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[alsa-devel] Problem: different volumes for different jacks in ALC888
by Vincent Petry 22 Nov '08

22 Nov '08

Hi, We have designed a new model for the following laptop: Amilo Xa3530 from Fujitsu Siemens (see bug report 0004240 for alsa-info). The chipset is Realtek ALC888. When using channel mode with 6ch or 8ch, the mic-in jack (0x18) is converted to an output (PIN_OUT 0x40). The problem is that the CLFE channel (0x0e) on the mic-in jack (0x18) is not loud enough. When reassigning the CLFE (0x0e) to the line-out jack (0x17), the volume is at a normal level. But in the codec there is no volume difference between the connection paths from mic-in and side path. I saw that the Mic-in also has a VREF value, but changing it doesn't affect the volume. AFAIK the VREF value only affects Has someone met this kind of issue before? Thanks, Vincent Petry

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Re: [alsa-devel] Creative X-Fi Xtreme Audio (CA0110 chip)
by Vedran Miletić 21 Nov '08

21 Nov '08

Well, don't know what's going on, but anway, snapshots of sound-unstable-2.6 are located here: http://ftp.kernel.org/pub/linux/kernel/people/tiwai/alsa/alsa-driver/alsa-d… Lastest one doesn't compile for me on Debian 2.6.26 (pcsp problems or something), but you can always isolate the exact drivers you need and it's likely that those will compile without problems. 2008/11/21 Dmitri Seletski <drjoms(a)gmail.com>: > Hi Vedran, once again, could I get experimental drivers for Creative XFI pro > gamer drivers?(link sent to manuel seemed to be broken) > Regards > > On Thu, Nov 20, 2008 at 1:51 PM, Vedran Miletić <rivanvx(a)gmail.com> wrote: >> >> CA0110 HDA mode is supported in this tree: >> >> http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-unstable-2.6.git;a=su… >> CA0111 (PCI version) is supported by snd-ca0106, not sure if CA0106 >> mode can be supported that way too. >> >> 2008/11/19 Manuel C. <manuel(a)ekerazha.com>: >> > Hi, >> > I want to bring to your attention the Creative X-Fi Xtreme Audio card >> > with CA0110 chip, it is also bundled with some MSI motherboards. >> > >> > CA0110 web page: http://www.creative.com/oem/products/chips/CA0110.asp >> > >> > >From that web page: >> > "CA0110 is a PCI Express audio chip using the CA0111 audio mode or High >> > Definition Audio mode. >> > >> > [...] >> > >> > CA0106 mode >> > - Same as original CA0111." >> > >> > So it should be a PCI Express version of the CA0111 chip (and CA0111 >> > could be similar to the CA0106 chip). >> > >> > The CA0110 chip isn't even listed in the Creative products matrix on the >> > wiki. Is a CA0110 driver under development? Thank you. >> > >> > _______________________________________________ >> > Alsa-devel mailing list >> > Alsa-devel(a)alsa-project.org >> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> > >> >> >> >> -- >> Vedran Miletić >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- Vedran Miletić

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[alsa-devel] [PATCH] ASoC: Add support for OMAP3 EVM
by Anuj Aggarwal 21 Nov '08

21 Nov '08

This patch adds ALSA SoC support for OMAP3 EVM using TWL4030 audio codec. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/Kconfig | 7 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3evm.c | 147 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 156 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3evm.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 6c56277..2a2a74e 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -38,5 +38,12 @@ config SND_OMAP_SOC_OMAP2EVM help Say Y if you want to add support for SoC audio on the omap2evm board. +config SND_OMAP_SOC_OMAP3EVM + tristate "SoC Audio support for OMAP3EVM board" + depends on SND_OMAP_SOC && MACH_OMAP3EVM + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the omap3evm board. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index f5da3cc..bbb3a77 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -10,9 +10,11 @@ snd-soc-n810-objs := n810.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o +snd-soc-omap3evm-objs := omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o +obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c new file mode 100644 index 0000000..570af55 --- /dev/null +++ b/sound/soc/omap/omap3evm.c @@ -0,0 +1,147 @@ +/* + * omap3evm.c -- ALSA SoC support for OMAP3 EVM + * + * Author: Anuj Aggarwal <anuj.aggarwal(a)ti.com> + * + * Based on sound/soc/omap/beagle.c by Steve Sakoman + * + * Copyright (C) 2008 Texas Instruments, Incorporated + * + * 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 version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind, + * whether express or implied; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#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 <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int omap3evm_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 ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3evm_ops = { + .hw_params = omap3evm_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3evm_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap3evm_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_omap3evm = { + .name = "omap3evm", + .dai_link = &omap3evm_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3evm_snd_devdata = { + .machine = &snd_soc_machine_omap3evm, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3evm_snd_device; + +static int __init omap3evm_soc_init(void) +{ + int ret; + + if (!machine_is_omap3evm()) { + pr_debug("Not OMAP3 EVM!\n"); + return -ENODEV; + } + pr_info("OMAP3 EVM SoC init\n"); + + omap3evm_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap3evm_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap3evm_snd_device, &omap3evm_snd_devdata); + omap3evm_snd_devdata.dev = &omap3evm_snd_device->dev; + *(unsigned int *)omap3evm_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(omap3evm_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap3evm_snd_device); + + return ret; +} + +static void __exit omap3evm_soc_exit(void) +{ + platform_device_unregister(omap3evm_snd_device); +} + +module_init(omap3evm_soc_init); +module_exit(omap3evm_soc_exit); + +MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM"); +MODULE_LICENSE("GPL"); -- 1.6.0.2.1172.ga5ed0

