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[alsa-devel] [PATCH 0/5] [RFC] ALSA ASoC Support for TLVaic23b audio codec
by Arun KS 30 Sep '08

30 Sep '08

Hi all, The following patches adds ASOC driver for TLVaic23b audio codec. Tested playback and capture on omap 5912osk board with some hacks in the omap platform and machine driver. Thanks, Arun -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

3 11

Re: [alsa-devel] Secure delivery
by ink＠jurassic.park.msu.ru 29 Sep '08

29 Sep '08

Bad Gateway: The message has been attached. ++++ Attachment: No Virus found ++++ F-Secure AntiVirus - www.f-secure.com

1 0

[alsa-devel] [PATCH 0/6] ASoC Blackfin and API updates
by Mark Brown 29 Sep '08

29 Sep '08

The following changes since commit ad7f41b60efde140df0f929e8d408ebe2e59cb0e: Jean Delvare (1): ALSA: Drop Vladimir Barinov's e-mail address adding the Blackfin updates Brian posted over the weekend plus some more usage of snd_soc_dapm_nc_pin() are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Cliff Cai (4): ASoC codec: AD73311 audio codec driver ASoC Blackfin: add I2S DAI support for AD73311 ASoC Blackfin: add asoc ad73311 driver supporting in Blackfin boards ASoC Blackfin: fix bug - Audio Latency on AD1981 with MMAP enabled Mark Brown (2): ASoC: Use snd_soc_dapm_nc_pin() in Zaurus machine drivers ASoC: Use snd_soc_dapm_nc_pin() in N810 machine driver sound/soc/blackfin/Kconfig | 16 +++ sound/soc/blackfin/Makefile | 3 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 42 +++++- sound/soc/blackfin/bf5xx-ac97.c | 1 - sound/soc/blackfin/bf5xx-ad73311.c | 240 +++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-i2s.c | 47 +++++-- sound/soc/blackfin/bf5xx-sport.h | 2 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad73311.c | 107 ++++++++++++++++ sound/soc/codecs/ad73311.h | 90 +++++++++++++ sound/soc/omap/n810.c | 6 +- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/spitz.c | 14 +- sound/soc/pxa/tosa.c | 4 +- 16 files changed, 547 insertions(+), 39 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad73311.c create mode 100644 sound/soc/codecs/ad73311.c create mode 100644 sound/soc/codecs/ad73311.h

2 8

[alsa-devel] [PATCH 4/5] [RFC] ALSA ASOC adds destination and source port for dma
by Arun KS 29 Sep '08

29 Sep '08

Adds destination and source ports for dma which is required for OMAP1 Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/omap-pcm.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 690bfea..9e670ef 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -97,6 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, prtd->dma_data = dma_data; err = omap_request_dma(dma_data->dma_req, dma_data->name, omap_pcm_dma_irq, substream, &prtd->dma_ch); + if (!cpu_is_omap1510()) { /* * Link channel with itself so DMA doesn't need any @@ -147,12 +148,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_or_dst_synch = OMAP_DMA_DST_SYNC; dma_params.src_start = runtime->dma_addr; dma_params.dst_start = dma_data->port_addr; + dma_params.dst_port = OMAP_DMA_PORT_MPUI; } else { dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC; dma_params.src_start = dma_data->port_addr; dma_params.dst_start = runtime->dma_addr; + dma_params.src_port = OMAP_DMA_PORT_MPUI; } /* * Set DMA transfer frame size equal to ALSA period size and frame -- 1.5.3.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

4 3

[alsa-devel] [PATCH] ALSA: cs46xx: Add PCI IDs for TerraTec and Hercules cards
by Vedran Miletić 29 Sep '08

29 Sep '08

This patch adds PCI IDs for: * TerraTec DMX XFire 1024 * Hercules Gamesurround Fortissimo II * Hercules Gamesurround Fortissimo III 7.1 All those cards were supported as generic CS46xx device, so they will work as before. I'm pretty sure that first two cards work, as they have same hardware design as reference card. Not sure about Fortissimo III, but this won't break it if it worked. Tested on TerraTec DMX XFire 1024. -- Vedran Miletić

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Re: [alsa-devel] [PATCH 1/5] [RFC] ALSA ASoC driver for TLVaic23b audio codec
by Arun KS 29 Sep '08

