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

[alsa-devel] Jack sense fails on Fujitsu p7120
by Robin Neatherway 06 Feb '11

06 Feb '11

I posted regarding this issue to alsa-users, but haven't had any response. I've come to the conclusion that I will have to modify the driver myself to rectify this problem, so I would appreciate some pointers as to how to get started with that. Running Xubuntu 10.10 with Alsa 1.0.23 The sound chip is Intel HDA, ALC260. I have tried setting options snd-hda-intel model=fujitsu in /etc/modprobe.d/alsa-base.conf, but this hasn't made any difference. There is no "Jack Sense" option in alsamixer. I've also tried a couple of values for fix_position. I've tried following the ubuntu wiki pages for Intel HDA and sound debugging in general, to no avail. When doing the jack sense test recommended on https://wiki.ubuntu.com/ALSA/JackSense the two files generated with and without the headphones plugged in were identical. This makes me think it is a driver problem. I pasted the output of `cat /proc/asound/card*/codec*` at http://pastebin.com/4NyvdsXf in case this is of any help.

3 15

[alsa-devel] ALC861-VD bad recording issue
by Nowardev-Team 05 Feb '11

05 Feb '11

This is the output of the alsa script to ge information about my system and the driver http://www.alsa-project.org/db/?f=9871b38d99bdd4cc91e398b60eb31361eb305395 <http://www.alsa-project.org/db/?f=9871b38d99bdd4cc91e398b60eb31361eb305395>the issue : when i record with mic i get always a bad recording with some tic tic tic tic , does not matter if i use low volume for the mic i can hear them anyway... if i use louder volume i can hear them ... i tried with arecord --quiet --file-type wav --rate=44100 -D plughw:Intel test.wav with this there is less rumors when i play with arecord --quiet --file-type wav --rate=44100 test.wav i got a lots of problems ... alsa (snd_pcm_recover) underrun occurred the strange thing is that on kubuntu 8.04 it was workin better ... that was my screenshot of alsamixer in the terminal http://nowardev.files.wordpress.com/2009/11/alsamixer.png now to get the " mic volume controller" i have to use options snd-hda-intel model=lenovo tried with all these lenovo is the only i can use ALC861VD/660VD ============== 3stack 3-jack 3stack-dig 3-jack with SPDIF OUT 6stack-dig 6-jack with SPDIF OUT 3stack-660 3-jack (for ALC660VD) 3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD) lenovo Lenovo 3000 C200 dallas Dallas laptops hp HP TX1000 asus-v1s ASUS V1Sn auto auto-config reading BIOS (default) but i get ... a lots of "controllers" like you can see here http://i0.simplest-image-hosting.net/168bf183b2abe8bc9188aacc163dd507/alsam… it seems that capture is the controller that can increase the mic volume ... :S i have tried every damned combinations of volumes but i get a lots of rumors always

1 0

[alsa-devel] issue with AD1989A HDA codec and alsamixer
by Vincent Meserette 05 Feb '11

05 Feb '11

Hi all, I am working on a system which includes a connexant AD1989A HDA codec connected to a ATOM processor. I have four microphones connected to the B and C ports of this codec. If I change one of the 3 capture gains present in the alsamixer (Capture, Capture 1 or Capture 3), I can't have any more signal on the B and C ports It seems to be due to the fact that ALSA breaks (for example for the Capture gain) the link between the ADC selector 0 and ADC_0 widgets of the codec, what can be easily seen with codecgraph. Have you ever had this kind of problem? Do you know how can I resolve it ? Many thanks for your asnwers and best regards Vincent

2 4

[alsa-devel] [PATCH] EDB93xx: Add support for CS4271 SPI-connected CODEC
by Alexander Sverdlin 04 Feb '11