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[alsa-devel] Where are dma interrupts handled in ASoC code?
by John Kacur 21 Nov '08

21 Nov '08

Where are dma overflow interrupts handled? For example I don't see this in sound/soc/s3c24xx-i2s.c Thanks

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[alsa-devel] twl4030 asoc kcontrols and widgets
by naveen krishna ch 21 Nov '08

21 Nov '08

Hi Alsa devel, I have been working on TWL4030 codec driver for ALSA SOC. I have taken sound/soc/codec/twl4030.c as reference from main line This Patch adds some kcontrols, widgets and interconnection map for some of the TWL4030 ASOC codec I was building it for a custom board as it was for OVERO Suggestions on the DAPM part of the driver would be helpful Thanks in advance. The patch is as follows. --- twl4030.c 2008-11-19 12:04:32.000000000 +0530 +++ /home/chnaveen/Desktop/twl4030.c 2008-11-21 15:08:06.000000000 +0530 @@ -16,7 +16,9 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA * 02110-1301 USA - * + * Modified by : Naveen Krishna Ch + * Added Kcontrols for OMAP3 WaterlooBoard + * Dated : 28th october 2008 */ #include <linux/module.h> @@ -29,24 +31,27 @@ #include <linux/i2c/twl4030.h> #include <sound/core.h> #include <sound/pcm.h> +#include <sound/jack.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> - +#include <asm/arch/twl4030.h> #include "twl4030.h" +static int twl4030_jack_func; + /* * twl4030 register cache & default register settings */ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* this register not used */ - 0x93, /* REG_CODEC_MODE (0x1) */ + 0x03, /* REG_CODEC_MODE (0x1) */ 0xc3, /* REG_OPTION (0x2) */ 0x00, /* REG_UNKNOWN (0x3) */ 0x00, /* REG_MICBIAS_CTL (0x4) */ - 0x24, /* REG_ANAMICL (0x5) */ - 0x04, /* REG_ANAMICR (0x6) */ + 0xb0, /* REG_ANAMICL (0x5) */ + 0x10, /* REG_ANAMICR (0x6) */ 0x0a, /* REG_AVADC_CTL (0x7) */ 0x00, /* REG_ADCMICSEL (0x8) */ 0x00, /* REG_DIGMIXING (0x9) */ @@ -67,22 +72,22 @@ 0x00, /* REG_ARX2VTXPGA (0x18) */ 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ - 0x4b, /* REG_ARXL2_APGA_CTL (0x1B) */ - 0x4b, /* REG_ARXR2_APGA_CTL (0x1C) */ + 0x2b, /* REG_ARXL2_APGA_CTL (0x1B) */ + 0x2b, /* REG_ARXR2_APGA_CTL (0x1C) */ 0x00, /* REG_ATX2ARXPGA (0x1D) */ 0x00, /* REG_BT_IF (0x1E) */ 0x00, /* REG_BTPGA (0x1F) */ 0x00, /* REG_BTSTPGA (0x20) */ - 0x00, /* REG_EAR_CTL (0x21) */ - 0x24, /* REG_HS_SEL (0x22) */ - 0x0a, /* REG_HS_GAIN_SET (0x23) */ - 0x00, /* REG_HS_POPN_SET (0x24) */ - 0x00, /* REG_PREDL_CTL (0x25) */ - 0x00, /* REG_PREDR_CTL (0x26) */ - 0x00, /* REG_PRECKL_CTL (0x27) */ - 0x00, /* REG_PRECKR_CTL (0x28) */ - 0x00, /* REG_HFL_CTL (0x29) */ - 0x00, /* REG_HFR_CTL (0x2A) */ + 0x20, /* REG_EAR_CTL (0x21) */ + 0x00, /* REG_HS_SEL (0x22) */ + 0x05, /* REG_HS_GAIN_SET (0x23) */ + 0x42, /* REG_HS_POPN_SET (0x24) */ + 0x20, /* REG_PREDL_CTL (0x25) */ + 0x20, /* REG_PREDR_CTL (0x26) */ + 0x20, /* REG_PRECKL_CTL (0x27) */ + 0x20, /* REG_PRECKR_CTL (0x28) */ + 0x1f, /* REG_HFL_CTL (0x29) */ + 0x1f, /* REG_HFR_CTL (0x2A) */ 0x00, /* REG_ALC_CTL (0x2B) */ 0x00, /* REG_ALC_SET1 (0x2C) */ 0x00, /* REG_ALC_SET2 (0x2D) */ @@ -112,9 +117,40 @@ 0x00, /* REG_VIBRA_CTL (0x45) */ 0x00, /* REG_VIBRA_SET (0x46) */ 0x00, /* REG_VIBRA_PWM_SET (0x47) */ - 0x00, /* REG_ANAMIC_GAIN (0x48) */ + 0x24, /* REG_ANAMIC_GAIN (0x48) */ 0x00, /* REG_MISC_SET_2 (0x49) */ }; +static void twl4030_ext_control(struct snd_soc_codec *codec) +{ + if (twl4030_jack_func) + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + snd_soc_dapm_sync(codec); +} + +static int twl4030_get_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = twl4030_jack_func; + + return 0; +} + +static int twl4030_set_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (twl4030_jack_func == ucontrol->value.integer.value[0]) + return 0; + + twl4030_jack_func = ucontrol->value.integer.value[0]; + twl4030_ext_control(codec); + + return 1; +} /* * read twl4030 register cache @@ -188,14 +224,83 @@ twl4030_write(codec, i, twl4030_reg[i]); } +static