29 Sep '08

Mark Brown wrote: > On Mon, Sep 29, 2008 at 03:15:04PM +0530, Arun KS wrote: >> ASOC Audio driver for TLVaic23b audio codec > > This looks good overall, thanks! There's quite a few comments below but > they're mostly minor things at the coding style issues. > > The only issue I've got overall is that the name of the codec seems to > vary through the driver - sometimes it's referred to as the tlv320aic23, > sometimes as aic23 and sometimes as tlvaic23. It'd be good if the > naming scheme were more consistent. My inclination would be to go with > tlv320aic23 since it will reduce the risk of collisions and the general > style is to use the full part number (then again, I know the tlv320aic3x > driver didn't do that). > >> --- a/sound/soc/codecs/Kconfig >> +++ b/sound/soc/codecs/Kconfig >> @@ -55,3 +55,7 @@ config SND_SOC_TWL4030 >> tristate >> depends on SND_SOC && TWL4030_CORE >> > > This patch is against the OMAP tree - please generate all submitted > patches against either the ASoC dev tree or the topic/asoc branch of: > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > > Even if other patches in the series depend on the OMAP tree there should > be no reason not to merge an I2C codec driver like this with no external > dependencies immediately. > >> +config SND_SOC_TLV320AIC23 >> + tristate >> + depends on I2C >> + > > Please also add this to the SND_SOC_ALL_CODECS Kconfig option. > >> +static int aic23_write(struct snd_soc_codec *codec, unsigned int reg, >> + unsigned int value) >> +{ >> + >> + u8 data; >> + >> + /* TLV320AIC23 has 7 bit address and 9 bits of data >> + * so we need to switch one data bit into reg and rest >> + * of data into val >> + */ >> + >> + if ((reg < 0 || reg > 9) && (reg != 15)) { >> + printk(KERN_WARNING "Invalid register R%d\n", reg); >> + return -1; >> + } > > It'd be nice if this said which device was being written to - adding in > the driver name to the printk(), for example. It'd make it easier for > someone seeing the message in their logs to work out where it came from. > >> + data = (reg << 1) | (value >> 8 & 0x01); >> + >> + aic23_write_reg_cache(codec, reg, value); >> + >> + if (codec->hw_write(codec->control_data, (u8) data, >> + (u8) (value & 0xff)) == 0) >> + return 0; >> + >> + printk(KERN_ERR "I2C: cannot write %03x to register R%d\n", value, reg); > > Again, saying where the error came from would be nice. > >> + SOC_SINGLE("Mic Mute", ANALOG_AUDIO_CONTROL_ADDR, 1, 0x01, 0), > > This should be "Mic Switch" to help UIs display the control correctly. > >> +/* aic23 related */ >> +static const struct aic23_samplerate_reg_info >> + rate_reg_info[NUMBER_SAMPLE_RATES_SUPPORTED] = { >> + {4000, 0x06, 1}, /* 4000 */ > > You could just use [] for the size of the array and let the compiler > figure it out - you actually use ARRAY_SIZE() rather than the #define > for the array size later anyway. > >> +static int aic23_set_bias_level(struct snd_soc_codec *codec, >> + enum snd_soc_bias_level level) >> +{ >> + u16 reg; >> + >> + switch (level) { >> + case SND_SOC_BIAS_ON: >> + reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR); >> + aic23_write(codec, POWER_DOWN_CONTROL_ADDR, >> + reg & (~DEVICE_POWER_OFF)); >> + /* Activate interface */ >> + reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR); >> + aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg | ACT_ON); >> + break; >> + case SND_SOC_BIAS_PREPARE: >> + break; >> + case SND_SOC_BIAS_STANDBY: >> + reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR); >> + aic23_write(codec, POWER_DOWN_CONTROL_ADDR, >> + reg | DEVICE_POWER_OFF); >> + /* Deactivate interface */ >> + reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR); >> + aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg & (~ACT_ON)); > > Hrm. This probably deserves a comment about what DEVICE_POWER_OFF does > that the normal DAPM control doesn't. Given that there are no bypass > paths there should be no risk of interfering with DAPM but it raises a > bit of an eyebrow. There is bypass for Line and sidetone for Mic. I will add that also and will make DEVICE_POWER_OFF also as DAPM control. Will send the patches against ASoC dev tree. > >> +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) >> + codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data; >> + codec->hw_read = NULL; >> + ret = i2c_add_driver(&aic23b_i2c_driver); >> + if (ret != 0) >> + printk(KERN_ERR "can't add i2c driver"); >> +#else >> + /* Add other interfaces here */ >> +#endif > > If the device doesn't support anything except I2C then please remove all > the conditionals on I2C support from the driver (they don't add > anything). If the device does support other control methods then please > change this to be something like: > > #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) > ... > #endif > /* Add other interfaces here */ > > That way if someone adds support for the other control interfaces > they'll be encouraged to do it in a way that allows kernels to be built > supporting multiple systems simultaneously. > >> +#define AIC23_CACHEREGNUM 16 >> +/* Codec TLV320AIC23 */ >> +#define LEFT_LINE_VOLUME_ADDR 0x00 >> +#define RIGHT_LINE_VOLUME_ADDR 0x01 >> +#define LEFT_CHANNEL_VOLUME_ADDR 0x02 >> +#define RIGHT_CHANNEL_VOLUME_ADDR 0x03 >> +#define ANALOG_AUDIO_CONTROL_ADDR 0x04 >> +#define DIGITAL_AUDIO_CONTROL_ADDR 0x05 >> +#define POWER_DOWN_CONTROL_ADDR 0x06 >> +#define DIGITAL_AUDIO_FORMAT_ADDR 0x07 >> +#define SAMPLE_RATE_CONTROL_ADDR 0x08 >> +#define DIGITAL_INTERFACE_ACT_ADDR 0x09 >> +#define RESET_CONTROL_ADDR 0x0F > > These (and most of the rest of the defines in the header) should all be > namespaced - there's a very good chance of them colliding with other > things and this header is going to be included in the machine drivers. > >> +/* Define to set the AIC23 as the master w.r.t McBSP */ >> +#define AIC23_MASTER > >> +#ifndef DEFAULT_BITPERSAMPLE >> +#define DEFAULT_BITPERSAMPLE 16 >> +#endif > >> +#define DEFAULT_SAMPLE_RATE 44100 >> +#define CODEC_CLOCK 12000000 >> +#define AUDIO_MCBSP OMAP_MCBSP1 > > None of these should be in this driver - they should all be part of the > machine driver. > >> +struct aic23_samplerate_reg_info { >> + u32 sample_rate; >> + u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ >> + u8 divider /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ >> +}; > > This could just be pushed into the driver. > >

