January 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 3/4] ASoC sst: Add mid machine driver
by Vinod Koul 04 Jan '11

04 Jan '11

This patch adds the mic machine driver The mid machine driver glues the msic and mid_platfrom driver to form the asoc sound driver Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/mid-x86/mid_machine.c | 317 +++++++++++++++++++++++++++++++++++ sound/soc/mid-x86/mid_machine.h | 37 ++++ sound/soc/mid-x86/mid_machine_lib.c | 54 ++++++ 3 files changed, 408 insertions(+), 0 deletions(-) create mode 100644 sound/soc/mid-x86/mid_machine.c create mode 100644 sound/soc/mid-x86/mid_machine.h create mode 100644 sound/soc/mid-x86/mid_machine_lib.c diff --git a/sound/soc/mid-x86/mid_machine.c b/sound/soc/mid-x86/mid_machine.c new file mode 100644 index 0000000..36786ce --- /dev/null +++ b/sound/soc/mid-x86/mid_machine.c @@ -0,0 +1,317 @@ +/* + * mid_machine.c - Intel MID Machine driver + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul <vinod.koul(a)intel.com> + * Author: Harsha Priya <priya.harsha(a)intel.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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/init.h> +#include <linux/device.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../codecs/msic.h" +#include "mid_machine.h" + +struct mid_mfld_device { + struct platform_device *mid_platform_device; + struct platform_device *mid_codec_device; + struct platform_device *mid_soc_device; +}; + +static unsigned int hs_switch; +static unsigned int lo_dac; + +/*sound card controls*/ +static const char *headset_switch_text[] = {"Earpiece", "Headset"}; + +static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"}; + +static const struct soc_enum headset_enum = + SOC_ENUM_SINGLE_EXT(2, headset_switch_text); + +static const struct soc_enum lo_enum = + SOC_ENUM_SINGLE_EXT(4, lo_text); + +static int headset_get_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = hs_switch; + return 0; +} + +static int headset_set_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] == hs_switch) + return 0; + + if (ucontrol->value.integer.value[0]) { + pr_debug("hs_set HS path\n"); + snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL"); + snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); + } else { + pr_debug("hs_set EP path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); + snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); + } + snd_soc_dapm_sync(&codec->dapm); + hs_switch = ucontrol->value.integer.value[0]; + + return 0; +} + +static void lo_enable_out_pins(struct snd_soc_codec *codec) +{ + snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTL"); + snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTL"); + snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT"); + snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT"); + if (hs_switch) { + snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL"); + snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); + } else { + snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); + snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); + } +} + +static int lo_get_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = lo_dac; + return 0; +} + +static int lo_set_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] == lo_dac) + return 0; + + /* + * we dont want to work with last state of lineout so just enable all + * pins and then disable pins not required + */ + lo_enable_out_pins(codec); + switch (ucontrol->value.integer.value[0]) { + case 0: + pr_debug("set vibra path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "VIB1OUT"); + snd_soc_dapm_disable_pin(&codec->dapm, "VIB2OUT"); + snd_soc_update_bits(codec, MSIC_LOCTL, 0x66, 0); + break; + + case 1: + pr_debug("set hs path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); + snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); + snd_soc_update_bits(codec, MSIC_LOCTL, 0x66, 0x22); + break; + + case 2: + pr_debug("set spkr path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTL"); + snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTR"); + snd_soc_update_bits(codec, MSIC_LOCTL, 0x66, 0x44); + break; + + case 3: + pr_debug("set null path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTL"); + snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTR"); + snd_soc_update_bits(codec, MSIC_LOCTL, 0x66, 0x66); + break; + } + snd_soc_dapm_sync(&codec->dapm); + lo_dac = ucontrol->value.integer.value[0]; + return 0; +} + +static const struct snd_kcontrol_new mid_snd_controls[] = { + SOC_ENUM_EXT("Playback Switch", headset_enum, + headset_get_switch, headset_set_switch), + SOC_ENUM_EXT("Lineout Mux", lo_enum, + lo_get_switch, lo_set_switch), +}; + +static int msic_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret_val; + + ret_val = snd_soc_add_controls(codec, mid_snd_controls, + ARRAY_SIZE(mid_snd_controls)); + if (ret_val) { + pr_err("soc_add_controls failed %d", ret_val); + return ret_val; + } + /*default is earpiece pin, userspace sets it explcitly*/ + snd_soc_dapm_disable_pin(dapm, "HPOUTL"); + snd_soc_dapm_disable_pin(dapm, "HPOUTR"); + /*default is lineout NC, userspace sets it explcitly*/ + snd_soc_dapm_disable_pin(dapm, "LINEOUTL"); + snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); + return snd_soc_dapm_sync(dapm); + lo_dac = 3; + hs_switch = 0; +} + +struct snd_soc_dai_link mfld_msic_dailink[] = { + { + .name = "Intel MSIC Headset", + .stream_name = "Codec", + .cpu_dai_name = "Headset-cpu-dai", + .codec_dai_name = "Intel MSIC Headset", + .codec_name = "mid-msic-codec", + .platform_name = "mid-audio-platform", + .init = msic_init, + }, + { + .name = "Intel MSIC Speaker", + .stream_name = "Speaker", + .cpu_dai_name = "Speaker-cpu-dai", + .codec_dai_name = "Intel MSIC Speaker", + .codec_name = "mid-msic-codec", + .platform_name = "mid-audio-platform", + .init = NULL, + }, + { + .name = "Intel MSIC Vibra1", + .stream_name = "Vibra1", + .cpu_dai_name = "Vibra 1-cpu-dai", + .codec_dai_name = "Intel MSIC Vibra1", + .codec_name = "mid-msic-codec", + .platform_name = "mid-audio-platform", + .init = NULL, + }, + { + .name = "Intel MSIC Vibra2", + .stream_name = "Vibra2", + .cpu_dai_name = "Vibra 2-cpu-dai", + .codec_dai_name = "Intel MSIC Vibra2", + .codec_name = "mid-msic-codec", + .platform_name = "mid-audio-platform", + .init = NULL, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_mfld = { + .name = "mid_msic_audio", + .dai_link = mfld_msic_dailink, + .num_links = ARRAY_SIZE(mfld_msic_dailink), +}; + +int __devinit snd_intelmid_mc_probe(struct platform_device *pdev) +{ + struct mid_mfld_device *dev_data; + int ret_val = 0; + + pr_debug("snd_intelmid_mc_probe called\n"); + + dev_data = kzalloc(sizeof(*dev_data), GFP_KERNEL); + if (!dev_data) { + pr_err("snd_intelmid_mc_probe mem alloc fail\n"); + return -ENOMEM; + } + + ret_val = create_device(&dev_data->mid_platform_device, + "mid-audio-platform", NULL); + if (ret_val) + return ret_val; + + ret_val = create_device(&dev_data->mid_codec_device, + "mid-msic-codec", NULL); + if (ret_val) + goto unreg_platform; + + ret_val = create_device(&dev_data->mid_soc_device, + "soc-audio", &snd_soc_card_mfld); + if (ret_val) + goto unreg_codec; + + platform_set_drvdata(pdev, dev_data); + + pr_debug("successfully exited probe\n"); + return ret_val; +unreg_codec: + platform_device_unregister(dev_data->mid_codec_device); +unreg_platform: + platform_device_unregister(dev_data->mid_platform_device); + return ret_val; +} + +static int snd_intelmid_remove(struct platform_device *pdev) +{ + struct mid_mfld_device *dev_data = platform_get_drvdata(pdev); + pr_debug("snd_intelmid_remove called\n"); + + platform_device_unregister(dev_data->mid_soc_device); + platform_device_unregister(dev_data->mid_platform_device); + platform_device_unregister(dev_data->mid_codec_device); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_intelmid_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "msic_audio", + }, + .probe = snd_intelmid_mc_probe, + .remove = __devexit_p(snd_intelmid_remove), +}; + +static int __init intelmidmc_soc_init(void) +{ + pr_debug("intelmidmc_soc_init called\n"); + return platform_driver_register(&snd_intelmid_mc_driver); +} +module_init(intelmidmc_soc_init); + +static void __exit intelmidmc_soc_exit(void) +{ + pr_debug("intelmidmc_soc_exit called\n"); + platform_driver_unregister(&snd_intelmid_mc_driver); +} +module_exit(intelmidmc_soc_exit); + +MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver"); +MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); +MODULE_AUTHOR("Harsha Priya <priya.harsha(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("mid-machine"); diff --git a/sound/soc/mid-x86/mid_machine.h b/sound/soc/mid-x86/mid_machine.h new file mode 100644 index 0000000..34dbbdd --- /dev/null +++ b/sound/soc/mid-x86/mid_machine.h @@ -0,0 +1,37 @@ +/* + * mid_machine.h - Intel MID Machine driver header file + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul <vinod.koul(a)intel.com> + * Author: Harsha Priya <priya.harsha(a)intel.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; version 2 of the License. + * + * 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 __MID_MACHINEDRV_H__ +#define __MID_MACHINEDRV_H__ + +#define MID_MONO 1 +#define MID_STEREO 2 +#define MID_MAX_CAP 5 + +int create_device(struct platform_device **device, char *name, + void *drv_data); +#endif + diff --git a/sound/soc/mid-x86/mid_machine_lib.c b/sound/soc/mid-x86/mid_machine_lib.c new file mode 100644 index 0000000..fce2861 --- /dev/null +++ b/sound/soc/mid-x86/mid_machine_lib.c @@ -0,0 +1,54 @@ +/* + * mid_machine_lib.c - Intel Machine driver library functions + * should be reused in other mid machine drivers + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul <vinod.koul(a)intel.com> + * Author: Harsha Priya <priya.harsha(a)intel.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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/platform_device.h> + +int create_device(struct platform_device **device, char *name, + void *drv_data) +{ + int ret_val = 0; + + pr_debug("%s device creation...", name); + + *device = platform_device_alloc(name, -1); + if (!*device) { + pr_err("%s device allocation failed\n", name); + return -ENOMEM; + } + + if (drv_data) + platform_set_drvdata(*device, drv_data); + + ret_val = platform_device_add(*device); + if (ret_val) { + pr_err("Unable to add platform device\n"); + platform_device_put(*device); + } + pr_debug("%s...created\n", name); + return ret_val; +} -- 1.7.2.3