const char *jack_function[] = {"Off", "Headphone"}; + +static const struct soc_enum twl4030_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function), +}; + static const struct snd_kcontrol_new twl4030_snd_controls[] = { - SOC_DOUBLE_R("Master Playback Volume", - TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, - 0, 127, 0), - SOC_DOUBLE_R("Capture Volume", - TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, - 0, 127, 0), + + /* Master Playback Volume Controls */ + SOC_DOUBLE_R("Master PLayback Course Gain ctrl", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 6, 3, 0), + SOC_DOUBLE_R("Master Playback Fine Gain ctrl", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 0, 63, 0), + + /* Playback Speaker volume controls */ + SOC_DOUBLE_R("Speaker R2+L2 Volume ctrls", + TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL, + 3, 17, 0), + + /* Capture Gain controls */ + SOC_DOUBLE_R("Master Capture Gain ctrl", + TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, + 0, 31, 0), + /* Loop gain controls*/ + SOC_DOUBLE("Loop Gain ctrl", TWL4030_REG_ATX2ARXPGA, + 3 , 0, 7, 0), + + SOC_DOUBLE("Main +Sub mic capture gain ctrl", + TWL4030_REG_ANAMIC_GAIN, 3 , 0, 5, 0), + + SOC_DOUBLE_R("External Speaker Volume control", + TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL, + 4, 3, 0), + + SOC_DOUBLE_R("Pre Car kit Volume control", + TWL4030_REG_PRECKL_CTL, TWL4030_REG_PRECKR_CTL, + 4, 3, 0), + + SOC_SINGLE("DACL2", TWL4030_REG_AVDAC_CTL, 3, 1, 0), + SOC_SINGLE("DACR2", TWL4030_REG_AVDAC_CTL, 2, 1, 0), + + SOC_ENUM_EXT("Jack Function", twl4030_enum[0], + twl4030_get_jack, twl4030_set_jack), +}; + +/* Right PGA Mixer control switches */ +static const struct snd_kcontrol_new twl4030_right_pga_mixer_controls[] = { + SOC_DAPM_SINGLE("HS Mic switch", TWL4030_REG_ANAMICL, 1, 1, 0), + SOC_DAPM_SINGLE("Aux/FM right switch", TWL4030_REG_ANAMICR, 2, 1, 0), + SOC_DAPM_SINGLE("Sub Mic switch", TWL4030_REG_ANAMICR, 0, 1, 0), +}; + +/* Left PGA Mixer control switches */ +static const struct snd_kcontrol_new twl4030_left_pga_mixer_controls[] = { + SOC_DAPM_SINGLE("HS Mic switch", TWL4030_REG_ANAMICL, 1, 1, 0), + SOC_DAPM_SINGLE("Aux/FM left switch", TWL4030_REG_ANAMICL, 2, 1, 0), + SOC_DAPM_SINGLE("Main Mic switch", TWL4030_REG_ANAMICL, 0, 1, 0), +}; + +/* Right DACR2 Mixer */ +static const struct snd_kcontrol_new twl4030_dacr2_mixer_controls[] = { + SOC_DAPM_SINGLE("Headset-R switch", TWL4030_REG_HS_SEL, 5, 1, 0), + SOC_DAPM_SINGLE("Handsfree-R switch", TWL4030_REG_HFR_CTL, 5, 1, 0), + SOC_DAPM_SINGLE("PRECK-R switch", TWL4030_REG_PRECKR_CTL, 2, 1, 0), + SOC_DAPM_DOUBLE("E class-D amp-R switch", TWL4030_REG_PREDR_CTL, 3, 2, 1, 0, 0), +}; + +/* Left DACL2 Mixer */ +static const struct snd_kcontrol_new twl4030_dacl2_mixer_controls[] = { + SOC_DAPM_SINGLE("Headset-L switch", TWL4030_REG_HS_SEL, 2, 1, 0), + SOC_DAPM_SINGLE("Handsfree-L switch", TWL4030_REG_HFL_CTL, 5, 1, 0), + SOC_DAPM_SINGLE("Ear peice switch", TWL4030_REG_EAR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("PRECK-L switch", TWL4030_REG_PRECKL_CTL, 2, 1, 0), + SOC_DAPM_DOUBLE("E class-D amp-L switch", TWL4031_REG_PREDL_CTL, 3, 2, 1, 0, 0), }; /* add non dapm controls */ @@ -215,27 +320,96 @@ } static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("INL"), - SND_SOC_DAPM_INPUT("INR"), - - SND_SOC_DAPM_OUTPUT("OUTL"), - SND_SOC_DAPM_OUTPUT("OUTR"), - - SND_SOC_DAPM_DAC("DACL", "Left Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DACR", "Right Playback", SND_SOC_NOPM, 0, 0), - - SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_HP("Headphone Jack", NULL), + + /* Left ADC for capture */ + SND_SOC_DAPM_ADC("ADCL", "Left Capture ADC", TWL4030_REG_AVADC_CTL, 3, 0), + + /* dapm widget (path domain) for left PGA mixer */ + SND_SOC_DAPM_MIXER("LINEIN-L", SND_SOC_NOPM, 0, 0, + &twl4030_left_pga_mixer_controls[0], + ARRAY_SIZE(twl4030_left_pga_mixer_controls)), + + /* Right ADC for capture*/ + SND_SOC_DAPM_ADC("ADCR", "Right Capture