #include <linux/spi/spi.h> #include <linux/spi/at73c213.h> @@ -114,7 +116,11 @@ snd_at73c213_write_reg(struct snd_at73c213 *chip, u8 reg, u8 val) static struct snd_pcm_hardware snd_at73c213_playback_hw = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, +#ifdef __BIG_ENDIAN .formats = SNDRV_PCM_FMTBIT_S16_BE, +#else + .formats = SNDRV_PCM_FMTBIT_S16_LE, +#endif .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, /* Replaced by chip->bitrate later. */ .rate_max = 50000, /* Replaced by chip->bitrate later. */ @@ -144,7 +150,8 @@ static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip) * as practical to the desired target rate. * * The DAC master clock (MCLK) is programmable, and is either 256 - * or (not here) 384 times the I2S output clock (BCLK). + * or (not here) 384 times the I2S frame clock (LRC), that is + * 8 or 12 times the output bit clock (BCLK). */ /* SSC clock / (bitrate * stereo * 16-bit). */ @@ -747,17 +754,40 @@ cleanup: */ static int __devinit snd_at73c213_ssc_init(struct snd_at73c213 *chip) { - /* - * Continuous clock output. - * Starts on falling TF. - * Delay 1 cycle (1 bit). - * Periode is 16 bit (16 - 1). - */ - ssc_writel(chip->ssc->regs, TCMR, - SSC_BF(TCMR_CKO, 1) - | SSC_BF(TCMR_START, 4) - | SSC_BF(TCMR_STTDLY, 1) - | SSC_BF(TCMR_PERIOD, 16 - 1)); + if (machine_is_sam9_l9261()) { + /* + * dimitar(a)olimex.com : AT73C213 needs MCLK and BCLK with equal + * phase. The MCLK from AT91SAM9XXX clock output and BCLK from + * the SSC TK clock output are not phase-synchronized. Thus + * the SAM9-L9261 board provides external hardware divison by + * 8 from MCLK to BCLK, ensuring zero phase difference. + * See: http://www.at91.com/phpbb/viewtopic.php?t=2633 + * + * TK is clock input. + * Clock source is TK pin. + * Starts on falling TF. + * Delay 1 cycle (1 bit). + * Period is 16 bit (16 - 1). + */ + ssc_writel(chip->ssc->regs, TCMR, + SSC_BF(TCMR_CKS, 2) + | SSC_BF(TCMR_CKO, 0) + | SSC_BF(TCMR_START, 4) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_PERIOD, 16 - 1)); + } else { + /* + * Continuous clock output. + * Starts on falling TF. + * Delay 1 cycle (1 bit). + * Period is 16 bit (16 - 1). + */ + ssc_writel(chip->ssc->regs, TCMR, + SSC_BF(TCMR_CKO, 1) + | SSC_BF(TCMR_START, 4) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_PERIOD, 16 - 1)); + } /* * Data length is 16 bit (16 - 1). * Transmit MSB first. -- 1.5.6.1

3 2

[alsa-devel] 2.6.34-rc1: regression: ^Z no longer stops sound
by Pavel Machek 15 Mar '10

15 Mar '10

Hi! In 2.6.33, hitting ^Z stopped mpg123 in a way you'd expect (both on thinkpad and geode notebook). On 2.6.34-rc1, hit ^Z and sound starts looping. Pavel -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

3 3

Re: [alsa-devel] [2.6.34-rc1 REGRESSION] alsa: no mixer controls (and no sound) with 2.6.24-rc1+ on Realtek ALC260
by Takashi Iwai 15 Mar '10

15 Mar '10

At Mon, 15 Mar 2010 01:06:17 +0100, Alessandro Guido wrote: > > With 2.6.34-rc1+ my laptop (Sony Vaio VGN-FS215S) can't reproduce any > sound. All controls are missing from alsamixer and this message is found > in dmesg: > > hda_codec: cannot build controlsfor #0 (error -22) > > Kernel config is attached, and the output from alsa-info.sh follows: Does the patch below fix? Takashi --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a83719..ba45868 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6477,7 +6477,7 @@ static struct alc_config_preset alc260_presets[] = { .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), - .adc_nids = alc260_adc_nids, + .adc_nids = alc260_dual_adc_nids, .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_capture_source,

1 0

Re: [alsa-devel] [PATCH]Support MCP89 and GT21x hdmi audio
by Jaroslav Kysela 14 Mar '10