3 8

[PATCH] ALSA: HDA: Add internal mic for IDT 92HD88B
by David Henningsson 04 Jan '11

04 Jan '11

BugLink: http://bugs.launchpad.net/bugs/696493 According to datasheet (and real-world testing), IDT 92HD88B can have internal mics at NID 0x11 and 0x20, so enable them accordingly. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 12 +++++++++++- 1 files changed, 11 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6131a92..c8d812e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -389,6 +389,9 @@ static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = { 0x11, 0x20, 0 }; +#define STAC92HD88XXX_NUM_DMICS STAC92HD83XXX_NUM_DMICS +#define stac92hd88xxx_dmic_nids stac92hd83xxx_dmic_nids + #define STAC92HD87B_NUM_DMICS 1 static hda_nid_t stac92hd87b_dmic_nids[STAC92HD87B_NUM_DMICS + 1] = { 0x11, 0 @@ -5462,11 +5465,18 @@ again: spec->num_dmics = stac92xx_connected_ports(codec, stac92hd87b_dmic_nids, STAC92HD87B_NUM_DMICS); - /* Fall through */ + spec->num_pins = ARRAY_SIZE(stac92hd88xxx_pin_nids); + spec->pin_nids = stac92hd88xxx_pin_nids; + spec->mono_nid = 0; + spec->num_pwrs = 0; + break; case 0x111d7666: case 0x111d7667: case 0x111d7668: case 0x111d7669: + spec->num_dmics = stac92xx_connected_ports(codec, + stac92hd88xxx_dmic_nids, + STAC92HD88XXX_NUM_DMICS); spec->num_pins = ARRAY_SIZE(stac92hd88xxx_pin_nids); spec->pin_nids = stac92hd88xxx_pin_nids; spec->mono_nid = 0; -- 1.7.1 --------------050904050906070104050107--

1 0

[alsa-devel] [PATCH] ALSA: lib: Fix static linking of libasound
by Daniel Chen 03 Jan '11

03 Jan '11

Hi, Attached is a patch from Samuel Thibault that fixes static linking of libasound.a (forwarded from Debian bug report #522544). Thanks, -Dan

4 6

[alsa-devel] [PATCH 1/4] ASoC sst: Add msic codec driver
by Vinod Koul 03 Jan '11