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[alsa-devel] [PATCH 2/5] [RFC] ALSA ASOC Support for 5912 osk
by Arun KS 29 Sep '08

29 Sep '08

Adding ASOC machine driver for osk Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/Kconfig | 8 ++ sound/soc/omap/Makefile | 3 +- sound/soc/omap/osk.c | 230 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 240 insertions(+), 1 deletions(-) create mode 100644 sound/soc/omap/osk.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index d7b8939..f2e6d01 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -22,3 +22,11 @@ config SND_OMAP_SOC_OVERO help Say Y if you want to add support for SoC audio on the Gumstix Overo. +config SND_OMAP_SOC_OSK + tristate "SoC Audio support for 5912 OSK" + depends on SND_OMAP_SOC && MACH_OMAP_OSK + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC23 + help + Say Y if you want to add support for SoC audio on 5912 OSK. + diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index b96b97b..ccb3ddd 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -8,7 +8,8 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o # OMAP Machine Support snd-soc-n810-objs := n810.o snd-soc-overo-objs := overo.o +snd-soc-osk-objs := osk.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o - +obj-$(CONFIG_SND_OMAP_SOC_OSK) += snd-soc-osk.o diff --git a/sound/soc/omap/osk.c b/sound/soc/omap/osk.c new file mode 100644 index 0000000..f4dfe66 --- /dev/null +++ b/sound/soc/omap/osk.c @@ -0,0 +1,230 @@ +/* + * osk.c -- SoC audio for 5912 OSK + * + * Copyright (C) 2008 Mistral Solutions + * + * Contact: Arun KS <arunks(a)mistralsolutions.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/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <linux/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/tlv320aic23.h" + +static struct clk *aic23_mclk; + +static int osk_startup(struct snd_pcm_substream *substream) +{ + return clk_enable(aic23_mclk); +} + +static void osk_shutdown(struct snd_pcm_substream *substream) +{ + clk_disable(aic23_mclk); +} + +static int osk_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return err; + } + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return err; + } + + /* Set the codec system clock for DAC and ADC */ + err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN); + + if (err < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return err; + } + + return err; +} + +static struct snd_soc_ops osk_ops = { + .startup = osk_startup, + .hw_params = osk_hw_params, + .shutdown = osk_shutdown, +}; + +static const struct snd_soc_dapm_widget aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic Jack"}, +}; + +static int osk_aic23_init(struct snd_soc_codec *codec) +{ + + /* Add OSK specific widgets */ + snd_soc_dapm_new_controls(codec, aic23_dapm_widgets, + ARRAY_SIZE(aic23_dapm_widgets)); + + /* Set up OSK specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link osk_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &aic23_dai, + .init = osk_aic23_init, + .ops = &osk_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_osk = { + .name = "5912OSK", + .dai_link = &osk_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device osk_snd_devdata = { + .machine = &snd_soc_machine_osk, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_aic23, +}; + +static struct platform_device *osk_snd_device; + +#define CODEC_CLOCK 12000000 + +static int __init osk_soc_init(void) +{ + int err; + u32 curRate; + struct device *dev; + + if (!(machine_is_omap_osk())) + return -ENODEV; + + osk_snd_device = platform_device_alloc("soc-audio", -1); + if (!osk_snd_device) + return -ENOMEM; + + platform_set_drvdata(osk_snd_device, &osk_snd_devdata); + osk_snd_devdata.dev = &osk_snd_device->dev; + *(unsigned int *)osk_dai.cpu_dai->private_data = 0; /* McBSP1 */ + err = platform_device_add(osk_snd_device); + if (err) + goto err1; + + dev = &osk_snd_device->dev; + + aic23_mclk = clk_get(dev, "mclk"); + if (IS_ERR(aic23_mclk)) { + printk(KERN_ERR "Could not get mclk clock\n"); + return -ENODEV; + } + + if (clk_get_usecount(aic23_mclk) > 0) { + /* MCLK is already in use */ + printk(KERN_WARNING + "MCLK in use at %d Hz. We change it to %d Hz\n", + (uint) clk_get_rate(aic23_mclk), CODEC_CLOCK); + } + + /* + * Configure 12 MHz output on MCLK. + */ + curRate = (uint) clk_get_rate(aic23_mclk); + if (curRate != CODEC_CLOCK) { + if (clk_set_rate(aic23_mclk, CODEC_CLOCK)) { + printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n"); + return -ECANCELED; + } + } + + printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n", + (uint) clk_get_rate(aic23_mclk), CODEC_CLOCK, + clk_get_usecount(aic23_mclk)); + + return 0; +err1: + clk_put(aic23_mclk); + platform_device_del(osk_snd_device); + platform_device_put(osk_snd_device); + + return err; + +} + +static void __exit osk_soc_exit(void) +{ + platform_device_unregister(osk_snd_device); +} + +module_init(osk_soc_init); +module_exit(osk_soc_exit); + +MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); +MODULE_DESCRIPTION("ALSA SoC 5912 OSK"); +MODULE_LICENSE("GPL"); -- 1.5.3.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