04 Feb '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Add support for CS4271 SPI-connected CODEC to EDB93xx. Modified according to latest comments by H Hartley Sweeten. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- arch/arm/mach-ep93xx/edb93xx.c | 83 ++++++++++++++++++++++++++++++++++++++++ 1 files changed, 83 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-ep93xx/edb93xx.c b/arch/arm/mach-ep93xx/edb93xx.c index 3f37567..2c2f84d 100644 --- a/arch/arm/mach-ep93xx/edb93xx.c +++ b/arch/arm/mach-ep93xx/edb93xx.c @@ -30,9 +30,13 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/i2c-gpio.h> +#include <linux/spi/spi.h> + +#include <sound/cs4271.h> #include <mach/hardware.h> #include <mach/fb.h> +#include <mach/ep93xx_spi.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> @@ -94,6 +98,83 @@ static void __init edb93xx_register_i2c(void) /************************************************************************* + * EDB93xx SPI peripheral handling + *************************************************************************/ +static struct cs4271_platform_data edb93xx_cs4271_data = { + .gpio_nreset = -EINVAL, /* filled in later */ +}; + +static int edb93xx_cs4271_hw_setup(struct spi_device *spi) +{ + return gpio_request_one(EP93XX_GPIO_LINE_EGPIO6, + GPIOF_OUT_INIT_HIGH, spi->modalias); +} + +static void edb93xx_cs4271_hw_cleanup(struct spi_device *spi) +{ + gpio_free(EP93XX_GPIO_LINE_EGPIO6); +} + +static void edb93xx_cs4271_hw_cs_control(struct spi_device *spi, int value) +{ + gpio_set_value(EP93XX_GPIO_LINE_EGPIO6, value); +} + +static struct ep93xx_spi_chip_ops edb93xx_cs4271_hw = { + .setup = edb93xx_cs4271_hw_setup, + .cleanup = edb93xx_cs4271_hw_cleanup, + .cs_control = edb93xx_cs4271_hw_cs_control, +}; + +static struct spi_board_info edb93xx_spi_board_info[] __initdata = { + { + .modalias = "cs4271", + .platform_data = &edb93xx_cs4271_data, + .controller_data = &edb93xx_cs4271_hw, + .max_speed_hz = 6000000, + .bus_num = 0, + .chip_select = 0, + .mode = SPI_MODE_3, + }, +}; + +static struct ep93xx_spi_info edb93xx_spi_info __initdata = { + .num_chipselect = ARRAY_SIZE(edb93xx_spi_board_info), +}; + +static void __init edb93xx_register_spi(void) +{ + if (machine_is_edb9301() || machine_is_edb9302()) + edb93xx_cs4271_data.gpio_nreset = EP93XX_GPIO_LINE_EGPIO1; + else if (machine_is_edb9302a() || machine_is_edb9307a()) + edb93xx_cs4271_data.gpio_nreset = EP93XX_GPIO_LINE_H(2); + else if (machine_is_edb9315a()) + edb93xx_cs4271_data.gpio_nreset = EP93XX_GPIO_LINE_EGPIO14; + + ep93xx_register_spi(&edb93xx_spi_info, edb93xx_spi_board_info, + ARRAY_SIZE(edb93xx_spi_board_info)); +} + + +/************************************************************************* + * EDB93xx I2S + *************************************************************************/ +static int __init edb93xx_has_audio(void) +{ + return (machine_is_edb9301() || machine_is_edb9302() || + machine_is_edb9302a() || machine_is_edb9307a() || + machine_is_edb9315a()); +} + +static void __init edb93xx_register_i2s(void) +{ + if (edb93xx_has_audio()) { + ep93xx_register_i2s(); + } +} + + +/************************************************************************* * EDB93xx pwm *************************************************************************/ static void __init edb93xx_register_pwm(void) @@ -149,6 +230,8 @@ static void __init edb93xx_init_machine(void) edb93xx_register_flash(); ep93xx_register_eth(&edb93xx_eth_data, 1); edb93xx_register_i2c(); + edb93xx_register_spi(); + edb93xx_register_i2s(); edb93xx_register_pwm(); edb93xx_register_fb(); }

2 1

[alsa-devel] [PATCH] EDB93xx: Add support for CS4271 SPI-connected CODEC
by Alexander Sverdlin 04 Feb '11