03 Jan '11

This patch adds the msic asoc codec driver. This driver currently supports only playback. Capture and jack detection to be added later Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/msic.c | 490 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/msic.h | 99 +++++++++ 4 files changed, 595 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/msic.c create mode 100644 sound/soc/codecs/msic.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 054191e..5f800da 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -31,6 +31,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_MSIC if INTEL_SCU_IPC select SND_SOC_PCM3008 select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C @@ -170,6 +171,9 @@ config SND_SOC_DA7210 config SND_SOC_MAX98088 tristate +config SND_SOC_MSIC + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6a1e17b..5e57586 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -16,6 +16,7 @@ snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o +snd-soc-msic-objs := msic.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-alc5623-objs := alc5623.o snd-soc-spdif-objs := spdif_transciever.o @@ -95,6 +96,7 @@ obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o +obj-$(CONFIG_SND_SOC_MSIC) +=snd-soc-msic.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o diff --git a/sound/soc/codecs/msic.c b/sound/soc/codecs/msic.c new file mode 100644 index 0000000..66d9bda --- /dev/null +++ b/sound/soc/codecs/msic.c @@ -0,0 +1,490 @@ +/* + * msic.c - Intel MSIC Codec driver + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul <vinod.koul(a)intel.com> + * Author: Harsha Priya <priya.harsha(a)intel.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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <asm/intel_scu_ipc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include "msic.h" + +#define MSIC_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) +#define MSIC_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) + +/* + * todo: + * capture paths + * jack detection + * PM functions + */ + +static inline unsigned int msic_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value = 0; + int ret; + + ret = intel_scu_ipc_ioread8(reg, &value); + if (ret) + pr_err("read of %x failed, err %d\n", reg, ret); + pr_debug("read for %x ret %x\n", reg, value); + return value; + +} + +static inline int msic_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + int ret; + + pr_debug("write for %x with %x\n", reg, value); + ret = intel_scu_ipc_iowrite8(reg, value); + if (ret) + pr_err("write of %x failed, err %d\n", reg, ret); + return ret; +} + +static int msic_set_vaud_bias(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + pr_debug("msic_set_vaud_bias %d\n", level); + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + pr_debug("vaud_bias _PREPARE\n"); + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { + pr_debug("vaud_bias powering up pll\n"); + /*power up the pll*/ + msic_write(codec, MSIC_AUDPLLCTRL, BIT(5)); + /*enable pcm 2*/ + snd_soc_update_bits(codec, MSIC_PCM2C2, BIT(0), BIT(0)); + } + break; + + case SND_SOC_BIAS_STANDBY: + pr_debug("vaud_bias _STANDBY\n"); + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + pr_debug("vaud_bias power up rail\n"); + /*power up the rail*/ + msic_write(codec, MSIC_VAUD, BIT(2)|BIT(1)|BIT(0)); + msleep(1); + } else if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) { + /*turn off pcm*/ + pr_debug("vaud_bias power dn pcm\n"); + snd_soc_update_bits(codec, MSIC_PCM2C2, BIT(0), 0); + msic_write(codec, MSIC_AUDPLLCTRL, 0); + } + break; + + + case SND_SOC_BIAS_OFF: + pr_debug("vaud_bias _OFF doing rail shutdown\n"); + msic_write(codec, MSIC_VAUD, BIT(3)); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +static int msic_vhs_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + pr_debug("msic_vhs_event %d\n", event); + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("VHS SND_SOC_DAPM_EVENT_ON doing rail startup now\n"); + /*power up the rail*/ + msic_write(w->codec, MSIC_VHSP, 0x3D); + msic_write(w->codec, MSIC_VHSN, 0x3F); + msleep(1); + } else if (SND_SOC_DAPM_EVENT_OFF(event)) { + pr_debug("VHS SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n"); + msic_write(w->codec, MSIC_VHSP, 0xC4); + msic_write(w->codec, MSIC_VHSN, 0x04); + } + return 0; +} + +static int msic_vihf_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + pr_debug("msic_vihf_event %d\n", event); + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("VIHF SND_SOC_DAPM_EVENT_ON doing rail startup now\n"); + /*power up the rail*/ + msic_write(w->codec, MSIC_VIHF, 0x27); + msleep(1); + } else if (SND_SOC_DAPM_EVENT_OFF(event)) { + pr_debug("VIHF SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n"); + msic_write(w->codec, MSIC_VIHF, 0x24); + } + return 0; +} + +/*DAPM widgets*/ +static const struct snd_soc_dapm_widget intel_msic_dapm_widgets[] = { + + /*all end points mic, hs etc*/ + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("EPOUT"), + SND_SOC_DAPM_OUTPUT("IHFOUTL"), + SND_SOC_DAPM_OUTPUT("IHFOUTR"), + SND_SOC_DAPM_OUTPUT("LINEOUTL"), + SND_SOC_DAPM_OUTPUT("LINEOUTR"), + SND_SOC_DAPM_OUTPUT("VIB1OUT"), + SND_SOC_DAPM_OUTPUT("VIB2OUT"), + + SND_SOC_DAPM_SUPPLY("Headset Rail", SND_SOC_NOPM, 0, 0, + msic_vhs_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Speaker Rail", SND_SOC_NOPM, 0, 0, + msic_vihf_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /*playback path driver enables*/ + SND_SOC_DAPM_PGA("Headset Left Playback", + MSIC_DRIVEREN, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headset Right Playback", + MSIC_DRIVEREN, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker Left