ADC", TWL4030_REG_AVADC_CTL, 1, 0), + + /* dapm widget (path domain) for right PGA mixer */ + SND_SOC_DAPM_MIXER("LINEIN-R", SND_SOC_NOPM, 0, 0, + &twl4030_right_pga_mixer_controls[0], + ARRAY_SIZE(twl4030_right_pga_mixer_controls)), + + /* dapm widget (path domain) for left DACL2 Mixer */ + + SND_SOC_DAPM_MIXER("DACL2 Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dacl2_mixer_controls[0], + ARRAY_SIZE(twl4030_dacl2_mixer_controls)), + + /* dapm widget (path domain) for DACR2 Mixer */ + SND_SOC_DAPM_MIXER("DACR2 Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dacr2_mixer_controls[0], + ARRAY_SIZE(twl4030_dacr2_mixer_controls)), + + /* Inputs Left*/ + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("Aux/fm left"), + SND_SOC_DAPM_INPUT("Main mic"), + + /* Inputs Right*/ + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("Aux/fm right"), + SND_SOC_DAPM_INPUT("Sub mic"), + + /* Outputs Left*/ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("IHF_LEFT"), + SND_SOC_DAPM_OUTPUT("EAR"), + SND_SOC_DAPM_OUTPUT("E class-D L"), + SND_SOC_DAPM_OUTPUT("PRECK-L"); + + /* Outputs Right */ + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("IHF_RIGHT"), + SND_SOC_DAPM_OUTPUT("E class-D R"), + SND_SOC_DAPM_OUTPUT("PRECK-R"); }; static const struct snd_soc_dapm_route intercon[] = { - /* outputs */ - {"OUTL", NULL, "DACL"}, - {"OUTR", NULL, "DACR"}, - - /* inputs */ - {"ADCL", NULL, "INL"}, - {"ADCR", NULL, "INR"}, + + /* ******** Left input ******** */ + {"LINEIN-L", "HS Mic switch", "HSMIC"}, + {"LINEIN-L", "Aux/FM left switch", "Aux/fm left"}, + {"LINEIN-L", "Main Mic switch", "Main mic"}, + + {"ADCL", NULL, "LINEIN-L"}, + + + /* ******** Right Input ******** */ + {"LINEIN-R", "HS Mic switch", "HSMIC"}, + {"LINEIN-R", "Aux/FM right switch", "Aux/fm right"}, + {"LINEIN-R", "Sub Mic switch", "Sub mic"}, + + {"ADCR", NULL, "LINEIN-R"}, + + /* ******** Left Output ******** */ + //{"DACL2", NULL, "DACL2 Mixer"}, + + {"DACL2 Mixer", "Headset-L switch", "HSOL"}, + {"DACL2 Mixer", "Handsfree-L switch", "IHF_LEFT"}, + {"DACL2 Mixer", "Ear peice switch", "EAR"}, + {"DACL2 Mixer", "PRECK-L switch", "PRECK-L"}, + {"DACL2 Mixer","E class-D amp-L switch","E class-D L"}, + + {"Headphone Jack",NULL, "HSOL"}, + /* ******** Right Output ******** */ + //{"DACR2", NULL, "DACR2 Mixer"}, + + {"DACR2 Mixer", "Headset-R switch","HSOR"}, + {"DACR2 Mixer", "Handsfree-R switch","IHF_RIGHT"}, + {"DACR2 Mixer", "PRECK-R switch", "PRECK-R"}, + {"DACR2 Mixer","E class-D amp-R switch","E class-D R"}, + + {"Headphone Jack",NULL, "HSOR"}, }; static int twl4030_add_widgets(struct snd_soc_codec *codec) @@ -280,7 +454,7 @@ /* toggle CODECPDZ as per TRM */ twl4030_clear_codecpdz(codec); twl4030_set_codecpdz(codec); - + /* program anti-pop with bias ramp delay */ popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); popn &= TWL4030_RAMP_DELAY; @@ -384,6 +558,9 @@ case 48000: mode |= TWL4030_APLL_RATE_48000; break; + case 96000: + mode |= TWL4030_APLL_RATE_96000; + break; default: printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n", params_rate(params)); @@ -434,6 +611,7 @@ u8 infreq; switch (freq) { + case 19200000: infreq = TWL4030_APLL_INFREQ_19200KHZ; break; @@ -504,20 +682,24 @@ return 0; } -#define TWL4030_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define TWL4030_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) + #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) struct snd_soc_dai twl4030_dai = { .name = "twl4030", .playback = { .stream_name = "Playback", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, .capture = { .stream_name = "Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, @@ -616,13 +798,14 @@ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) - return -ENOMEM; - + return -ENOMEM; + socdev->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); - + + twl4030_socdev = socdev; twl4030_init(socdev); Patch ends.. I need suggestions on the DAPM part of the driver. Thanks and Regards, (: Naveen Krishna Ch :)