04 Feb '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Add support for CS4271 SPI-connected CODEC to EDB93xx Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- arch/arm/mach-ep93xx/edb93xx.c | 85 ++++++++++++++++++++++++++++++++++++++++ 1 files changed, 85 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-ep93xx/edb93xx.c b/arch/arm/mach-ep93xx/edb93xx.c index 3f37567..c1e73ee 100644 --- a/arch/arm/mach-ep93xx/edb93xx.c +++ b/arch/arm/mach-ep93xx/edb93xx.c @@ -30,13 +30,17 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/i2c-gpio.h> +#include <linux/spi/spi.h> #include <mach/hardware.h> #include <mach/fb.h> +#include <mach/ep93xx_spi.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> +#include <sound/cs4271.h> + static void __init edb93xx_register_flash(void) { @@ -94,6 +98,85 @@ static void __init edb93xx_register_i2c(void) /************************************************************************* + * EDB93xx SPI peripheral handling + *************************************************************************/ +static struct cs4271_platform_data edb93xx_cs4271_data = { + .gpio_nreset = -EINVAL, /* filled in later */ +}; + +static int edb93xx_cs4271_hw_setup(struct spi_device *spi) +{ + return gpio_request_one(EP93XX_GPIO_LINE_EGPIO6, + GPIOF_OUT_INIT_HIGH, spi->modalias); +} + +static void edb93xx_cs4271_hw_cleanup(struct spi_device *spi) +{ + gpio_free(EP93XX_GPIO_LINE_EGPIO6); +} + +static void edb93xx_cs4271_hw_cs_control(struct spi_device *spi, int value) +{ + gpio_set_value(EP93XX_GPIO_LINE_EGPIO6, value); +} + +static struct ep93xx_spi_chip_ops edb93xx_cs4271_hw = { + .setup = edb93xx_cs4271_hw_setup, + .cleanup = edb93xx_cs4271_hw_cleanup, + .cs_control = edb93xx_cs4271_hw_cs_control, +}; + +static struct spi_board_info edb93xx_spi_board_info[] __initdata = { + { + .modalias = "cs4271", + .platform_data = &edb93xx_cs4271_data, + .controller_data = &edb93xx_cs4271_hw, + .max_speed_hz = 6000000, + .bus_num = 0, + .chip_select = 0, + .mode = SPI_MODE_3, + }, +}; + +static struct ep93xx_spi_info edb93xx_spi_info __initdata = { + .num_chipselect = ARRAY_SIZE(edb93xx_spi_board_info), +}; + +static void __init edb93xx_register_spi(void) +{ + if (machine_is_edb9301() || machine_is_edb9302()) + edb93xx_cs4271_data.gpio_nreset = EP93XX_GPIO_LINE_EGPIO1; + else if (machine_is_edb9302a() || machine_is_edb9307a()) + edb93xx_cs4271_data.gpio_nreset = EP93XX_GPIO_LINE_H(2); + else if (machine_is_edb9315a()) + edb93xx_cs4271_data.gpio_nreset = EP93XX_GPIO_LINE_EGPIO14; + else + return; + + ep93xx_register_spi(&edb93xx_spi_info, edb93xx_spi_board_info, + ARRAY_SIZE(edb93xx_spi_board_info)); +} + + +/************************************************************************* + * EDB93xx I2S + *************************************************************************/ +static int __init edb93xx_has_audio(void) +{ + return (machine_is_edb9301() || machine_is_edb9302() || + machine_is_edb9302a() || machine_is_edb9307a() || + machine_is_edb9315a()); +} + +static void __init edb93xx_register_i2s(void) +{ + if (edb93xx_has_audio()) { + ep93xx_register_i2s(); + } +} + + +/************************************************************************* * EDB93xx pwm *************************************************************************/ static void __init edb93xx_register_pwm(void) @@ -149,6 +232,8 @@ static void __init edb93xx_init_machine(void) edb93xx_register_flash(); ep93xx_register_eth(&edb93xx_eth_data, 1); edb93xx_register_i2c(); + edb93xx_register_spi(); + edb93xx_register_i2s(); edb93xx_register_pwm(); edb93xx_register_fb(); }

3 2

[alsa-devel] [PATCH] ASoC: ep93xx-ac97: remove extra empty line
by Mika Westerberg 04 Feb '11

04 Feb '11

Signed-off-by: Mika Westerberg <mika.westerberg(a)iki.fi> --- sound/soc/ep93xx/ep93xx-ac97.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c index 68a0bae..104e95c 100644 --- a/sound/soc/ep93xx/ep93xx-ac97.c +++ b/sound/soc/ep93xx/ep93xx-ac97.c @@ -253,7 +253,6 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, struct ep93xx_ac97_info *info = snd_soc_dai_get_drvdata(dai); unsigned v = 0; - switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: -- 1.7.2.3

3 2

[alsa-devel] [PATCH 2/3] ASoC: mid-x86 machine driver - add jack detect support
by Harsha Priya 04 Feb '11