Playback", + MSIC_DRIVEREN, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker Right Playback", + MSIC_DRIVEREN, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Vibra1 Playback", + MSIC_DRIVEREN, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Vibra2 Playback", + MSIC_DRIVEREN, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Earpiece Playback", + MSIC_DRIVEREN, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Lineout Left Playback", + MSIC_LOCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Lineout Right Playback", + MSIC_LOCTL, 4, 0, NULL, 0), + + /*playback path filter enable*/ + SND_SOC_DAPM_PGA("Headset Left Filter", + MSIC_HSEPRXCTRL, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headset Right Filter", + MSIC_HSEPRXCTRL, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker Left Filter", + MSIC_IHFRXCTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker Right Filter", + MSIC_IHFRXCTRL, 1, 0, NULL, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset", + MSIC_DACCONFIG, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset", + MSIC_DACCONFIG, 1, 0), + SND_SOC_DAPM_DAC("IHFDAC Left", "Speaker", + MSIC_DACCONFIG, 2, 0), + SND_SOC_DAPM_DAC("IHFDAC Right", "Speaker", + MSIC_DACCONFIG, 3, 0), + SND_SOC_DAPM_DAC("Vibra1 DAC", "Vibra1", + MSIC_VIB1C5, 1, 0), + SND_SOC_DAPM_DAC("Vibra2 DAC", "Vibra2", + MSIC_VIB2C5, 1, 0), +}; + +static const struct snd_soc_dapm_route msic_audio_map[] = { + /*headset and earpiece map*/ + { "HPOUTL", NULL, "Headset Left Playback" }, + { "HPOUTR", NULL, "Headset Right Playback" }, + { "EPOUT", NULL, "Earpiece Playback" }, + { "Headset Left Playback", NULL, "Headset Left Filter"}, + { "Headset Right Playback", NULL, "Headset Right Filter"}, + { "Earpiece Playback", NULL, "Headset Left Filter"}, + { "Headset Left Filter", NULL, "HSDAC Left"}, + { "Headset Right Filter", NULL, "HSDAC Right"}, + { "HSDAC Left", NULL, "Headset Rail"}, + { "HSDAC Right", NULL, "Headset Rail"}, + + /*speaker map*/ + { "IHFOUTL", "NULL", "Speaker Left Playback"}, + { "IHFOUTR", "NULL", "Speaker Right Playback"}, + { "Speaker Left Playback", NULL, "Speaker Left Filter"}, + { "Speaker Right Playback", NULL, "Speaker Right Filter"}, + { "Speaker Left Filter", NULL, "IHFDAC Left"}, + { "Speaker Right Filter", NULL, "IHFDAC Right"}, + { "IHFDAC Left", NULL, "Speaker Rail"}, + { "IHFDAC Right", NULL, "Speaker Rail"}, + + /*vibra map*/ + {"VIB1OUT", NULL, "Vibra1 Playback"}, + {"Vibra1 Playback", NULL, "Vibra1 DAC"}, + + {"VIB2OUT", NULL, "Vibra2 Playback"}, + {"Vibra2 Playback", NULL, "Vibra2 DAC"}, + + /*lineout*/ + {"LINEOUTL", NULL, "Lineout Left Playback"}, + {"LINEOUTR", NULL, "Lineout Right Playback"}, + {"Lineout Left Playback", NULL, "Headset Left Filter"}, + {"Lineout Left Playback", NULL, "Speaker Left Filter"}, + {"Lineout Left Playback", NULL, "Vibra1 DAC"}, + {"Lineout Right Playback", NULL, "Headset Right Filter"}, + {"Lineout Right Playback", NULL, "Speaker Right Filter"}, + {"Lineout Right Playback", NULL, "Vibra2 DAC"}, +}; + +/*speaker and headset mutes, for audio pops and clicks*/ +static int msic_pcm_hs_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, MSIC_HSLVOLCTRL, BIT(7), (!mute << 7)); + snd_soc_update_bits(dai->codec, MSIC_HSRVOLCTRL, BIT(7), (!mute << 7)); + return 0; +} + +static int msic_pcm_spkr_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, MSIC_IHFLVOLCTRL, BIT(7), (!mute << 7)); + snd_soc_update_bits(dai->codec, MSIC_IHFRVOLCTRL, BIT(7), (!mute << 7)); + return 0; +} + +int msic_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + unsigned int format, rate; + + pr_debug("pcm hw params\n"); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + format = 3; + break; + + case SNDRV_PCM_FORMAT_S24_LE: + format = 0; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(dai->codec, MSIC_PCM2C2, BIT(4)|BIT(5), format); + + switch (params_rate(params)) { + case 48000: + pr_debug("RATE_48000\n"); + rate = 0; + break; + + case 44100: + pr_debug("RATE_44100\n"); + rate = 1; + break; + + default: + pr_err("ERR rate %d\n", params_rate(params)); + return -EINVAL; + } + snd_soc_update_bits(dai->codec, MSIC_PCM1C1, BIT(7), rate); + + return 0; +} + +/** Codec DAI section **/ +static struct snd_soc_dai_ops intel_msic_headset_dai_ops = { + .digital_mute = msic_pcm_hs_mute, + .hw_params = msic_pcm_hw_params, +}; + +static struct snd_soc_dai_ops intel_msic_speaker_dai_ops = { + .digital_mute = msic_pcm_spkr_mute, + .hw_params = msic_pcm_hw_params, +}; + +static struct snd_soc_dai_ops intel_msic_vib1_dai_ops = { + .hw_params = msic_pcm_hw_params, +}; + +static struct snd_soc_dai_ops intel_msic_vib2_dai_ops = { + .hw_params = msic_pcm_hw_params, +}; + +struct snd_soc_dai_driver intel_msic_dais[] = { +{ + .name = "Intel MSIC Headset", + .playback = { + .stream_name = "Headset", + .channels_min = 2, + .channels_max = 2, + .rates = MSIC_RATES, + .formats = MSIC_FORMATS, + }, + .ops = &intel_msic_headset_dai_ops, +}, +{ .name = "Intel MSIC Speaker", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = MSIC_RATES, + .formats = MSIC_FORMATS, + }, + .ops = &intel_msic_speaker_dai_ops, +}, +{ .name = "Intel MSIC Vibra", + .playback = { + .stream_name = "Vibra1", + .channels_min = 1, + .channels_max = 1, + .rates = MSIC_RATES, + .formats = MSIC_FORMATS, + }, + .ops = &intel_msic_vib1_dai_ops, +}, +{ .name = "Intel MSIC Haptic", + .playback = { + .stream_name = "Vibra2", + .channels_min = 1, + .channels_max = 1, + .rates = MSIC_RATES, + .formats = MSIC_FORMATS, + }, + .ops = &intel_msic_vib2_dai_ops, +}, +}; + +/** codec registration **/ +static int msic_codec_probe(struct snd_soc_codec *codec) +{ + int ret; + + pr_debug("codec_probe called\n"); + + codec->name = "msic"; + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + codec->dapm.idle_bias_off = 1; + + /*PCM interface */ + msic_write(codec, MSIC_PCM2RXSLOT01, 0x10); + msic_write(codec, MSIC_PCM2RXSLOT23, 0x32); + msic_write(codec, MSIC_PCM2RXSLOT45, 0x54); + /* pcm port setting */ + msic_write(codec, MSIC_PCM1C1, 0x00); + msic_write(codec, MSIC_PCM2C1, 0x01); + msic_write(codec, MSIC_PCM2C2, 0x0A); + msic_write(codec, MSIC_HSMIXER, BIT(0)|BIT(4)); + /*vendor vibra workround, the vibras are muted by + *custom register so unmute them*/ + msic_write(codec, MSIC_SSR5, 0x80); + msic_write(codec, MSIC_SSR6, 0x80); + msic_write(codec, MSIC_VIB1C5, 0x00); + msic_write(codec, MSIC_VIB2C5, 0x00); + /*configure vibras for pcm port*/ + msic_write(codec, MSIC_VIB1C3, 0x00); + msic_write(codec, MSIC_VIB2C3, 0x00); + + /*soft mute ramp time*/ + msic_write(codec, MSIC_SOFTMUTE, 0x3); + /* + * fix the initial volume at 1dB, + * default in +9dB, + * 1dB give optimal swing on DAC, amps + **/ + msic_write(codec, MSIC_HSLVOLCTRL, 0x08); + msic_write(codec, MSIC_HSRVOLCTRL, 0x08); + msic_write(codec, MSIC_IHFLVOLCTRL, 0x08); + msic_write(codec, MSIC_IHFRVOLCTRL, 0x08); + /*dac mode and lineout workaround*/ + msic_write(codec, MSIC_SSR2, 0x10); + msic_write(codec, MSIC_SSR3, 0x40); + + ret = snd_soc_dapm_new_controls(&codec->dapm, intel_msic_dapm_widgets, + ARRAY_SIZE(intel_msic_dapm_widgets)); + if (ret) + pr_err("soc_dapm_new_control failed %d", ret); + ret = snd_soc_dapm_add_routes(&codec->dapm, msic_audio_map, + ARRAY_SIZE(msic_audio_map)); + if (ret) + pr_err("soc_dapm_add_routes failed %d", ret); + + return ret; +} + +static int msic_codec_remove(struct snd_soc_codec *codec) +{ + pr_debug("codec_remove called\n"); + msic_set_vaud_bias(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +struct snd_soc_codec_driver intel_msic_codec = { + .probe = msic_codec_probe, + .remove = msic_codec_remove, + .read = msic_read, + .write = msic_write, + .set_bias_level = msic_set_vaud_bias, +}; + +static int intelmid_codec_probe(struct platform_device *pdev) +{ + pr_debug("codec device probe called for %s\n", dev_name(&pdev->dev)); + return snd_soc_register_codec(&pdev->dev, &intel_msic_codec, + intel_msic_dais, ARRAY_SIZE(intel_msic_dais)); +} + +static int intelmid_codec_remove(struct platform_device *pdev) +{ + pr_debug("codec device remove called\n"); + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver intelmid_codec_driver = { + .driver = { + .name = "mid-msic-codec", + .owner = THIS_MODULE, + }, + .probe = intelmid_codec_probe, + .remove = intelmid_codec_remove, +}; + +static int __init intel_mid_msic_init(void) +{ + pr_debug("driver init called\n"); + return platform_driver_register(&intelmid_codec_driver); +} +module_init(intel_mid_msic_init); + +static void __exit intel_mid_msic_exit(void) +{ + pr_debug("driver exit called\n"); + platform_driver_unregister(&intelmid_codec_driver); +} +module_exit(intel_mid_msic_exit); + +MODULE_DESCRIPTION("ASoC Intel(R) MSIC codec driver"); +MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); +MODULE_AUTHOR("Harsha Priya <priya.harsha(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("msic-codec"); diff --git a/sound/soc/codecs/msic.h b/sound/soc/codecs/msic.h new file mode 100644 index 0000000..6cec236 --- /dev/null +++ b/sound/soc/codecs/msic.h @@ -0,0 +1,99 @@ +/* + * msic.h - Intel MSIC Codec driver + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul <vinod.koul(a)intel.com> + * Author: Harsha Priya <priya.harsha(a)intel.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; version 2 of the License. + * + * 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 _INTEL_MSIC_CODEC_H +#define _INTEL_MSIC_CODEC_H + +/*register map*/ +#define MSIC_VAUD 0xDB +#define MSIC_VHSP 0xDC +#define MSIC_VHSN 0xDD +#define MSIC_VIHF 0xC9 + +#define MSIC_AUDPLLCTRL 0x240 +#define MSIC_MSIC_DMICBUF0123 0x241 +#define MSIC_MSIC_DMICBUF45 0x242 +#define MSIC_MSIC_DMICGPO 0x244 +#define MSIC_DMICMUX 0x245 +#define MSIC_DMICLK 0x246 +#define MSIC_MICBIAS 0x247 +#define MSIC_ADCCONFIG 0x248 +#define MSIC_MICAMP1 0x249 +#define MSIC_MICAMP2 0x24A +#define MSIC_NOISEMUX 0x24B +#define MSIC_AUDIOMUX12 0x24C +#define MSIC_AUDIOMUX34 0x24D +#define MSIC_AUDIOSINC 0x24E +#define MSIC_AUDIOTXEN 0x24F +#define MSIC_HSEPRXCTRL 0x250 +#define MSIC_IHFRXCTRL 0x251 +#define MSIC_HSMIXER 0x256 +#define MSIC_DACCONFIG 0x257 +#define MSIC_SOFTMUTE 0x258 +#define MSIC_HSLVOLCTRL 0x259 +#define MSIC_HSRVOLCTRL 0x25A +#define MSIC_IHFLVOLCTRL 0x25B +#define MSIC_IHFRVOLCTRL 0x25C +#define MSIC_DRIVEREN 0x25D +#define MSIC_LOCTL 0x25E +#define MSIC_VIB1C1 0x25F +#define MSIC_VIB1C2 0x260 +#define MSIC_VIB1C3 0x261 +#define MSIC_VIB1SPIPCM1 0x262 +#define MSIC_VIB1SPIPCM2 0x263 +#define MSIC_VIB1C5 0x264 +#define MSIC_VIB2C1 0x265 +#define MSIC_VIB2C2 0x266 +#define MSIC_VIB2C3 0x267 +#define MSIC_VIB2SPIPCM1 0x268 +#define MSIC_VIB2SPIPCM2 0x269 +#define MSIC_VIB2C5 0x26A +#define MSIC_BTNCTRL1 0x26B +#define MSIC_BTNCTRL2 0x26C +#define MSIC_PCM1TXSLOT01 0x26D +#define MSIC_PCM1TXSLOT23 0x26E +#define MSIC_PCM1TXSLOT45 0x26F +#define MSIC_PCM1RXSLOT0_3 0x270 +#define MSIC_PCM1RXSLOT45 0x271 +#define MSIC_PCM2TXSLOT01 0x272 +#define MSIC_PCM2TXSLOT23 0x273 +#define MSIC_PCM2TXSLOT45 0x274 +#define MSIC_PCM2RXSLOT01 0x275 +#define MSIC_PCM2RXSLOT23 0x276 +#define MSIC_PCM2RXSLOT45 0x277 +#define MSIC_PCM1C1 0x278 +#define MSIC_PCM1C2 0x279 +#define MSIC_PCM1C3 0x27A +#define MSIC_PCM2C1 0x27B +#define MSIC_PCM2C2 0x27C +/*end codec register defn*/ + +/*vendor defn these are not part of avp*/ +#define MSIC_SSR2 0x381 +#define MSIC_SSR3 0x382 +#define MSIC_SSR5 0x384 +#define MSIC_SSR6 0x385 + +#endif -- 1.7.2.3