4 7

[alsa-devel] [PATCH 1/1] ASoC: Fix typo in snd_soc_card update documentation
by Mark Brown 21 Nov '08

21 Nov '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Documentation/sound/alsa/soc/machine.txt | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt index 4a9f51e..bab7711 100644 --- a/Documentation/sound/alsa/soc/machine.txt +++ b/Documentation/sound/alsa/soc/machine.txt @@ -9,7 +9,7 @@ the audio subsystem with the kernel as a platform device and is represented by the following struct:- /* SoC machine */ -struct snd_soc_card {_ +struct snd_soc_card { char *name; int (*probe)(struct platform_device *pdev); -- 1.5.6.5

2 1

[alsa-devel] [PATCH 0/10] ASoC updates
by Mark Brown 21 Nov '08

21 Nov '08

The following changes since commit 9b0db7e7fd20d5a38844e9435f7d4246ea44978a: Mark Brown (1): ASoC: Convert blackfin machines to use DAI accessor functions are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Jarkko Nikula (1): ASoC: OMAP: Fix preprocessor filled DAI name in McBSP DAI Karl Beldan (2): ASoC: ssm2602: Fix priv substreams refs ASoC: ssm2602: Update supported stream formats Mark Brown (7): ASoC: Rename snd_soc_card to snd_soc_machine ASoC: Add abbreviation to maintainers for searchability ASoC: Move DAI structure definitions into new soc-dai.h ASoC: Merge snd_soc_ops into snd_soc_dai_ops ASoC: Add Marvell Zylonite machine support ASoC: Staticise pxa2xx_pcm_ops ASoC: Improve error reporting for AC97 reset failures Documentation/sound/alsa/soc/machine.txt | 8 +- MAINTAINERS | 2 +- include/sound/soc-dai.h | 225 ++++++++++++++++++++++++++++++ include/sound/soc.h | 156 +-------------------- sound/soc/atmel/atmel_ssc_dai.c | 21 ++-- sound/soc/atmel/playpaq_wm8510.c | 4 +- sound/soc/atmel/sam9g20_wm8731.c | 4 +- sound/soc/au1x/psc-ac97.c | 5 +- sound/soc/au1x/psc-i2s.c | 8 +- sound/soc/au1x/sample-ac97.c | 4 +- sound/soc/blackfin/bf5xx-ad1980.c | 6 +- sound/soc/blackfin/bf5xx-ad73311.c | 6 +- sound/soc/blackfin/bf5xx-i2s.c | 12 +- sound/soc/blackfin/bf5xx-ssm2602.c | 6 +- sound/soc/codecs/ac97.c | 3 +- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/ak4535.c | 5 +- sound/soc/codecs/cs4270.c | 11 +- sound/soc/codecs/ssm2602.c | 39 ++++- sound/soc/codecs/tlv320aic23.c | 19 ++-- sound/soc/codecs/tlv320aic26.c | 5 +- sound/soc/codecs/tlv320aic3x.c | 5 +- sound/soc/codecs/twl4030.c | 5 +- sound/soc/codecs/uda134x.c | 12 +- sound/soc/codecs/uda1380.c | 15 +-- sound/soc/codecs/wm8510.c | 5 +- sound/soc/codecs/wm8580.c | 12 +- sound/soc/codecs/wm8728.c | 5 +- sound/soc/codecs/wm8731.c | 11 +- sound/soc/codecs/wm8750.c | 5 +- sound/soc/codecs/wm8753.c | 25 ++-- sound/soc/codecs/wm8900.c | 5 +- sound/soc/codecs/wm8903.c | 11 +- sound/soc/codecs/wm8971.c | 5 +- sound/soc/codecs/wm8990.c | 6 +- sound/soc/codecs/wm9712.c | 8 +- sound/soc/codecs/wm9713.c | 23 ++-- sound/soc/davinci/davinci-evm.c | 4 +- sound/soc/davinci/davinci-i2s.c | 20 ++-- sound/soc/davinci/davinci-sffsdr.c | 7 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_ssi.c | 14 +- sound/soc/fsl/mpc5200_psc_i2s.c | 17 ++- sound/soc/fsl/mpc8610_hpcd.c | 6 +- sound/soc/fsl/soc-of-simple.c | 10 +- sound/soc/omap/n810.c | 4 +- sound/soc/omap/omap-mcbsp.c | 16 ++- sound/soc/omap/omap2evm.c | 7 +- sound/soc/omap/omap3beagle.c | 4 +- sound/soc/omap/osk5912.c | 4 +- sound/soc/omap/overo.c | 4 +- sound/soc/pxa/Makefile | 2 + sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/e800_wm9712.c | 6 +- sound/soc/pxa/em-x270.c | 4 +- sound/soc/pxa/palm27x.c | 4 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/pxa-ssp.c | 20 +-- sound/soc/pxa/pxa2xx-ac97.c | 9 +- sound/soc/pxa/pxa2xx-i2s.c | 15 ++- sound/soc/pxa/pxa2xx-pcm.c | 2 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/tosa.c | 6 +- sound/soc/pxa/zylonite.c | 219 +++++++++++++++++++++++++++++ sound/soc/s3c24xx/ln2440sbc_alc650.c | 6 +- sound/soc/s3c24xx/neo1973_wm8753.c | 6 +- sound/soc/s3c24xx/s3c2412-i2s.c | 8 +- sound/soc/s3c24xx/s3c2443-ac97.c | 8 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 9 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 4 +- sound/soc/s3c24xx/smdk2443_wm9710.c | 6 +- sound/soc/sh/hac.c | 3 +- sound/soc/sh/sh7760-ac97.c | 4 +- sound/soc/sh/ssi.c | 16 +- sound/soc/soc-core.c | 164 +++++++++++----------- sound/soc/soc-dapm.c | 6 +- 76 files changed, 851 insertions(+), 516 deletions(-) create mode 100644 include/sound/soc-dai.h create mode 100644 sound/soc/pxa/zylonite.c