4 3

[alsa-devel] ICH9 Sound
by Ronald Horn 29 Sep '08

29 Sep '08

Hello, I'm using a HP Pavilion dv7-1045eg. The Problem is I have no sound, because there is no driver witch works. Here are some reports from my system: / lspci -nv/ 00:1b.0 0403: 8086:293e (rev 03) Subsystem: 103c:30f4 Flags: bus master, fast devsel, latency 0, IRQ 22 Memory at df300000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel / cat /proc/asound/cards/ 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xdf300000 irq 22 /head -n 1 /proc/asound/card0/codec*/ ==> /proc/asound/card0/codec#0 <== Codec: IDT 92HD71B7X ==> /proc/asound/card0/codec#2 <== Codec: Generic 10de ID 6 I'm using openSuse 11.0 with Kernel 2.6.25.16-0.1-pae. Maybe your project could develop some drivers. Greetings Ronald

2 1

[alsa-devel] [PATCH] ASoC: Fix another cs4270 error path
by Jean Delvare 29 Sep '08

29 Sep '08

Conversion to new-style i2c driver missed the error path of the probe function. Fix it. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) --- linux-2.6.27-rc7.orig/sound/soc/codecs/cs4270.c 2008-09-27 17:45:47.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/codecs/cs4270.c 2008-09-27 18:10:52.000000000 +0200 @@ -610,17 +610,12 @@ static int cs4270_i2c_probe(struct i2c_c return 0; error: - if (codec->control_data) { - i2c_detach_client(i2c_client); - codec->control_data = NULL; - } + codec->control_data = NULL; kfree(codec->reg_cache); codec->reg_cache = NULL; codec->reg_cache_size = 0; - kfree(i2c_client); - return ret; } -- Jean Delvare

2 1

[alsa-devel] [PATCH 1/5] [RFC] ALSA ASoC driver for TLVaic23b audio codec
by Arun KS 29 Sep '08