3 5

[alsa-devel] [PATCH 2/4] ASoC sst: Add mid platform driver
by Vinod Koul 03 Jan '11

03 Jan '11

This patch adds the platform driver for mid asoc drivers This platfrom driver sends commands to sst dsp engine driver for the dai operations. For this purpose it depends on intel_sst driver which is currently in staging tree Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/mid-x86/mid_platform.c | 365 ++++++++++++++++++++++++++++++++++ sound/soc/mid-x86/mid_platform.h | 71 +++++++ sound/soc/mid-x86/mid_platform_lib.c | 122 +++++++++++ 3 files changed, 558 insertions(+), 0 deletions(-) create mode 100644 sound/soc/mid-x86/mid_platform.c create mode 100644 sound/soc/mid-x86/mid_platform.h create mode 100644 sound/soc/mid-x86/mid_platform_lib.c diff --git a/sound/soc/mid-x86/mid_platform.c b/sound/soc/mid-x86/mid_platform.c new file mode 100644 index 0000000..3d26e6e --- /dev/null +++ b/sound/soc/mid-x86/mid_platform.c @@ -0,0 +1,365 @@ +/* + * mid_platform.c - Intel MID common Platform driver + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul <vinod.koul(a)intel.com> + * Author: Harsha Priya <priya.harsha(a)intel.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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../../drivers/staging/intel_sst/intel_sst.h" +#include "mid_platform.h" + +static struct snd_pcm_hardware snd_intelmad_pcm_hw = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_DOUBLE | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_MMAP| + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 | + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24 | + SNDRV_PCM_FMTBIT_S32 | SNDRV_PCM_FMTBIT_U32), + .rates = (SNDRV_PCM_RATE_8000| + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000), + .rate_min = MID_MIN_RATE, + .rate_max = MID_MAX_RATE, + .channels_min = MID_MIN_CHANNEL, + .channels_max = MID_MAX_CHANNEL, + .buffer_bytes_max = MID_MAX_BUFFER, + .period_bytes_min = MID_MIN_PERIOD_BYTES, + .period_bytes_max = MID_MAX_PERIOD_BYTES, + .periods_min = MID_MIN_PERIODS, + .periods_max = MID_MAX_PERIODS, + .fifo_size = MID_FIFO_SIZE, +}; + +/*MFLD - MSIC*/ +struct snd_soc_dai_driver mfld_dai[] = { +{ + .name = "Headset-cpu-dai", + .id = 0, + .playback = { + .channels_min = MID_STEREO, + .channels_max = MID_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Speaker-cpu-dai", + .id = 1, + .playback = { + .channels_min = MID_MONO, + .channels_max = MID_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Vibra 1-cpu-dai", + .id = 2, + .playback = { + .channels_min = MID_MONO, + .channels_max = MID_MONO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Vibra 2-cpu-dai", + .id = 3, + .playback = { + .channels_min = MID_MONO, + .channels_max = MID_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +}; + +static int mid_platform_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + struct mid_runtime_stream *stream; + int ret_val = 0; + + pr_debug("mid_platform_open called\n"); + + runtime = substream->runtime; + runtime->hw = snd_intelmad_pcm_hw; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + stream->stream_info.str_id = 0; + stream->stream_status = INIT; + stream->stream_info.mad_substream = substream; + /* allocate memory for SST API set */ + stream->sstdrv_ops = kzalloc(sizeof(*stream->sstdrv_ops), + GFP_KERNEL); + if (!stream->sstdrv_ops) { + pr_err("sst: mem allocation for ops fail\n"); + kfree(stream); + return -ENOMEM; + } + stream->sstdrv_ops->scard_ops = NULL; + stream->sstdrv_ops->vendor_id = MSIC_VENDOR_ID; + stream->sstdrv_ops->module_name = "msic_audio"; + + /* registering with SST driver to get access to SST APIs to use */ + ret_val = register_sst_card(stream->sstdrv_ops); + if (ret_val) { + pr_err("sst: sst card registration failed\n"); + return ret_val; + } + runtime->private_data = stream; + return snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); +} + +static int mid_platform_close(struct snd_pcm_substream *substream) +{ + struct mid_runtime_stream *stream; + int ret_val = 0, str_id; + + pr_debug("mid_platform_close called\n"); + + stream = substream->runtime->private_data; + str_id = stream->stream_info.str_id; + + if (str_id) + ret_val = stream->sstdrv_ops->control_set( + SST_SND_FREE, &str_id); + kfree(stream->sstdrv_ops); + kfree(stream); + return ret_val; +} + +static int mid_platform_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct mid_runtime_stream *stream; + int ret_val = 0, str_id; + + pr_debug("mid_platform_pcm_prepare called\n"); + + stream = substream->runtime->private_data; + str_id = stream->stream_info.str_id; + if (stream->stream_info.str_id) { + ret_val = stream->sstdrv_ops->control_set( + SST_SND_DROP, &str_id); + return ret_val; + } + + ret_val = snd_intelmad_alloc_stream(substream); + if (ret_val < 0) + return ret_val; + snprintf(substream->pcm->id, sizeof(substream->pcm->id), + "%d", stream->stream_info.str_id); + + ret_val = snd_intelmad_init_stream(substream); + if (ret_val) + return ret_val; + substream->runtime->hw.info = SNDRV_PCM_INFO_BLOCK_TRANSFER; + return ret_val; +} + +static int mid_platform_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + int ret_val = 0, str_id; + struct mid_runtime_stream *stream; + + pr_debug("mid_platform_pcm_trigger called\n"); + + stream = substream->runtime->private_data; + + str_id = stream->stream_info.str_id; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + pr_debug("sst: Trigger Start\n"); + ret_val = stream->sstdrv_ops->control_set( + SST_SND_START, &str_id); + if (ret_val) + return ret_val; + stream->stream_status = RUNNING; + stream->stream_info.mad_substream = substream; + break; + case SNDRV_PCM_TRIGGER_STOP: + pr_debug("sst: in stop\n"); + ret_val = stream->sstdrv_ops->control_set( + SST_SND_DROP, &str_id); + if (ret_val) + return ret_val; + stream->stream_status = DROPPED; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + pr_debug("sst: in pause\n"); + ret_val = stream->sstdrv_ops->control_set( + SST_SND_PAUSE, &str_id); + if (ret_val) + return ret_val; + stream->stream_status = PAUSED; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + pr_debug("sst: in pause release\n"); + ret_val = stream->sstdrv_ops->control_set( + SST_SND_RESUME, &str_id); + if (ret_val) + return ret_val; + stream->stream_status = RUNNING; + break; + default: + return -EINVAL; + } + return ret_val; +} + + +static snd_pcm_uframes_t mid_platform_pcm_pointer + (struct snd_pcm_substream *substream) +{ + /* struct snd_pcm_runtime *runtime = substream->runtime; */ + struct mid_runtime_stream *stream; + int ret_val; + struct pcm_stream_info *str_info; + + pr_debug("mid_platform_pcm_pointer called\n"); + + stream = substream->runtime->private_data; + if (stream->stream_status == INIT) + return 0; + str_info = &stream->stream_info; + ret_val = stream->sstdrv_ops->control_set( + SST_SND_BUFFER_POINTER, &str_info); + if (ret_val) { + pr_err("sst: error code = 0x%x\n", ret_val); + return ret_val; + } + + return stream->stream_info.buffer_ptr; +} + + +static struct snd_pcm_ops intel_mid_platform_ops = { + .open = mid_platform_open, + .close = mid_platform_close, + .ioctl = snd_pcm_lib_ioctl, + .prepare = mid_platform_pcm_prepare, + .trigger = mid_platform_pcm_trigger, + .pointer = mid_platform_pcm_pointer, +}; + +static void