04 Feb '11

This patch adds the jack detection in mid-86 machine driver Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/mid-x86/mfld_machine.c | 186 ++++++++++++++++++++++++++++++++------ 1 files changed, 159 insertions(+), 27 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 7925851..7abd3fc 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -27,6 +27,7 @@ #include <linux/init.h> #include <linux/device.h> #include <linux/slab.h> +#include <linux/io.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -35,10 +36,38 @@ #define MID_MONO 1 #define MID_STEREO 2 #define MID_MAX_CAP 5 +#define IRQ_BASE 0xFFFF7FCD +#define IRQ_SIZE 1 static unsigned int hs_switch; static unsigned int lo_dac; +struct mfld_mc_private { + struct platform_device *socdev; + struct workqueue_struct *mfld_jack_wq; + void __iomem *int_base; + struct snd_soc_codec *codec; +} mc_drv_ctx; + +struct snd_soc_jack mfld_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin mfld_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "AMIC1", + .mask = SND_JACK_MICROPHONE, + }, +}; +/* jack detection voltage zones */ +static struct snd_soc_jack_zone mfld_zones[] = { + {SN95031_VOL_START, SN95031_VOL_AM_HS, SND_JACK_HEADPHONE}, + {SN95031_VOL_AM_HS, SN95031_VOL_HS, SND_JACK_HEADSET}, +}; + /* sound card controls */ static const char *headset_switch_text[] = {"Earpiece", "Headset"}; @@ -67,13 +96,11 @@ static int headset_set_switch(struct snd_kcontrol *kcontrol, 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_enable_pin(&codec->dapm, "Headphones"); 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_disable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); } snd_soc_dapm_sync(&codec->dapm); @@ -91,12 +118,10 @@ static void lo_enable_out_pins(struct snd_soc_codec *codec) 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_enable_pin(&codec->dapm, "Headphones"); 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_disable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); } } @@ -130,8 +155,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, 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, "Headphones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22); break; @@ -162,12 +186,36 @@ static const struct snd_kcontrol_new mfld_snd_controls[] = { lo_get_switch, lo_set_switch), }; +static const struct snd_soc_dapm_widget mfld_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route mfld_map[] = { + {"Headphones", NULL, "HPOUTR"}, + {"Headphones", NULL, "HPOUTL"}, + {"Mic", NULL, "AMIC1"}, +}; + static int mfld_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; + /* Add jack sense widgets */ + snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets)); + + /* Set up the map */ + snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map)); + + /* always connected */ + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Mic"); + snd_soc_dapm_sync(dapm); + + mc_drv_ctx.codec = codec; + ret_val = snd_soc_add_controls(codec, mfld_snd_controls, ARRAY_SIZE(mfld_snd_controls)); if (ret_val) { @@ -175,8 +223,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) 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"); + snd_soc_dapm_disable_pin(dapm, "Headphones"); /* default is lineout NC, userspace sets it explcitly */ snd_soc_dapm_disable_pin(dapm, "LINEOUTL"); snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); @@ -185,7 +232,32 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) /* we dont use linein in this so set to NC */ snd_soc_dapm_disable_pin(dapm, "LINEINL"); snd_soc_dapm_disable_pin(dapm, "LINEINR"); - return snd_soc_dapm_sync(dapm); + snd_soc_dapm_sync(dapm); + + /* Headset jack detection */ + ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1, &mfld_jack); + + if (ret_val) { + pr_err("jack creation failed\n"); + return ret_val; + } + + ret_val = snd_soc_jack_add_pins(&mfld_jack, + ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins); + if (ret_val) { + pr_err("adding jack pins failed\n"); + return ret_val; + } + ret_val = snd_soc_jack_add_zones(&mfld_jack, + ARRAY_SIZE(mfld_zones), mfld_zones); + if (ret_val) { + pr_err("adding jack zones failed\n"); + return ret_val; + } + disable_jack_btn(codec); + return ret_val; } struct snd_soc_dai_link mfld_msic_dailink[] = { @@ -234,37 +306,97 @@ static struct snd_soc_card snd_soc_card_mfld = { .num_links = ARRAY_SIZE(mfld_msic_dailink), }; +void snd_mfld_jack_detection(struct work_struct *work) +{ + struct mfld_jack_msg_wq *jack_msg = + container_of(work, struct mfld_jack_msg_wq, wq); + + sn95031_jack_detection(&mfld_jack, jack_msg); + kfree(jack_msg); +} + +static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev) +{ + struct mfld_jack_msg_wq *jack_msg; + struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev; + + jack_msg = kzalloc(sizeof(*jack_msg), GFP_ATOMIC); + memcpy_fromio(&jack_msg->intr_id, + ((void *)(mc_private->int_base)), + sizeof(u8)); + jack_msg->codec = mc_private->codec; + pr_debug("snd_mfld_jack_intr_handler called value %x\n", + jack_msg->intr_id); + INIT_WORK(&jack_msg->wq, snd_mfld_jack_detection); + queue_work(mc_private->mfld_jack_wq, &jack_msg->wq); + + return IRQ_HANDLED; +} + static int __devinit snd_mfld_mc_probe(struct platform_device *pdev) { - struct platform_device *socdev; - int ret_val = 0; + int ret_val = 0, irq; pr_debug("snd_mfld_mc_probe called\n"); - socdev = platform_device_alloc("soc-audio", -1); - if (!socdev) { + /* retrive the irq number */ + irq = platform_get_irq(pdev, 0); + /* audio interrupt base of SRAM location where + * interrupts are stored by System FW */ + mc_drv_ctx.int_base = ioremap_nocache(IRQ_BASE, IRQ_SIZE); + if (!mc_drv_ctx.int_base) { + pr_err("Mapping of cache failed\n"); + ret_val = -ENOMEM; + goto unalloc; + } + /* create botoom half handler for interrupt */ + mc_drv_ctx.mfld_jack_wq = create_workqueue("mfld_jack_wq"); + if (!mc_drv_ctx.mfld_jack_wq) { + pr_err("cannot create jack workqueue\n"); + ret_val = -ENOMEM; + goto unalloc; + } + /* register for interrupt */ + ret_val = request_irq(irq, snd_mfld_jack_intr_handler, + IRQF_SHARED, pdev->dev.driver->name, + &mc_drv_ctx); + if (ret_val) { + pr_err("cannot register IRQ\n"); + goto destroy_wq; + } + /* create soc device */ + mc_drv_ctx.socdev = platform_device_alloc("soc-audio", -1); + if (!mc_drv_ctx.socdev) { pr_err("soc-audio device allocation failed\n"); - return -ENOMEM; + ret_val = -ENOMEM; + goto destroy_wq; } - platform_set_drvdata(socdev, &snd_soc_card_mfld); - ret_val = platform_device_add(socdev); + platform_set_drvdata(mc_drv_ctx.socdev, &snd_soc_card_mfld); + ret_val = platform_device_add(mc_drv_ctx.socdev); if (ret_val) { pr_err("Unable to add soc-audio device, err %d\n", ret_val); - platform_device_put(socdev); + goto unregister; } - - platform_set_drvdata(pdev, socdev); - + platform_set_drvdata(pdev, &mc_drv_ctx); pr_debug("successfully exited probe\n"); return ret_val; + +unregister: + platform_device_put(mc_drv_ctx.socdev); +destroy_wq: + destroy_workqueue(mc_drv_ctx.mfld_jack_wq); +unalloc: + return ret_val; } static int __devexit snd_mfld_mc_remove(struct platform_device *pdev) { - struct platform_device *socdev = platform_get_drvdata(pdev); - pr_debug("snd_mfld_mc_remove called\n"); + struct mfld_mc_private *mc_private_data = platform_get_drvdata(pdev); - platform_device_unregister(socdev); + pr_debug("snd_mfld_mc_remove called\n"); + free_irq(platform_get_irq(pdev, 0), mc_private_data); + destroy_workqueue(mc_private_data->mfld_jack_wq); + platform_device_unregister(mc_private_data->socdev); platform_set_drvdata(pdev, NULL); return 0; } -- 1.7.3.4