3 14

[alsa-devel] [PATCH 0/1] ASoC: OMAP: Fix preprocessor filled DAI name in McBSP DAI
by Jarkko Nikula 21 Nov '08

21 Nov '08

IMO, this is a minor fix, so generated on top of topic/asoc. -- Jarkko

2 2

[alsa-devel] Fixes and misc improvements for alsa-lib, alsa-plugins, alsa-utils
by flameeyes＠gmail.com 21 Nov '08

21 Nov '08

I've taken some time to fix some autotools nitpicks and some generic optimisations, I hope they can be useful: git://git.flameeyes.eu/others/alsa/alsa-lib.git flameeyes Diego E. 'Flameeyes' Pettenò (5): Don't use AC_CANONICAL_SYSTEM, only use AC_CANONICAL_HOST. Add the attributes.m4 macro file from xine/lscube. Check for --no-undefined linker flag and use it. Make sure that python libraries are passed through LIBADD. Make some static tables and strings constants. git://git.flameeyes.eu/others/alsa/alsa-plugins.git flameeyes Diego E. 'Flameeyes' Pettenò (6): Add the attributes.m4 macro file from xine/lscube. Check for --no-undefined linker flag and use it. Cleanup flags in maemo/Makefile.am [RFC] Don't use pow() for calculating a power of 2, use shift instead. Make some static tables and strings constants. Mark as static the functions not used outside their unit. git://git.flameeyes.eu/others/alsa/alsa-utils.git flameeyes Diego E. 'Flameeyes' Pettenò (4): Remove some dead code (comparisons between 0 and unsigned integers). Remove some unused variables. Mark static the functions not used outside their unit. Make some static tables and strings constants. -- Diego "Flameeyes" Pettenò http://blog.flameeyes.eu/

2 5

[alsa-devel] [PATCH] ARM DaVinci: Add PCM3008 ALSA SoC driver for the Lyrtech SFFSDR board
by Hugo Villeneuve 21 Nov '08