intel_mid_pcm_free(struct snd_pcm *pcm) +{ + pr_debug("intel_mid_pcm_free called\n"); + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +int intel_mid_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int retval = 0; + + pr_debug("intel_mid_pcm_new called\n"); + + if (dai->driver->playback.channels_min || + dai->driver->capture.channels_min) { + retval = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + MID_MIN_BUFFER, MID_MAX_BUFFER); + if (retval) { + pr_err("dma buffer allocationf fail\n"); + return retval; + } + } + + return retval; +} +struct snd_soc_platform_driver intelmid_soc_platform_drv = { + .ops = &intel_mid_platform_ops, + .pcm_new = intel_mid_pcm_new, + .pcm_free = intel_mid_pcm_free, +}; + +static int intelmid_platform_probe(struct platform_device *pdev) +{ + int ret; + + pr_debug("intelmid_platform_probe called\n"); + ret = snd_soc_register_platform(&pdev->dev, &intelmid_soc_platform_drv); + if (ret) { + pr_err("registering soc platform failed\n"); + return ret; + } + return snd_soc_register_dais(&pdev->dev, + mfld_dai, ARRAY_SIZE(mfld_dai)); +} + +static int intelmid_platform_remove(struct platform_device *pdev) +{ + + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(mfld_dai)); + + snd_soc_unregister_platform(&pdev->dev); + pr_debug("intelmid_platform_remove sucess\n"); + + return 0; +} + +static struct platform_driver intelmid_platform_driver = { + .driver = { + .name = "mid-audio-platform", + .owner = THIS_MODULE, + }, + .probe = intelmid_platform_probe, + .remove = intelmid_platform_remove, +}; + +static int __init intel_mid_soc_platform_init(void) +{ + pr_debug("intel_mid_soc_platform_init called\n"); + return platform_driver_register(&intelmid_platform_driver); +} +module_init(intel_mid_soc_platform_init); + +static void __exit intel_mid_soc_platform_exit(void) +{ + platform_driver_unregister(&intelmid_platform_driver); + pr_debug("intel_mid_soc_platform_exit sucess\n"); +} +module_exit(intel_mid_soc_platform_exit); + +MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); +MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); +MODULE_AUTHOR("Harsha Priya <priya.harsha(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("mid-platform"); diff --git a/sound/soc/mid-x86/mid_platform.h b/sound/soc/mid-x86/mid_platform.h new file mode 100644 index 0000000..e042266 --- /dev/null +++ b/sound/soc/mid-x86/mid_platform.h @@ -0,0 +1,71 @@ +/* + * mid_platform.h - Intel MID Platform driver header file + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul <vinod.koul(a)intel.com> + * Author: Harsha Priya <priya.harsha(a)intel.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; version 2 of the License. + * + * 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 __MID_PLATFORMDRV_H__ +#define __MID_PLATFORMDRV_H__ + +#define MID_MONO 1 +#define MID_STEREO 2 +#define MID_MAX_CAP 5 + +#define MID_MIN_RATE 8000 +#define MID_MAX_RATE 48000 +#define MID_MIN_CHANNEL 1 +#define MID_MAX_CHANNEL 5 +#define MID_MAX_BUFFER (800*1024) +#define MID_MIN_BUFFER (800*1024) +#define MID_MIN_PERIOD_BYTES 32 +#define MID_MAX_PERIOD_BYTES MID_MAX_BUFFER +#define MID_MIN_PERIODS 2 +#define MID_MAX_PERIODS (1024*2) +#define MID_FIFO_SIZE 0 +#define SST_CARD_NAMES "intel_mid_card" +#define MSIC_VENDOR_ID 3 + +int snd_intelmad_alloc_stream(struct snd_pcm_substream *substream); +int snd_intelmad_init_stream(struct snd_pcm_substream *substream); + +struct mid_runtime_stream { + int stream_status; + struct pcm_stream_info stream_info; + struct intel_sst_card_ops *sstdrv_ops; + unsigned long long buffer_ptr; +}; + +struct mid_dev_data { + struct snd_soc_dai_driver *dai; + int num_dais; +}; + +enum mid_drv_status { + INIT = 1, + STARTED, + RUNNING, + PAUSED, + DROPPED, +}; + +#endif diff --git a/sound/soc/mid-x86/mid_platform_lib.c b/sound/soc/mid-x86/mid_platform_lib.c new file mode 100644 index 0000000..531bf81 --- /dev/null +++ b/sound/soc/mid-x86/mid_platform_lib.c @@ -0,0 +1,122 @@ +/* + * mid_platform_lib.c - Intel MID Platform driver lib functions + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul <vinod.koul(a)intel.com> + * Author: Harsha Priya <priya.harsha(a)intel.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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/io.h> +#include <sound/pcm.h> +#include "../../../drivers/staging/intel_sst/intel_sst.h" +#include "../../../drivers/staging/intel_sst/intel_sst_ioctl.h" +#include "mid_platform.h" + +int snd_intelmad_alloc_stream(struct snd_pcm_substream *substream) +{ + struct mid_runtime_stream *stream = + substream->runtime->private_data; + struct snd_sst_stream_params param = {{{0,},},}; + struct snd_sst_params str_params = {0}; + int ret_val; + + /* set codec params and inform SST driver the same */ + + param.uc.pcm_params.codec = SST_CODEC_TYPE_PCM; + param.uc.pcm_params.num_chan = (u8) substream->runtime->channels; + param.uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits; + param.uc.pcm_params.reserved = 0; + param.uc.pcm_params.sfreq = substream->runtime->rate; + param.uc.pcm_params.ring_buffer_size = + snd_pcm_lib_buffer_bytes(substream); + param.uc.pcm_params.period_count = substream->runtime->period_size; + param.uc.pcm_params.ring_buffer_addr = + virt_to_phys(substream->dma_buffer.area); + substream->runtime->dma_area = substream->dma_buffer.area; + + pr_debug("period_cnt = %d\n", param.uc.pcm_params.period_count); + pr_debug("sfreq= %d, wd_sz = %d\n", + param.uc.pcm_params.sfreq, param.uc.pcm_params.pcm_wd_sz); + + str_params.sparams = param; + str_params.codec = SST_CODEC_TYPE_PCM; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + str_params.ops = STREAM_OPS_PLAYBACK; + str_params.device_type = substream->pcm->device + 1; + pr_debug("Playbck stream,Device %d\n", + substream->pcm->device); + } else { + str_params.ops = STREAM_OPS_CAPTURE; + str_params.device_type = SND_SST_DEVICE_CAPTURE; + pr_debug("Capture stream,Device %d\n", + substream->pcm->device); + } + ret_val = stream->sstdrv_ops->control_set(SST_SND_ALLOC, &str_params); + pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); + if (ret_val < 0) + return ret_val; + + stream->stream_info.str_id = ret_val; + pr_debug("str id : %d\n", stream->stream_info.str_id); + + return ret_val; +} + + +void period_elapsed(void *mad_substream) +{ + struct snd_pcm_substream *substream = mad_substream; + struct mid_runtime_stream *stream; + + if (!substream || !substream->runtime) + return; + stream = substream->runtime->private_data; + if (!stream) + return; + + if (stream->stream_status != RUNNING) + return; + pr_debug("calling period elapsed\n"); + snd_pcm_period_elapsed(substream); + return; +} + +int snd_intelmad_init_stream(struct snd_pcm_substream *substream) +{ + struct mid_runtime_stream *stream = + substream->runtime->private_data; + int ret_val; + + pr_debug("setting buffer ptr param\n"); + stream->stream_status = INIT; + stream->stream_info.period_elapsed = period_elapsed; + stream->stream_info.mad_substream = substream; + stream->stream_info.buffer_ptr = 0; + stream->stream_info.sfreq = substream->runtime->rate; + ret_val = stream->sstdrv_ops->control_set(SST_SND_STREAM_INIT, + &stream->stream_info); + if (ret_val) + pr_err("control_set ret error %d\n", ret_val); + return ret_val; + +} -- 1.7.2.3