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[alsa-devel] [PATCH 3/3] ASoC: sn95031 codec - adding jack detection/reporting
by Harsha Priya 04 Feb '11

04 Feb '11

This patch adds the jack detection logic in sn95031 codec. This also adds code for reading the ADC to figure out the mic bias and therby report jack type detected Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/sn95031.c | 185 +++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/sn95031.h | 56 +++++++++++++ 2 files changed, 237 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 40e285d..07b6ea7 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -27,6 +27,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> +#include <linux/input.h> #include <asm/intel_scu_ipc.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -39,12 +40,129 @@ #define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) #define SN95031_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) +/* adc helper functions */ + +/* enables mic bias voltage */ +void sn95031_enable_mic_bias(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_VAUD, BIT(2)|BIT(1)|BIT(0)); + snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(2), BIT(2)); +} + +/* Enable/Disable the ADC depending on the argument */ +static void configure_adc(struct snd_soc_codec *sn95031_codec, int val) +{ + int value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); + + if (val) { + /* Enable and start the ADC */ + value |= (SN95031_ADC_ENBL | SN95031_ADC_START); + value &= (~ADC_NO_LOOP); + } else + /* Just stop the ADC */ + value &= (~SN95031_ADC_START); + snd_soc_write(sn95031_codec, SN95031_ADC1CNTL1, value); +} + /* - * todo: - * capture paths - * jack detection - * PM functions + * finds an empty channel for conversion + * If the ADC is not enabled then start using 0th channel + * itself. Otherwise find an empty channel by looking for a + * channel in which the stopbit is set to 1. returns the index + * of the first free channel if succeeds or an error code. + * + * Context: can sleep + * */ +static int find_free_channel(struct snd_soc_codec *sn95031_codec) +{ + int ret = 0, i, value; + + /* check whether ADC is enabled */ + value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); + + if ((value & SN95031_ADC_ENBL) == 0) + return 0; + + /* ADC is already enabled; Looking for an empty channel */ + for (i = 0; i < ADC_CHANLS_MAX; i++) { + value = snd_soc_read(sn95031_codec, ADC_CHNL_START_ADDR + i); + if (value & SN95031_STOPBIT_MASK) { + ret = i; + break; + } + } + return (ret > ADC_LOOP_MAX) ? (-EINVAL) : ret; +} + +/* Initialize the ADC for reading thermistor values. Can sleep. */ +static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec) +{ + int base_addr, chnl_addr; + int value; + static int channel_index; + + /* Index of the first channel in which the stop bit is set */ + channel_index = find_free_channel(sn95031_codec); + if (channel_index < 0) { + pr_err("No free ADC channels"); + return channel_index; + } + + base_addr = ADC_CHNL_START_ADDR + channel_index; + + if (!(channel_index == 0 || channel_index == ADC_LOOP_MAX)) { + /* Reset stop bit for channels other than 0 and 12 */ + value = snd_soc_read(sn95031_codec, base_addr); + /* Set the stop bit to zero */ + snd_soc_write(sn95031_codec, base_addr, value & 0xEF); + /* Index of the first free channel */ + base_addr++; + channel_index++; + } + + /* Since this is the last channel, set the stop bit + to 1 by ORing the DIE_SENSOR_CODE with 0x10 */ + snd_soc_write(sn95031_codec, base_addr, AUDIO_DETECT_CODE | 0x10); + + chnl_addr = ADC_DATA_START_ADDR + 2 * channel_index; + pr_debug("mid_initialize : %x", chnl_addr); + configure_adc(sn95031_codec, 1); + return chnl_addr; +} + + +/* reads the ADC registers and gets the mic bias value in mV. */ +static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) +{ + u16 adc_adr = sn95031_initialize_adc(codec); + u16 adc_val1, adc_val2; + unsigned int mic_bias; + + sn95031_enable_mic_bias(codec); + + /* Enable the sound card for conversion before reading */ + snd_soc_write(codec, SN95031_ADC1CNTL3, 0x05); + /* Re-toggle the RRDATARD bit */ + snd_soc_write(codec, SN95031_ADC1CNTL3, 0x04); + + /* Read the higher bits of data */ + msleep(1000); + adc_val1 = snd_soc_read(codec, adc_adr); + adc_adr++; + adc_val2 = snd_soc_read(codec, adc_adr); + + /* Adding