14 Mar '10

On Sat, 13 Mar 2010, VDR User wrote: > [pid 9629] 1268511516.215224 ioctl(18, 0x400c4150, 0xb43631a4) = 0 ^^^^ here xine feeds new samples to the ALSA driver (0x400c4150 == SNDRV_PCM_IOCTL_WRITEI_FRAMES) > [pid 9629] 1268511516.215275 ioctl(18, 0x80044121, 0xb43632cc) = 0 > [pid 9627] 1268511520.700109 ioctl(10, 0xc0104652, 0xb5443f00) = 0 > [pid 9629] 1268511520.700340 ioctl(18, 0x80044121, 0xb43632cc) = -1 > EPIPE (Broken pipe) ^^^^ next iteration (0x80044121 == SNDRV_PCM_IOCTL_DELAY) - see time difference - 4.485 seconds from last WRITEI It seems like a xine bug or the system scheduler bug or any kernel bug affecting the system scheduler. The ALSA subsystem is not reponsible for this behaviour - the ring buffer in your case is for about 0.372 seconds of samples. This limit must not be crossed for the click-free playback. Ask the xine developers for more help or try another player. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

1 0

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 13 Mar '10

13 Mar '10

The following changes since commit fad837c16cdd856c68ce2e1335ad0fe836ed8ecd: Mark Brown (1): Merge commit 'v2.6.34-rc1' into for-2.6.35 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Jassi Brar (3): ASoC: S3C64XX: I2S: Make BCLK independent of sample size ASoC: S3C: I2Sv2: Segregate hw_params callback ASoC: PCM_RATE: Check for KNOT and CONTINUOUS flags Kuninori Morimoto (1): ASoC: da7210: Add 11025/22050/44100/88200 rate support Miguel Aguilar (4): MFD: DaVinci Voice Codec ASoC: DaVinci: Voice Codec Interface ASoC: DaVinci: CQ93VC Voice Codec DaVinci: DM365: Voice Codec support for the DM365 EVM Peter Ujfalusi (4): ASoC: tlv320dac33: Start/stop sequence change ASoC: tlv320dac33: Add option for keeping the BCLK running OMAP3: McBSP: Add interface for FIFO caused delay query ASoC: OMAP3: Report delay caused by the internal FIFO arch/arm/mach-davinci/board-dm365-evm.c | 4 + arch/arm/plat-omap/include/plat/mcbsp.h | 6 + arch/arm/plat-omap/mcbsp.c | 55 ++++++ drivers/mfd/Kconfig | 4 + drivers/mfd/Makefile | 1 + drivers/mfd/davinci_voicecodec.c | 189 +++++++++++++++++++ include/linux/mfd/davinci_voicecodec.h | 126 +++++++++++++ include/sound/tlv320dac33-plat.h | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cq93vc.c | 298 +++++++++++++++++++++++++++++++ sound/soc/codecs/cq93vc.h | 29 +++ sound/soc/codecs/da7210.c | 90 +++++++++- sound/soc/codecs/tlv320dac33.c | 38 ++--- sound/soc/davinci/Kconfig | 27 +++- sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-evm.c | 45 +++++- sound/soc/davinci/davinci-vcif.c | 272 ++++++++++++++++++++++++++++ sound/soc/davinci/davinci-vcif.h | 28 +++ sound/soc/omap/omap-mcbsp.c | 26 +++ sound/soc/s3c24xx/s3c-i2s-v2.c | 27 +--- sound/soc/s3c24xx/s3c2412-i2s.c | 37 ++++ sound/soc/soc-core.c | 12 ++ 23 files changed, 1270 insertions(+), 53 deletions(-) create mode 100644 drivers/mfd/davinci_voicecodec.c create mode 100644 include/linux/mfd/davinci_voicecodec.h create mode 100644 sound/soc/codecs/cq93vc.c create mode 100644 sound/soc/codecs/cq93vc.h create mode 100644 sound/soc/davinci/davinci-vcif.c create mode 100644 sound/soc/davinci/davinci-vcif.h

2 1

[alsa-devel] [PATCHv5 0/2] TWL6040 audio codec initial support
by Olaya, Margarita 13 Mar '10

