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

[alsa-devel] [PATCH] ASoC: Relax restrictions on WM8962 BCLK configuration
by Mark Brown 02 Oct '10

02 Oct '10

The restrictions on configuring BCLK are overly cautious, other constraints in the system should ensure that reconfiguration is not possible when the device is sufficiently active to be unable to support reclocking. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 6 ------ 1 files changed, 0 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index be34146..38cbf85 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -911,12 +911,6 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec) int clocking2 = 0; int aif2 = 0; - /* If the CODEC is powered on we can configure BCLK */ - if (codec->bias_level != SND_SOC_BIAS_OFF) { - dev_dbg(codec->dev, "Bias is off, can't configure BCLK\n"); - return; - } - if (!wm8962->bclk) { dev_dbg(codec->dev, "No BCLK rate configured\n"); return; -- 1.7.1

2 2

[alsa-devel] [PATCH] ASoC: Add support for WM8962 GPIO outputs
by Mark Brown 02 Oct '10

02 Oct '10

The WM8962 features five GPIOs, add support for controlling their output state via gpiolib. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/wm8962.h | 1 + sound/soc/codecs/wm8962.c | 119 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8962.h | 1 + 3 files changed, 121 insertions(+), 0 deletions(-) diff --git a/include/sound/wm8962.h b/include/sound/wm8962.h index 9722aac..2b5306c 100644 --- a/include/sound/wm8962.h +++ b/include/sound/wm8962.h @@ -15,6 +15,7 @@ #define WM8962_GPIO_SET 0x10000 struct wm8962_pdata { + int gpio_base; u32 gpio_init[WM8962_MAX_GPIO]; /* Setup for microphone detection, raw value to be written to diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 38cbf85..eb66c66 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -17,6 +17,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/gcd.h> +#include <linux/gpio.h> #include <linux/i2c.h> #include <linux/input.h> #include <linux/platform_device.h> @@ -70,6 +71,10 @@ struct wm8962_priv { struct work_struct beep_work; int beep_rate; #endif + +#ifdef CONFIG_GPIOLIB + struct gpio_chip gpio_chip; +#endif }; /* We can't use the same notifier block for more than one supply and @@ -1646,6 +1651,118 @@ static void wm8962_free_beep(struct snd_soc_codec *codec) } #endif +#ifdef CONFIG_GPIOLIB +static inline struct wm8962_priv *gpio_to_wm8962(struct gpio_chip *chip) +{ + return container_of(chip, struct wm8962_priv, gpio_chip); +} + +static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset) +{ + struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); + struct snd_soc_codec *codec = wm8962->codec; + int mask = 0; + int val; + + /* The WM8962 GPIOs aren't linearly numbered. For simplicity + * we export linear numbers and error out if the unsupported + * ones are requsted. + */ + switch (offset + 1) { + case 2: + mask = WM8962_CLKOUT2_SEL_MASK; + val = 1 << WM8962_CLKOUT2_SEL_SHIFT; + break; + case 3: + mask = WM8962_CLKOUT3_SEL_MASK; + val = 1 << WM8962_CLKOUT3_SEL_SHIFT; + break; + case 5: + case 6: + break; + default: + return -EINVAL; + } + + /* Some of the GPIOs are behind MFP configuration */ + if (mask) + snd_soc_update_bits(codec, WM8962_ANALOGUE_CLOCKING1, + mask, val); + + return 0; +} + +static void wm8962_gpio_set(struct gpio_chip *chip, unsigned offset, int value) +{ + struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); + struct snd_soc_codec *codec = wm8962->codec; + + snd_soc_update_bits(codec, WM8962_GPIO_BASE + offset, + WM8962_GP2_LVL, value << WM8962_GP2_LVL_SHIFT); +} + +static int wm8962_gpio_direction_out(struct gpio_chip *chip, + unsigned offset, int value) +{ + struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); + struct snd_soc_codec *codec = wm8962->codec; + int val; + + /* Force function 1 (logic output) */ + val = (1 << WM8962_GP2_FN_SHIFT) | (value << WM8962_GP2_LVL_SHIFT); + + return snd_soc_update_bits(codec, WM8962_GPIO_BASE + offset, + WM8962_GP2_FN_MASK | WM8962_GP2_LVL, val); +} + +static struct gpio_chip wm8962_template_chip = { + .label = "wm8962", + .owner = THIS_MODULE, + .request = wm8962_gpio_request, + .direction_output = wm8962_gpio_direction_out, + .set = wm8962_gpio_set, + .can_sleep = 1, +}; + +static void wm8962_init_gpio(struct snd_soc_codec *codec) +{ + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); + int ret; + + wm8962->gpio_chip = wm8962_template_chip; + wm8962->gpio_chip.ngpio = WM8962_MAX_GPIO; + wm8962->gpio_chip.dev = codec->dev; + + if (pdata && pdata->gpio_base) + wm8962->gpio_chip.base = pdata->gpio_base; + else + wm8962->gpio_chip.base = -1; + + ret = gpiochip_add(&wm8962->gpio_chip); + if (ret != 0) + dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); +} + +static void wm8962_free_gpio(struct snd_soc_codec *codec) +{ + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = gpiochip_remove(&wm8962->gpio_chip); + if (ret != 0) + dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); +} +#else +static void wm8962_init_gpio(struct snd_soc_codec *codec) +{ +} + +static void wm8962_free_gpio(struct snd_soc_codec *codec) +{ +} +#endif + static int wm8962_probe(struct snd_soc_codec *codec) { int ret; @@ -1778,6 +1895,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) wm8962_add_widgets(codec); wm8962_init_beep(codec); + wm8962_init_gpio(codec); if (i2c->irq) { if (pdata && pdata->irq_active_low) { @@ -1828,6 +1946,7 @@ static int wm8962_remove(struct snd_soc_codec *codec) if (i2c->irq) free_irq(i2c->irq, codec); + wm8962_free_gpio(codec); wm8962_free_beep(codec); for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) regulator_unregister_notifier(wm8962->supplies[i].consumer, diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h index 6145399..fce1199 100644 --- a/sound/soc/codecs/wm8962.h +++ b/sound/soc/codecs/wm8962.h @@ -181,6 +181,7 @@ #define WM8962_EQ39 0x175 #define WM8962_EQ40 0x176 #define WM8962_EQ41 0x177 +#define WM8962_GPIO_BASE 0x200 #define WM8962_GPIO_2 0x201 #define WM8962_GPIO_3 0x202 #define WM8962_GPIO_5 0x204 -- 1.7.1