21 Nov '08

The PCM3008 is used on the Lyrtech SFFSDR board, in conjunction with an FPGA that generates the bit clock and the master clock Signed-off-by: Hugo Villeneuve <hugo.villeneuve(a)lyrtech.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm3008.c | 223 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/pcm3008.h | 25 ++++ sound/soc/davinci/Kconfig | 9 ++ sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-sffsdr.c | 145 +++++++++++++++++++++++ 7 files changed, 410 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/pcm3008.c create mode 100644 sound/soc/codecs/pcm3008.h create mode 100644 sound/soc/davinci/davinci-sffsdr.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5df7402..dc58ce2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -80,6 +80,10 @@ config SND_SOC_TWL4030 tristate depends on TWL4030_CORE +config SND_SOC_PCM3008 + tristate + depends on SFFSDR_FPGA + config SND_SOC_UDA1380 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3b9b58a..7af88e7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,7 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o +snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o @@ -26,6 +27,7 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o +obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c new file mode 100644 index 0000000..20d8b94 --- /dev/null +++ b/sound/soc/codecs/pcm3008.c @@ -0,0 +1,223 @@ +/* + * ALSA Soc PCM3008 codec support + * + * Author: Hugo Villeneuve + * Copyright (C) 2008 Lyrtech inc + * + * Based on AC97 Soc codec, original copyright follow: + * + * Copyright 2005 Wolfson Microelectronics PLC. + * + * 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. + * + * Generic PCM3008 support. + */ + +#include <linux/init.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/gpio.h> +#include <asm/plat-sffsdr/sffsdr-fpga.h> + +#include "pcm3008.h" + +#define PCM3008_VERSION "0.1" + +#define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +static int pcm3008_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int fs; + + /* Fsref can be 32000, 44100 or 48000. */ + fs = params_rate(params); + + printk(KERN_INFO "pcm3008_hw_params: rate = %d Hz\n", fs); + + return sffsdr_fpga_set_codec_fs(fs); +} + +struct snd_soc_dai pcm3008_dai = { + .name = "PCM3008 HiFi", + .type = SND_SOC_DAI_I2S, + .playback = { + .stream_name = "PCM3008 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = PCM3008_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "PCM3008 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = PCM3008_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .hw_params = pcm3008_hw_params, + }, +}; +EXPORT_SYMBOL_GPL(pcm3008_dai); + +static void pcm3008_gpio_free(struct pcm3008_setup_data *setup) +{ + gpio_free(setup->dem0_pin); + gpio_free(setup->dem1_pin); + gpio_free(setup->pdad_pin); + gpio_free(setup->pdda_pin); +} + +static int pcm3008_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct pcm3008_setup_data *setup = socdev->codec_data; + int ret = 0; + + printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (!socdev->codec) + return -ENOMEM; + + codec = socdev->codec; + mutex_init(&codec->mutex); + + codec->name = "PCM3008"; + codec->owner = THIS_MODULE; + codec->dai = &pcm3008_dai; + codec->num_dai = 1; + codec->write = NULL; + codec->read = NULL; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + /* Register PCMs. */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "pcm3008: failed to create pcms\n"); + goto pcm_err; + } + + /* Register Card. */ + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "pcm3008: failed to register card\n"); + goto card_err; + } + + /* DEM1 DEM0 DE-EMPHASIS_MODE + * Low Low De-emphasis 44.1 kHz ON + * Low High De-emphasis OFF + * High Low De-emphasis 48 kHz ON + * High High De-emphasis 32 kHz ON + */ + + /* Configure DEM0 GPIO (turning OFF DAC De-emphasis). */ + ret = gpio_request(setup->dem0_pin, "codec_dem0"); + if (ret == 0) + ret = gpio_direction_output(setup->dem0_pin, 1); + if (ret != 0) + goto gpio_err; + + /* Configure DEM1 GPIO (turning OFF DAC De-emphasis). */ + ret = gpio_request(setup->dem1_pin, "codec_dem1"); + if (ret == 0) + ret = gpio_direction_output(setup->dem1_pin, 0); + if (ret != 0) + goto gpio_err; + + /* Configure PDAD GPIO. */ + ret = gpio_request(setup->pdad_pin, "codec_pdad"); + if (ret == 0) + ret = gpio_direction_output(setup->pdad_pin, 1); + if (ret != 0) + goto gpio_err; + + /* Configure PDDA GPIO. */ + ret = gpio_request(setup->pdda_pin, "codec_pdda"); + if (ret == 0) + ret = gpio_direction_output(setup->pdda_pin, 1); + if (ret != 0) + goto gpio_err; + + return ret; + +gpio_err: + pcm3008_gpio_free(setup); +card_err: + snd_soc_free_pcms(socdev); +pcm_err: + kfree(socdev->codec); + + return ret; +} + +static int pcm3008_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct pcm3008_setup_data *setup = socdev->codec_data; + + if (!codec) + return 0; + + pcm3008_gpio_free(setup); + snd_soc_free_pcms(socdev); + kfree(socdev->codec); + + return 0; +} + +#ifdef CONFIG_PM +static int pcm3008_soc_suspend(struct platform_device *pdev, pm_message_t msg) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct pcm3008_setup_data *setup = socdev->codec_data; + + printk(KERN_INFO "pcm3008_soc_suspend(): TODO\n"); + + gpio_set_value(setup->pdad_pin, 