lower two bits to the higher bits */ + mic_bias = (adc_val1 << 2) + (adc_val2 & 3); + mic_bias = (mic_bias * ADC_ONE_LSB_MULTIPLIER) / 1000; + pr_debug("mic bias = %dmV\n", mic_bias); + return mic_bias; +} +EXPORT_SYMBOL_GPL(sn95031_get_mic_bias); + +/*end - adc helper functions */ + +/* codec read/write functions */ static inline unsigned int sn95031_read(struct snd_soc_codec *codec, unsigned int reg) @@ -70,6 +188,8 @@ static inline int sn95031_write(struct snd_soc_codec *codec, return ret; } +/* end - codec read/write functions */ + static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -649,6 +769,63 @@ struct snd_soc_dai_driver sn95031_dais[] = { }, }; +int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack, + struct snd_soc_codec *codec) +{ + int micbias = sn95031_get_mic_bias(codec); + + int jack_type = snd_soc_jack_get_type(mfld_jack, micbias); + + /* If american headest, program the gpio control registers */ + if (micbias >= SN95031_VOL_HP && micbias < SN95031_VOL_AM_HS) + snd_soc_update_bits(codec, AUDIO_GPIO_CTRL, BIT(1), BIT(1)); + else + snd_soc_update_bits(codec, AUDIO_GPIO_CTRL, BIT(1), BIT(0)); + + pr_debug("jack type detected = %d\n", jack_type); + if (jack_type == SND_JACK_HEADSET) + enable_jack_btn(codec); + return jack_type; +} + +void sn95031_jack_detection(struct snd_soc_jack *mfld_jack, + struct mfld_jack_msg_wq *jack_msg) +{ + unsigned int status, mask; + static int jack_state = SND_JACK_HEADSET; + struct snd_soc_codec *codec = jack_msg->codec; + + if (!codec) { + pr_err("codec is null\n"); + return; + } + pr_debug("interrupt id read in sram = 0x%x\n", jack_msg->intr_id); + if (jack_msg->intr_id & 0x1) { + pr_debug("short_push detected\n"); + mask = status = SND_JACK_BTN_0; + } else if (jack_msg->intr_id & 0x2) { + pr_debug("long_push detected\n"); + mask = status = SND_JACK_BTN_1; + } else if (jack_msg->intr_id & 0x4) { + mask = status = sn95031_get_headset_state(mfld_jack, codec); + jack_state = mask; + } else if (jack_msg->intr_id & 0x8) { + pr_debug("headset or headphones removed\n"); + status = 0; + mask = jack_state; + disable_jack_btn(codec); + } else { + pr_err("unidentified interrupt\n"); + return; + } + + snd_soc_jack_report(mfld_jack, status, mask); + /*button pressed and released */ + if (status == SND_JACK_BTN_0 || status == SND_JACK_BTN_1) + snd_soc_jack_report(mfld_jack, 0, mask); +} +EXPORT_SYMBOL_GPL(sn95031_jack_detection); + /* codec registration */ static int sn95031_codec_probe(struct snd_soc_codec *codec) { diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index e2b17d9..dbae58d 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -25,6 +25,7 @@ */ #ifndef _SN95031_H #define _SN95031_H +#include <sound/jack.h> /*register map*/ #define SN95031_VAUD 0xDB @@ -96,4 +97,59 @@ #define SN95031_SSR5 0x384 #define SN95031_SSR6 0x385 +/* ADC registers */ + +#define SN95031_ADC1CNTL1 0x1C0 +#define SN95031_ADC_ENBL 0x10 +#define SN95031_ADC_START 0x08 +#define SN95031_ADC1CNTL3 0x1C2 +#define SN95031_ADCTHERM_ENBL 0x04 +#define SN95031_ADCRRDATA_ENBL 0x05 +#define SN95031_STOPBIT_MASK 16 +#define SN95031_ADCTHERM_MASK 4 +#define ADC_CHANLS_MAX 15 /* Number of ADC channels */ +#define ADC_LOOP_MAX (ADC_CHANLS_MAX - 1) +#define ADC_NO_LOOP 0x07 + +/* ADC channel code values */ +#define AUDIO_DETECT_CODE 0x06 +#define AUDIO_GPIO_CTRL 0x070 +/* ADC base addresses */ +#define ADC_CHNL_START_ADDR 0x1C5 /* increments by 1 */ +#define ADC_DATA_START_ADDR 0x1D4 /* increments by 2 */ +/* multipier to convert to mV */ +#define ADC_ONE_LSB_MULTIPLIER 2346 + +enum soc_mic_bias_zones { + SN95031_VOL_START = 0, + /* mic bias volutage range for Headphones*/ + SN95031_VOL_HP = 400, + /* mic bias volutage range for American Headset*/ + SN95031_VOL_AM_HS = 650, + /* mic bias volutage range for Headset*/ + SN95031_VOL_HS = 2000, + SN95031_VOL_UNDEFINED, +}; + + +struct mfld_jack_msg_wq { + u8 intr_id; + struct snd_soc_codec *codec; + struct work_struct wq; +}; + +static inline void disable_jack_btn(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_BTNCTRL2, 0x00); +} + +static inline void enable_jack_btn(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_BTNCTRL1, 0x77); + snd_soc_write(codec, SN95031_BTNCTRL2, 0x01); +} + +extern void sn95031_jack_detection(struct snd_soc_jack *mfld_jack, + struct mfld_jack_msg_wq *jack_msg); + #endif -- 1.7.3.4