13 Mar '10

Following patch series add initial support for TWL6040 codec driver. The patch set is based on the series sent from Misael Lopez in October 2009. Changes from v4: - Rename twl6030_codec as twl6040_codec - Rebase the 6 codec patches into a single patch - Rename twl6040_read as twl6040_read_reg_volatile - Use msleep instead of large mdelays - Add more trigger sources - Restructure set_bias_level function to avoid transitions like PREPARE->ON Regards, Margarita --- Misael Lopez Cruz (1): ASoC: TWL6040: Add twl6040 codec driver Margarita Olaya Cabrera (1): OMAP4: PMIC: Rename twl6030_codec as twl6040_codec drivers/mfd/twl-core.c | 4 +- include/linux/i2c/twl.h | 6 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6040.c | 1223 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 141 +++++ 6 files changed, 1375 insertions(+), 5 deletions(-) create mode 100644 sound/soc/codecs/twl6040.c create mode 100644 sound/soc/codecs/twl6040.h

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[alsa-devel] [PATCH] ASoC: PCM_RATE: Check for KNOT and CONTINUOUS flags
by Jassi Brar 12 Mar '10

12 Mar '10

For ASoC, if either CPU or CODEC driver has set the flag, the MACHINE driver should be given a chance to figure out if the dai, that set the flag, can accomodate a rate that it does not explicitly specify but is specified by the dai at the other end of the link. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/soc-core.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 06c38d1..eb73aab 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -404,6 +404,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->playback.formats & cpu_dai->playback.formats; runtime->hw.rates = codec_dai->playback.rates & cpu_dai->playback.rates; + if (codec_dai->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai->playback.rates; + if (cpu_dai->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai->playback.rates; } else { runtime->hw.rate_min = max(codec_dai->capture.rate_min, @@ -421,6 +427,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->capture.formats & cpu_dai->capture.formats; runtime->hw.rates = codec_dai->capture.rates & cpu_dai->capture.rates; + if (codec_dai->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai->capture.rates; + if (cpu_dai->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai->capture.rates; } snd_pcm_limit_hw_rates(runtime); -- 1.6.2.5

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[alsa-devel] [PATCH v2 1/5] MFD: DaVinci Voice Codec
by miguel.aguilar＠ridgerun.com 12 Mar '10