29 Sep '08

ASOC Audio driver for TLVaic23b audio codec Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic23.c | 634 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23.h | 143 +++++++++ 4 files changed, 783 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic23.c create mode 100644 sound/soc/codecs/tlv320aic23.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b1a5eed..7be0b37 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -55,3 +55,7 @@ config SND_SOC_TWL4030 tristate depends on SND_SOC && TWL4030_CORE +config SND_SOC_TLV320AIC23 + tristate + depends on I2C + diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a519ced..adb5aa6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -11,6 +11,7 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-twl4030-objs := twl4030.o +snd-soc-tlv320aic23-objs := tlv320aic23.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o @@ -25,3 +26,4 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c new file mode 100644 index 0000000..eefeb2d --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.c @@ -0,0 +1,634 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks(a)mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/tlv320aic23.c by Vladimir Barinov + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Notes: + * The AIC23 is a driver for a low power stereo audio + * codec aic23 + * + * The machine layer should disable unsupported inputs/outputs by + * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "tlv320aic23.h" + +#define AUDIO_NAME "aic23" +#define AIC23_VERSION "0.1" + +/* codec private data */ +struct aic23_priv { + unsigned int sysclk; + int master; +}; + +/* + * AIC23 register cache + */ +static const u16 aic23_reg[AIC23_CACHEREGNUM] = { + 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ + 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ + 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ +}; + +/* + * read aic23 register cache + */ +static inline unsigned int aic23_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AIC23_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write aic23 register cache + */ +static inline void aic23_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u16 value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AIC23_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the aic23 register space + */ +static int aic23_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + + u8 data; + + /* TLV320AIC23 has 7 bit address and 9 bits of data + * so we need to switch one data bit into reg and rest + * of data into val + */ + + if ((reg < 0 || reg > 9) && (reg != 15)) { + printk(KERN_WARNING "Invalid register R%d\n", reg); + return -1; + } + + data = (reg << 1) | (value >> 8 & 0x01); + + aic23_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, (u8) data, + (u8) (value & 0xff)) == 0) + return 0; + + printk(KERN_ERR "I2C: cannot write %03x to register R%d\n", value, reg); + + return -EIO; +} + +static const struct snd_kcontrol_new aic23_snd_controls[] = { + SOC_DOUBLE_R("PCM Playback Volume", LEFT_CHANNEL_VOLUME_ADDR, + RIGHT_CHANNEL_VOLUME_ADDR, 0, 0x7f, 0), + SOC_SINGLE("PCM Playback Switch", DIGITAL_AUDIO_CONTROL_ADDR, 3, 0x01, + 1), + SOC_DOUBLE_R("Line Input Mute", LEFT_LINE_VOLUME_ADDR, + RIGHT_LINE_VOLUME_ADDR, 7, 0x01, 0), + SOC_DOUBLE_R("Line Input Volume", LEFT_LINE_VOLUME_ADDR, + RIGHT_LINE_VOLUME_ADDR, 0, 0x1f, 0), + SOC_SINGLE("Mic Mute", ANALOG_AUDIO_CONTROL_ADDR, 1, 0x01, 0), + SOC_SINGLE("Mic Booster Switch", ANALOG_AUDIO_CONTROL_ADDR, 0, 0x01, 0), + +}; + +/* add non dapm controls */ +static int aic23_add_controls(struct snd_soc_codec *codec) +{ + + int err, i; + + for (i = 0; i < ARRAY_SIZE(aic23_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&aic23_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; + +} +static const char *aic23_rec_src[] = { "Line", "Mic" }; + +static const struct soc_enum aic23_enum[] = { + SOC_ENUM_SINGLE(ANALOG_AUDIO_CONTROL_ADDR, 2, 2, aic23_rec_src), +}; + +static const struct snd_kcontrol_new aic23_rec_src_mux_controls = +SOC_DAPM_ENUM("Route", aic23_enum[0]); + +/* PGA Mixer controls for Line and Mic switch */ +static const struct snd_kcontrol_new aic23_pga_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Switch", POWER_DOWN_CONTROL_ADDR, 0, 1, 1), + SOC_DAPM_SINGLE("Mic