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Re: [alsa-devel] [Alsa-devel] how to cross compile the alsa util?
by Leonid 04 Feb '11

04 Feb '11

Hello alls: Please help. I also cross compiled alsa utils trhough CodeSourcery: PATH=$PATH:/opt/arm-2007q1/bin CC=arm-none-linux-gnueabi-gcc ./configure --host=i686-pc-linux-gnu \ --target=arm-none-linux-gnueabi \ --with-alsa-prefix=/usr/lib \ --with-alsa-inc-prefix=/usr/include/alsa configure out... --cut checking for ncursesw5-config... yes checking for curses library... ncursesw checking for curses header name...<ncurses.h> checking for curses compiler flags... -I/usr/include/ncursesw checking for curses NLS support... yes checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking panel.h usability... no checking panel.h presence... no checking for panel.h... no configure: error: required curses helper header not found but libncursesw5 and aibncursesw5-dev is installed to host: more... I changet the configuration and add option --with-curses=ncurses (libncurses5-dev also installed): PATH=$PATH:/opt/arm-2007q1/bin CC=arm-none-linux-gnueabi-gcc ./configure --host=i686-pc-linux-gnu \ --target=arm-none-linux-gnueabi \ --with-alsa-prefix=/usr/lib \ --with-alsa-inc-prefix=/usr/include/alsa \ --with-curses=ncurses out... --cut checking for ncurses5-config... yes checking for curses library... ncurses checking for curses header name...<ncurses.h> checking for curses compiler flags... checking for curses NLS support... no checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking panel.h usability... no checking panel.h presence... no checking for panel.h... no configure: error: required curses helper header not found the same effect. what am i doing wrong? can be cross compiled ncurses? -- with best wishes Leonid On Mon, 31 May 2004, Roc Wu wrote: > Hello alls: > > Sorry for post the mail again. I posted it several > days ago, but no response. Anybody can give me some > hints? > > I cross compiled the alsa-lib-1.04 to arm platform. > The 2.6.6 kernel including alsa driver is ok on our > ARM board, so I want to test the driver. how to do it? > > I think maybe the first step is to crosscompile the > alsa utils to play a wave audio file. > > CC=arm-linux-gcc ./configure --host=i686-linux > --target=arm-linux ./configure --help --- cut --- --with-alsa-prefix=PFX Prefix where Alsa library is installed (optional) --with-alsa-inc-prefix=PFX Prefix where include libraries are (optional) --- cut --- Jaroslav ----- Jaroslav Kysela<[EMAIL PROTECTED]> Linux Kernel Sound Maintainer ALSA Project, SuSE Labs