12 Mar '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This is the MFD driver for the DaVinci Voice codec, it has two clients: * Voice codec interface * Voice codec CQ93VC Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- drivers/mfd/Kconfig | 4 + drivers/mfd/Makefile | 1 + drivers/mfd/davinci_voicecodec.c | 189 ++++++++++++++++++++++++++++++++ include/linux/mfd/davinci_voicecodec.h | 126 +++++++++++++++++++++ 4 files changed, 320 insertions(+), 0 deletions(-) create mode 100644 drivers/mfd/davinci_voicecodec.c create mode 100644 include/linux/mfd/davinci_voicecodec.h diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 8782978..6e93c0b 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -43,6 +43,10 @@ config MFD_SH_MOBILE_SDHI This driver supports the SDHI hardware block found in many SuperH Mobile SoCs. +config MFD_DAVINCI_VOICECODEC + tristate + select MFD_CORE + config MFD_DM355EVM_MSP bool "DaVinci DM355 EVM microcontroller" depends on I2C && MACH_DAVINCI_DM355_EVM diff --git a/drivers/mfd/Makefile b/drivers/mfd/Makefile index ca2f2c4..f5c617f 100644 --- a/drivers/mfd/Makefile +++ b/drivers/mfd/Makefile @@ -9,6 +9,7 @@ obj-$(CONFIG_MFD_SH_MOBILE_SDHI) += sh_mobile_sdhi.o obj-$(CONFIG_HTC_EGPIO) += htc-egpio.o obj-$(CONFIG_HTC_PASIC3) += htc-pasic3.o +obj-$(CONFIG_MFD_DAVINCI_VOICECODEC) += davinci_voicecodec.o obj-$(CONFIG_MFD_DM355EVM_MSP) += dm355evm_msp.o obj-$(CONFIG_MFD_T7L66XB) += t7l66xb.o diff --git a/drivers/mfd/davinci_voicecodec.c b/drivers/mfd/davinci_voicecodec.c new file mode 100644 index 0000000..9886aa8 --- /dev/null +++ b/drivers/mfd/davinci_voicecodec.c @@ -0,0 +1,189 @@ +/* + * DaVinci Voice Codec Core Interface for TI platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/pcm.h> + +#include <linux/mfd/davinci_voicecodec.h> + +u32 davinci_vc_read(struct davinci_vc *davinci_vc, int reg) +{ + return __raw_readl(davinci_vc->base + reg); +} + +void davinci_vc_write(struct davinci_vc *davinci_vc, + int reg, u32 val) +{ + __raw_writel(val, davinci_vc->base + reg); +} + +static int __init davinci_vc_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc; + struct resource *res, *mem; + struct mfd_cell *cell = NULL; + int ret; + + davinci_vc = kzalloc(sizeof(struct davinci_vc), GFP_KERNEL); + if (!davinci_vc) { + dev_dbg(&pdev->dev, + "could not allocate memory for private data\n"); + return -ENOMEM; + } + + davinci_vc->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(davinci_vc->clk)) { + dev_dbg(&pdev->dev, + "could not get the clock for voice codec\n"); + ret = -ENODEV; + goto fail1; + } + clk_enable(davinci_vc->clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no mem resource\n"); + ret = -ENODEV; + goto fail2; + } + + davinci_vc->pbase = res->start; + davinci_vc->base_size = resource_size(res); + + mem = request_mem_region(davinci_vc->pbase, davinci_vc->base_size, + pdev->name); + if (!mem) { + dev_err(&pdev->dev, "VCIF region already claimed\n"); + ret = -EBUSY; + goto fail2; + } + + davinci_vc->base = ioremap(davinci_vc->pbase, davinci_vc->base_size); + if (!davinci_vc->base) { + dev_err(&pdev->dev, "can't ioremap mem resource.\n"); + ret = -ENOMEM; + goto fail3; + } + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + return -ENXIO; + } + + davinci_vc->davinci_vcif.dma_tx_channel = res->start; + davinci_vc->davinci_vcif.dma_tx_addr = + (dma_addr_t)(io_v2p(davinci_vc->base) + DAVINCI_VC_WFIFO); + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + return -ENXIO; + } + + davinci_vc->davinci_vcif.dma_rx_channel = res->start; + davinci_vc->davinci_vcif.dma_rx_addr = + (dma_addr_t)(io_v2p(davinci_vc->base) + DAVINCI_VC_RFIFO); + + davinci_vc->dev = &pdev->dev; + davinci_vc->pdev = pdev; + + /* Voice codec interface client */ + cell = &davinci_vc->cells[DAVINCI_VC_VCIF_CELL]; + cell->name = "davinci_vcif"; + cell->driver_data = davinci_vc; + + /* Voice codec CQ93VC client */ + cell = &davinci_vc->cells[DAVINCI_VC_CQ93VC_CELL]; + cell->name = "cq93vc"; + cell->driver_data = davinci_vc; + + ret = mfd_add_devices(&pdev->dev, pdev->id, davinci_vc->cells, + DAVINCI_VC_CELLS, NULL, 0); + if (ret != 0) { + dev_err(&pdev->dev, "fail to register client devices\n"); + goto fail4; + } + + return 