0); + gpio_set_value(setup->pdda_pin, 0); + + return 0; +} + +static int pcm3008_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct pcm3008_setup_data *setup = socdev->codec_data; + + printk(KERN_INFO "pcm3008_soc_resume(): TODO\n"); + + gpio_set_value(setup->pdad_pin, 1); + gpio_set_value(setup->pdda_pin, 1); + + return 0; +} +#else +#define pcm3008_soc_suspend NULL +#define pcm3008_soc_resume NULL +#endif + +struct snd_soc_codec_device soc_codec_dev_pcm3008 = { + .probe = pcm3008_soc_probe, + .remove = pcm3008_soc_remove, + .suspend = pcm3008_soc_suspend, + .resume = pcm3008_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_pcm3008); + +MODULE_DESCRIPTION("Soc PCM3008 driver"); +MODULE_AUTHOR("Hugo Villeneuve"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/pcm3008.h b/sound/soc/codecs/pcm3008.h new file mode 100644 index 0000000..81c82c9 --- /dev/null +++ b/sound/soc/codecs/pcm3008.h @@ -0,0 +1,25 @@ +/* + * PCM3008 ALSA SoC Layer + * + * Author: Hugo Villeneuve + * Copyright (C) 2008 Lyrtech inc + * + * 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 __LINUX_SND_SOC_PCM3008_H +#define __LINUX_SND_SOC_PCM3008_H + +struct pcm3008_setup_data { + u8 dem0_pin; + u8 dem1_pin; + u8 pdad_pin; + u8 pdda_pin; +}; + +extern struct snd_soc_codec_device soc_codec_dev_pcm3008; +extern struct snd_soc_dai pcm3008_dai; + +#endif diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 8f7e338..7978b81 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -17,3 +17,12 @@ config SND_DAVINCI_SOC_EVM help Say Y if you want to add support for SoC audio on TI DaVinci EVM platform. + +config SND_DAVINCI_SOC_SFFSDR + tristate "SoC Audio support for SFFSDR" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR + select SND_DAVINCI_SOC_I2S + select SND_SOC_PCM3008 + help + Say Y if you want to add support for SoC audio on + Lyrtech SFFSDR board. diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index ca772e5..ca8bae1 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -7,5 +7,7 @@ obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o +snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c new file mode 100644 index 0000000..b232388 --- /dev/null +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -0,0 +1,145 @@ +/* + * ASoC driver for Lyrtech SFFSDR board. + * + * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.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/gpio.h> +#include <asm/dma.h> +#include <mach/mcbsp.h> +#include <mach/edma.h> + +#include "../codecs/pcm3008.h" +#include "davinci-pcm.h" +#include "davinci-i2s.h" + +static int sffsdr_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 *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + /* Set cpu DAI configuration: + * CLKX and CLKR are the inputs for the Sample Rate Generator. + * FSX and FSR are outputs, driven by the sample Rate Generator. */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_RIGHT_J | + SND_SOC_DAIFMT_CBM_CFS | + SND_SOC_DAIFMT_IB_NF); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops sffsdr_ops = { + .hw_params = sffsdr_hw_params, +}; + +/* davinci-sffsdr digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sffsdr_dai = { + .name = "PCM3008", /* Codec name */ + .stream_name = "PCM3008 HiFi", + .cpu_dai = &davinci_i2s_dai, + .codec_dai = &pcm3008_dai, + .ops = &sffsdr_ops, +}; + +/* davinci-sffsdr audio machine driver */ +static struct snd_soc_machine snd_soc_machine_sffsdr = { + .name = "DaVinci SFFSDR", + .dai_link = &sffsdr_dai, + .num_links = 1, +}; + +/* sffsdr audio private data */ +static struct pcm3008_setup_data sffsdr_pcm3008_setup = { + .dem0_pin = GPIO(45), + .dem1_pin = GPIO(46), + .pdad_pin = GPIO(47), + .pdda_pin = GPIO(38), +}; + +/* sffsdr audio subsystem */ +static struct snd_soc_device sffsdr_snd_devdata = { + .machine = &snd_soc_machine_sffsdr, + .platform = &davinci_soc_platform, + .codec_dev = &soc_codec_dev_pcm3008, + .codec_data = &sffsdr_pcm3008_setup, +}; + +static struct resource sffsdr_snd_resources[] = { + { + .start = DAVINCI_MCBSP_BASE, + .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data sffsdr_snd_data = { + .tx_dma_ch = DAVINCI_DMA_MCBSP_TX, + .rx_dma_ch = DAVINCI_DMA_MCBSP_RX, +}; + +static struct platform_device *sffsdr_snd_device; + +static int __init sffsdr_init(void) +{ + int ret; + + sffsdr_snd_device = platform_device_alloc("soc-audio", 0); + if (!sffsdr_snd_device) { + printk(KERN_ERR "platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(sffsdr_snd_device, &sffsdr_snd_devdata); + sffsdr_snd_devdata.dev = &sffsdr_snd_device->dev; + sffsdr_snd_device->dev.platform_data = &sffsdr_snd_data; + + ret = platform_device_add_resources(sffsdr_snd_device, + sffsdr_snd_resources, + ARRAY_SIZE(sffsdr_snd_resources)); + if (ret) { + printk(KERN_ERR "platform device add ressources failed\n"); + goto error; + } + + ret = platform_device_add(sffsdr_snd_device); + if (ret) + goto error; + + return ret; + +error: + platform_device_put(sffsdr_snd_device); + return ret; +} + +static void __exit sffsdr_exit(void) +{ + platform_device_unregister(sffsdr_snd_device); +} + +module_init(sffsdr_init); +module_exit(sffsdr_exit); + +MODULE_AUTHOR("Hugo Villeneuve"); +MODULE_DESCRIPTION("Lyrtech SFFSDR ASoC driver"); +MODULE_LICENSE("GPL"); -- 1.5.5

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