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[alsa-devel] [PATCH 12/19] twl4030: mfd_cell is now implicitly available to drivers
by Andres Salomon 04 Feb '11

04 Feb '11

No need to explicitly set the cell's platform_data/data_size. In this case, move the various platform_data pointers to driver_data. All of the clients which make use of it are also changed. Signed-off-by: Andres Salomon <dilinger(a)queued.net> --- drivers/input/misc/twl4030-vibra.c | 2 +- drivers/mfd/twl4030-codec.c | 6 ++---- sound/soc/codecs/twl4030.c | 9 ++++++--- 3 files changed, 9 insertions(+), 8 deletions(-) diff --git a/drivers/input/misc/twl4030-vibra.c b/drivers/input/misc/twl4030-vibra.c index 014dd4a..7d7602a 100644 --- a/drivers/input/misc/twl4030-vibra.c +++ b/drivers/input/misc/twl4030-vibra.c @@ -196,7 +196,7 @@ static SIMPLE_DEV_PM_OPS(twl4030_vibra_pm_ops, static int __devinit twl4030_vibra_probe(struct platform_device *pdev) { - struct twl4030_codec_vibra_data *pdata = pdev->dev.platform_data; + struct twl4030_codec_vibra_data *pdata = platform_get_drvdata(pdev); struct vibra_info *info; int ret; diff --git a/drivers/mfd/twl4030-codec.c b/drivers/mfd/twl4030-codec.c index 9a4b196..d44ad30 100644 --- a/drivers/mfd/twl4030-codec.c +++ b/drivers/mfd/twl4030-codec.c @@ -208,15 +208,13 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) if (pdata->audio) { cell = &codec->cells[childs]; cell->name = "twl4030-codec"; - cell->platform_data = pdata->audio; - cell->data_size = sizeof(*pdata->audio); + cell->driver_data = pdata->audio; childs++; } if (pdata->vibra) { cell = &codec->cells[childs]; cell->name = "twl4030-vibra"; - cell->platform_data = pdata->vibra; - cell->data_size = sizeof(*pdata->vibra); + cell->driver_data = pdata->vibra; childs++; } diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e4d464b..3721671 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -130,6 +130,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { /* codec private data */ struct twl4030_priv { struct snd_soc_codec codec; + struct twl4030_codec_audio_data *pdata; unsigned int codec_powered; @@ -297,8 +298,8 @@ static inline void twl4030_reset_registers(struct snd_soc_codec *codec) static void twl4030_init_chip(struct snd_soc_codec *codec) { - struct twl4030_codec_audio_data *pdata = dev_get_platdata(codec->dev); struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); + struct twl4030_codec_audio_data *pdata = twl4030->pdata; u8 reg, byte; int i = 0; @@ -732,9 +733,9 @@ static int aif_event(struct snd_soc_dapm_widget *w, static void headset_ramp(struct snd_soc_codec *codec, int ramp) { - struct twl4030_codec_audio_data *pdata = codec->dev->platform_data; unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); + struct twl4030_codec_audio_data *pdata = twl4030->pdata; /* Base values for ramp delay calculation: 2^19 - 2^26 */ unsigned int ramp_base[] = {524288, 1048576, 2097152, 4194304, 8388608, 16777216, 33554432, 67108864}; @@ -2251,6 +2252,7 @@ static int twl4030_soc_resume(struct snd_soc_codec *codec) static int twl4030_soc_probe(struct snd_soc_codec *codec) { + struct twl4030_codec_audio_data *pdata = dev_get_drvdata(codec->dev); struct twl4030_priv *twl4030; twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL); @@ -2258,6 +2260,7 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) printk("Can not allocate memroy\n"); return -ENOMEM; } + twl4030->pdata = pdata; snd_soc_codec_set_drvdata(codec, twl4030); /* Set the defaults, and power up the codec */ twl4030->sysclk = twl4030_codec_get_mclk() / 1000; @@ -2297,7 +2300,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { static int __devinit twl4030_codec_probe(struct platform_device *pdev) { - struct twl4030_codec_audio_data *pdata = pdev->dev.platform_data; + struct twl4030_codec_audio_data *pdata = platform_get_drvdata(pdev); if (!pdata) { dev_err(&pdev->dev, "platform_data is missing\n"); -- 1.7.2.3

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