0; + +fail4: + iounmap(davinci_vc->base); +fail3: + release_mem_region(davinci_vc->pbase, davinci_vc->base_size); +fail2: + clk_disable(davinci_vc->clk); + clk_put(davinci_vc->clk); + davinci_vc->clk = NULL; +fail1: + kfree(davinci_vc); + + return ret; +} + +static int __devexit davinci_vc_remove(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + + mfd_remove_devices(&pdev->dev); + + iounmap(davinci_vc->base); + release_mem_region(davinci_vc->pbase, davinci_vc->base_size); + + clk_disable(davinci_vc->clk); + clk_put(davinci_vc->clk); + davinci_vc->clk = NULL; + + kfree(davinci_vc); + + return 0; +} + +static struct platform_driver davinci_vc_driver = { + .driver = { + .name = "davinci_voicecodec", + .owner = THIS_MODULE, + }, + .remove = __devexit_p(davinci_vc_remove), +}; + +static int __init davinci_vc_init(void) +{ + return platform_driver_probe(&davinci_vc_driver, davinci_vc_probe); +} +module_init(davinci_vc_init); + +static void __exit davinci_vc_exit(void) +{ + platform_driver_unregister(&davinci_vc_driver); +} +module_exit(davinci_vc_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci Voice Codec Core Interface"); +MODULE_LICENSE("GPL"); diff --git a/include/linux/mfd/davinci_voicecodec.h b/include/linux/mfd/davinci_voicecodec.h new file mode 100644 index 0000000..0ab6132 --- /dev/null +++ b/include/linux/mfd/davinci_voicecodec.h @@ -0,0 +1,126 @@ +/* + * DaVinci Voice Codec Core Interface for TI platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __LINUX_MFD_DAVINCI_VOICECODEC_H_ +#define __LINUX_MFD_DAVINIC_VOICECODEC_H_ + +#include <linux/kernel.h> +#include <linux/platform_device.h> +#include <linux/mfd/core.h> + +#include <mach/edma.h> + +/* + * Register values. + */ +#define DAVINCI_VC_PID 0x00 +#define DAVINCI_VC_CTRL 0x04 +#define DAVINCI_VC_INTEN 0x08 +#define DAVINCI_VC_INTSTATUS 0x0c +#define DAVINCI_VC_INTCLR 0x10 +#define DAVINCI_VC_EMUL_CTRL 0x14 +#define DAVINCI_VC_RFIFO 0x20 +#define DAVINCI_VC_WFIFO 0x24 +#define DAVINCI_VC_FIFOSTAT 0x28 +#define DAVINCI_VC_TST_CTRL 0x2C +#define DAVINCI_VC_REG05 0x94 +#define DAVINCI_VC_REG09 0xA4 +#define DAVINCI_VC_REG12 0xB0 + +/* DAVINCI_VC_CTRL bit fields */ +#define DAVINCI_VC_CTRL_MASK 0x5500 +#define DAVINCI_VC_CTRL_RSTADC BIT(0) +#define DAVINCI_VC_CTRL_RSTDAC BIT(1) +#define DAVINCI_VC_CTRL_RD_BITS_8 BIT(4) +#define DAVINCI_VC_CTRL_RD_UNSIGNED BIT(5) +#define DAVINCI_VC_CTRL_WD_BITS_8 BIT(6) +#define DAVINCI_VC_CTRL_WD_UNSIGNED BIT(7) +#define DAVINCI_VC_CTRL_RFIFOEN BIT(8) +#define DAVINCI_VC_CTRL_RFIFOCL BIT(9) +#define DAVINCI_VC_CTRL_RFIFOMD_WORD_1 BIT(10) +#define DAVINCI_VC_CTRL_WFIFOEN BIT(12) +#define DAVINCI_VC_CTRL_WFIFOCL BIT(13) +#define DAVINCI_VC_CTRL_WFIFOMD_WORD_1 BIT(14) + +/* DAVINCI_VC_INT bit fields */ +#define DAVINCI_VC_INT_MASK 0x3F +#define DAVINCI_VC_INT_RDRDY_MASK BIT(0) +#define DAVINCI_VC_INT_RERROVF_MASK BIT(1) +#define DAVINCI_VC_INT_RERRUDR_MASK BIT(2) +#define DAVINCI_VC_INT_WDREQ_MASK BIT(3) +#define DAVINCI_VC_INT_WERROVF_MASKBIT BIT(4) +#define DAVINCI_VC_INT_WERRUDR_MASK BIT(5) + +/* DAVINCI_VC_REG05 bit fields */ +#define DAVINCI_VC_REG05_PGA_GAIN 0x07 + +/* DAVINCI_VC_REG09 bit fields */ +#define DAVINCI_VC_REG09_MUTE 0x40 +#define DAVINCI_VC_REG09_DIG_ATTEN 0x3F + +/* DAVINCI_VC_REG12 bit fields */ +#define DAVINCI_VC_REG12_POWER_ALL_ON 0xFD +#define DAVINCI_VC_REG12_POWER_ALL_OFF 0x00 + +#define DAVINCI_VC_CELLS 2 + +enum davinci_vc_cells { + DAVINCI_VC_VCIF_CELL, + DAVINCI_VC_CQ93VC_CELL, +}; + +struct davinci_vcif { + struct platform_device *pdev; + u32 dma_tx_channel; + u32 dma_rx_channel; + dma_addr_t dma_tx_addr; + dma_addr_t dma_rx_addr; +}; + +struct cq93vc { + struct platform_device *pdev; + struct snd_soc_codec *codec; + u32 sysclk; +}; + +struct davinci_vc; + +struct davinci_vc { + /* Device data */ + struct device *dev; + struct platform_device *pdev; + struct clk *clk; + + /* Memory resources */ + void __iomem *base; + resource_size_t pbase; + size_t base_size; + + /* MFD cells */ + struct mfd_cell cells[DAVINCI_VC_CELLS]; + + /* Client devices */ + struct davinci_vcif davinci_vcif; + struct cq93vc cq93vc; +}; + +#endif -- 1.6.0.4

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