Switch", POWER_DOWN_CONTROL_ADDR, 1, 1, 1), +}; + +static const struct snd_soc_dapm_widget aic23_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", POWER_DOWN_CONTROL_ADDR, 3, 1), + SND_SOC_DAPM_PGA("OUT", POWER_DOWN_CONTROL_ADDR, 4, 1, NULL, 0), + + SND_SOC_DAPM_ADC("ADC", "Capture", POWER_DOWN_CONTROL_ADDR, 2, 1), + SND_SOC_DAPM_MIXER("PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic23_pga_mixer_controls[0], + ARRAY_SIZE(aic23_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0, + &aic23_rec_src_mux_controls), + + SND_SOC_DAPM_OUTPUT("LHPOUT"), + SND_SOC_DAPM_OUTPUT("RHPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_INPUT("LLINEIN"), + SND_SOC_DAPM_INPUT("RLINEIN"), + SND_SOC_DAPM_INPUT("MICIN"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"OUT", NULL, "DAC"}, + + {"LHPOUT", NULL, "OUT"}, + {"RHPOUT", NULL, "OUT"}, + + {"LOUT", NULL, "OUT"}, + {"ROUT", NULL, "OUT"}, + + {"Capture Source", "Line", "LLINEIN"}, + {"Capture Source", "Line", "RLINEIN"}, + + {"Capture Source", "Mic", "MICIN"}, + + {"PGA Mixer", "Line Switch", "Capture Source"}, + {"PGA Mixer", "Mic Switch", "Capture Source"}, + + {"ADC", NULL, "PGA Mixer"}, + +}; + +/* aic23 related */ +static const struct aic23_samplerate_reg_info + rate_reg_info[NUMBER_SAMPLE_RATES_SUPPORTED] = { + {4000, 0x06, 1}, /* 4000 */ + {8000, 0x06, 0}, /* 8000 */ + {16000, 0x0C, 1}, /* 16000 */ + {22050, 0x11, 1}, /* 22050 */ + {24000, 0x00, 1}, /* 24000 */ + {32000, 0x0C, 0}, /* 32000 */ + {44100, 0x11, 0}, /* 44100 */ + {48000, 0x00, 0}, /* 48000 */ + {88200, 0x1F, 0}, /* 88200 */ + {96000, 0x0E, 0}, /* 96000 */ +}; + +static int aic23_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, aic23_dapm_widgets, + ARRAY_SIZE(aic23_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int aic23_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface_reg, data; + u8 count = 0; + + iface_reg = + aic23_read_reg_cache(codec, + DIGITAL_AUDIO_FORMAT_ADDR) & ~(0x03 << 2); + + /* Search for the right sample rate */ + /* Verify what happens if the rate is not supported + * now it goes to 96Khz */ + while ((rate_reg_info[count].sample_rate != params_rate(params)) && + (count < ARRAY_SIZE(rate_reg_info))) { + count++; + } + + data = (rate_reg_info[count].divider << CLKIN_SHIFT) | + (rate_reg_info[count].control << BOSR_SHIFT) | USB_CLK_ON; + + aic23_write(codec, SAMPLE_RATE_CONTROL_ADDR, data); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_reg |= (0x01 << 2); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_reg |= (0x02 << 2); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_reg |= (0x03 << 2); + break; + } + aic23_write(codec, DIGITAL_AUDIO_FORMAT_ADDR, iface_reg); + + return 0; +} + +static int aic23_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg; + + reg = + aic23_read_reg_cache(codec, + DIGITAL_AUDIO_CONTROL_ADDR) & ~DACM_MUTE; + + if (mute) + aic23_write(codec, DIGITAL_AUDIO_CONTROL_ADDR, reg | DACM_MUTE); + + else + aic23_write(codec, DIGITAL_AUDIO_CONTROL_ADDR, reg); + + return 0; +} + +static int aic23_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 aic23_priv *aic23 = codec->private_data; + + aic23->sysclk = freq; + return 0; +} + +static int aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic23_priv *aic23 = codec->private_data; + u16 iface_reg; + + iface_reg = + aic23_read_reg_cache(codec, DIGITAL_AUDIO_FORMAT_ADDR) & (~0x03); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aic23->master = 1; + iface_reg |= MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + aic23->master = 0; + break; + default: + return -EINVAL; + + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface_reg |= FOR_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= FOR_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg |= FOR_LJUST; + break; + default: + return -EINVAL; + + } + + aic23_write(codec, DIGITAL_AUDIO_FORMAT_ADDR, iface_reg); + + return 0; +} + +static int aic23_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR); + aic23_write(codec, POWER_DOWN_CONTROL_ADDR, + reg & (~DEVICE_POWER_OFF)); + /* Activate interface */ + reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR); + aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg | ACT_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR); + aic23_write(codec, POWER_DOWN_CONTROL_ADDR, + reg | DEVICE_POWER_OFF); + /* Deactivate