3 4

Re: [alsa-devel] regarding references taken on platform and codec driver modules
by Harsha, Priya 03 Jan '11

03 Jan '11

>Hi, > >In soc-core.c, I see that try_module_get() is used to get reference count on the >codec and platform drivers. But I don't see try_module_put() to remove the >reference count after usage. > >Because of this, my reference count on codec and platform driver is never >reduced and I am not able to do rmmod of these drivers. But I can rmmod the >machine driver as there are no reference count taken on it. > >Is this intentional to avoid rmmod of these modules? Or this is to be fixed > Please ignore this email. >-Harsha

1 0

[alsa-devel] Recall: regarding references taken on platform and codec driver modules
by Harsha, Priya 03 Jan '11

03 Jan '11

Harsha, Priya would like to recall the message, "regarding references taken on platform and codec driver modules".

1 0

[alsa-devel] regarding references taken on platform and codec driver modules
by Harsha, Priya 03 Jan '11

03 Jan '11

Hi, In soc-core.c, I see that try_module_get() is used to get reference count on the codec and platform drivers. But I don't see try_module_put() to remove the reference count after usage. Because of this, my reference count on codec and platform driver is never reduced and I am not able to do rmmod of these drivers. But I can rmmod the machine driver as there are no reference count taken on it. Is this intentional to avoid rmmod of these modules? Or this is to be fixed -Harsha

1 0

[alsa-devel] [PATCH 0/4] Modify configures to support Samsung ASoC dependancyless
by Seungwhan Youn 03 Jan '11

03 Jan '11

Hi, These patches modify configuration to make audio support configurations on platform(or arch) from sound/soc/samsung. This prevents that audio config consider which board(or arch) can(or should) support ASoC audio driver, and also prevents audio config dependancy on future Samsung SoCs. These patches are modify below features:- o Add audio configurations on plat(or arch). o Modify audio configuration depends on plat(or arch) selection on ASoC. This patch-set is based on two different branches that :- o ARM patches are based on Kukjin Kim's git branch 'for-next' (commit id - 30fb9a94ec22435add3d2ea2d1b01c1a5ad6164c) o ASoC patches are based on Mark Brown's git branch 'for-next' (commit id - 722bc28384accb67c0bfbbe1914fd82d4d0c996a) This patch-set contains followings :- o To Kukjin Kim, Ben Dooks and Russell King, - [PATCH 1/4] ARM: Kconfig: Add audio support dependency - [PATCH 2/4] ARM: SAMSUNG: Add SMDK audio configuration o To Jassi Brar, Mark Brown and Liam Girdwood, - [PATCH 3/4] ASoC: SAMSUNG: Modify to indepedant depends on configuration - [PATCH 4/4] ASoC: SAMSUNG: Apply modifications for SMDK audio configuations Marry X-mas, Claude(Seungwhan Youn)

6 13

[alsa-devel] Support for X-Fi external front panel - how hard would it be to add it?
by Veselin Georgiev 03 Jan '11

03 Jan '11

Hi guys, I have a Creative X-fi Fatal1ty edition card that goes along with an external front panel (I/O) module, mounted in the 5.25" bay. So I see there is no support for the front panel in the ALSA X-fi driver, and I'm wondering how hard would it be to add it? Is it at all feasible? I'm hazy on the following stuff: - Does one need to reverse-engineer how the sound card communicates with the front panel, or there are datasheets about that available? - Correct me if I'm wrong, but AFAIK, the emu10k1 driver works fine with SB Live! or Audigy-based front-panels, so why not borrow some code from there and hack it into the ctxfi? I'm ready to try out some hacking with this and get my hands dirty with code if required. On a related note, there should be some official statement in the wiki saying that the front panel is not supported, as I did a lot of googling, checking various ubuntu/fedora forum posts that hinted the lack of support, and I'm still puzzled... :s Best Regards, Veselin Georgiev

1 0