interface */ + reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR); + aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg & (~ACT_ON)); + + break; + case SND_SOC_BIAS_OFF: + break; + } + + codec->bias_level = level; + + return 0; +} + +#define AIC23_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai aic23_dai = { + .name = "aic23", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .ops = { + .hw_params = aic23_hw_params, + }, + .dai_ops = { + .digital_mute = aic23_mute, + .set_sysclk = aic23_set_dai_sysclk, + .set_fmt = aic23_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(aic23_dai); + +static int aic23_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int aic23_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u8 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(aic23_reg); i++) { + data[0] = i; + data[1] = cache[i]; + codec->hw_write(codec->control_data, data, 2); + } + + aic23_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +/* + * initialise the AIC23 driver + * register the mixer and dsp interfaces with the kernel + */ +static int aic23_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + u16 reg; + + codec->name = "aic23"; + codec->owner = THIS_MODULE; + codec->read = aic23_read_reg_cache; + codec->write = aic23_write; + codec->set_bias_level = aic23_set_bias_level; + codec->dai = &aic23_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(aic23_reg); + codec->reg_cache = kmemdup(aic23_reg, sizeof(aic23_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* Reset codec */ + aic23_write(codec, RESET_CONTROL_ADDR, 0); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "aic23: failed to create pcms\n"); + goto pcm_err; + } + + aic23_write(codec, DIGITAL_AUDIO_CONTROL_ADDR, DEEMP_44K); + + /* Unmute input */ + reg = aic23_read_reg_cache(codec, LEFT_LINE_VOLUME_ADDR); + aic23_write(codec, LEFT_LINE_VOLUME_ADDR, + (reg & (~LIM_MUTED)) | (LRS_ENABLED)); + reg = aic23_read_reg_cache(codec, RIGHT_LINE_VOLUME_ADDR); + aic23_write(codec, RIGHT_LINE_VOLUME_ADDR, + (reg & (~LIM_MUTED)) | LRS_ENABLED); + reg = aic23_read_reg_cache(codec, ANALOG_AUDIO_CONTROL_ADDR); + aic23_write(codec, ANALOG_AUDIO_CONTROL_ADDR, + (reg | DAC_SELECTED) & (~MICM_MUTED) & (~BYPASS_ON)); + + /* Default output volume */ + aic23_write(codec, LEFT_CHANNEL_VOLUME_ADDR, + DEFAULT_OUTPUT_VOLUME & OUTPUT_VOLUME_MASK); + aic23_write(codec, RIGHT_CHANNEL_VOLUME_ADDR, + DEFAULT_OUTPUT_VOLUME & OUTPUT_VOLUME_MASK); + + /* off, with power on */ + aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + aic23_add_controls(codec); + aic23_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "aic23: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} +static struct snd_soc_device *aic23_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * If the i2c layer weren't so broken, we could pass this kind of data + * around + */ +static int aic23_codec_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct snd_soc_device *socdev = aic23_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = aic23_init(socdev); + if (ret < 0) { + printk(KERN_ERR "aic23: failed to initialise AIC23\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} +static int __exit aic23_i2c_remove(struct i2c_client *i2c) +{ + + put_device(&i2c->dev); + return 0; +} + +static const struct i2c_device_id tlvaic23b_id[] = { + {"tlvaic23b", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tlvaic23b_id); + +static struct i2c_driver aic23b_i2c_driver = { + .driver = { + .name = "tlvaic23b", + }, + .probe = aic23_codec_probe, + .remove = __exit_p(aic23_i2c_remove), + .id_table = tlvaic23b_id, +}; + +#endif + +static int aic23_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct aic23_priv *aic23; + int ret = 0; + + printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + aic23 = kzalloc(sizeof(struct aic23_priv), GFP_KERNEL); + if (aic23 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = aic23; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + aic23_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data; + codec->hw_read = NULL; + ret = i2c_add_driver(&aic23b_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); +#else + /* Add other interfaces here */ +#endif + return ret; +} + +static int aic23_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + /* power down chip */ + if (codec->control_data) + aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&aic23b_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} +struct snd_soc_codec_device soc_codec_dev_aic23 = { + .probe = aic23_probe, + .remove = aic23_remove, + .suspend = aic23_suspend, + .resume = aic23_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_aic23); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); +MODULE_AUTHOR("Arun KS"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h new file mode 100644 index 0000000..9c3808c --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.h @@ -0,0 +1,143 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks(a)mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _AIC23_H +#define _AIC23_H + +#define AIC23_CACHEREGNUM 16 +/* Codec TLV320AIC23 */ +#define LEFT_LINE_VOLUME_ADDR 0x00 +#define RIGHT_LINE_VOLUME_ADDR 0x01 +#define LEFT_CHANNEL_VOLUME_ADDR 0x02 +#define RIGHT_CHANNEL_VOLUME_ADDR 0x03 +#define ANALOG_AUDIO_CONTROL_ADDR 0x04 +#define DIGITAL_AUDIO_CONTROL_ADDR 0x05 +#define POWER_DOWN_CONTROL_ADDR 0x06 +#define DIGITAL_AUDIO_FORMAT_ADDR 0x07 +#define SAMPLE_RATE_CONTROL_ADDR 0x08 +#define DIGITAL_INTERFACE_ACT_ADDR 0x09 +#define RESET_CONTROL_ADDR 0x0F + +/* Left (right) line input volume control register */ +#define LRS_ENABLED 0x0100 +#define LIM_MUTED 0x0080 +#define LIV_DEFAULT 0x0017 +#define LIV_MAX 0x001f +#define LIV_MIN 0x0000 + +/* Left (right) channel headphone volume control register */ +#define LZC_ON 0x0080 +#define LHV_DEFAULT 0x0079 +#define LHV_MAX 0x007f +#define LHV_MIN 0x0000 + +/* Analog audio path control register */ +#define STA_REG(x) ((x)<<6) +#define STE_ENABLED 0x0020 +#define DAC_SELECTED 0x0010 +#define BYPASS_ON 0x0008 +#define INSEL_MIC 0x0004 +#define MICM_MUTED 0x0002 +#define MICB_20DB 0x0001 + +/* Digital audio path control register */ +#define DACM_MUTE 0x0008 +#define DEEMP_32K 0x0002 +#define DEEMP_44K 0x0004 +#define DEEMP_48K 0x0006 +#define ADCHP_ON 0x0001 + +/* Power control down register */ +#define DEVICE_POWER_OFF 0x0080 +#define CLK_OFF 0x0040 +#define OSC_OFF 0x0020 +#define OUT_OFF 0x0010 +#define DAC_OFF 0x0008 +#define ADC_OFF 0x0004 +#define MIC_OFF 0x0002 +#define LINE_OFF 0x0001 + +/* Digital audio interface register */ +#define MS_MASTER 0x0040 +#define LRSWAP_ON 0x0020 +#define LRP_ON 0x0010 +#define IWL_16 0x0000 +#define IWL_20 0x0004 +#define IWL_24 0x0008 +#define IWL_32 0x000C +#define FOR_I2S 0x0002 +#define FOR_DSP 0x0003 +#define FOR_LJUST 0x0001 + +/* Sample rate control register */ +#define CLKOUT_HALF 0x0080 +#define CLKIN_HALF 0x0040 +#define BOSR_384fs 0x0002 /* BOSR_272fs when in USB mode */ +#define USB_CLK_ON 0x0001 +#define SR_MASK 0xf +#define CLKOUT_SHIFT 7 +#define CLKIN_SHIFT 6 +#define SR_SHIFT 2 +#define BOSR_SHIFT 1 + +/* Digital interface register */ +#define ACT_ON 0x0001 + +/* Define to set the AIC23 as the master w.r.t McBSP */ +#define AIC23_MASTER + +#define NUMBER_SAMPLE_RATES_SUPPORTED 10 + +/* + * AUDIO related MACROS + */ +#ifndef DEFAULT_BITPERSAMPLE +#define DEFAULT_BITPERSAMPLE 16 +#endif + +#define DEFAULT_SAMPLE_RATE 44100 +#define CODEC_CLOCK 12000000 +#define AUDIO_MCBSP OMAP_MCBSP1 + +#define DEFAULT_OUTPUT_VOLUME 0x70 +#define DEFAULT_INPUT_VOLUME 0x10 /* 0 ==> mute line in */ + +#define OUTPUT_VOLUME_MIN LHV_MIN +#define OUTPUT_VOLUME_MAX LHV_MAX +#define OUTPUT_VOLUME_RANGE (OUTPUT_VOLUME_MAX - OUTPUT_VOLUME_MIN) +#define OUTPUT_VOLUME_MASK OUTPUT_VOLUME_MAX + +#define INPUT_VOLUME_MIN LIV_MIN +#define INPUT_VOLUME_MAX LIV_MAX +#define INPUT_VOLUME_RANGE (INPUT_VOLUME_MAX - INPUT_VOLUME_MIN) +#define INPUT_VOLUME_MASK INPUT_VOLUME_MAX + +#define SIDETONE_MASK 0x1c0 +#define SIDETONE_0 0x100 +#define SIDETONE_6 0x000 +#define SIDETONE_9 0x040 +#define SIDETONE_12 0x080 +#define SIDETONE_18 0x0c0 + +#define DEFAULT_ANALOG_AUDIO_CONTROL (DAC_SELECTED | STE_ENABLED | \ + INSEL_MIC | MICB_20DB) + +struct aic23_samplerate_reg_info { + u32 sample_rate; + u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ + u8 divider /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ +}; + +extern struct snd_soc_dai aic23_dai; +extern struct snd_soc_codec_device soc_codec_dev_aic23; + +#endif /* _AIC23_H */ + -- 1.5.3.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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