2 1

[alsa-devel] Acer Aspire One Volume Low
by Alexander Lam 02 Oct '10

02 Oct '10

Hi all, I have an Acer Aspire one (the original one) with a realtek alc268 codec (intel hda) In windows XP, I would say that the maximum volume is somewhere around 5 times higher than in linux. I am running kernel 2.6.35.6. I actually have the spec sheet for the codec (from a google search) and the mixer ranges that are specified in the kernel seem sane,so I'm clueless as to what's happening. Maybe there's some mystery bit to set, because the windows XP Realtek driver has a "limit output" option that when turned on, makes the output be about as loud as in linux. (or maybe this option just cuts the mixer values in half...) Suggestions? Lacking that, is there a tool already written to read back the codec configuration in windows XP? -- Alexander Lam

1 0

Re: [alsa-devel] [PATCH 2/2] omap: zoom: Move new code introduced by ASoC m-c to board-zoom-peripherals
by Liam Girdwood 02 Oct '10

02 Oct '10

On Fri, 2010-09-24 at 11:31 -0500, Lopez Cruz, Misael wrote: > Hi Jarkko, > > > ASoC Multi-Component Support moves some code from sound/soc/omap/zoom2.c > > into > > arch/arm/mach-omap2/board-zoom2.c. However, that code should go to > > board-zoom-peripherals.c instead as there is common code and registration > > for zoom boards. > > > > Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> > > Cc: Vikram Pandita <vikram.pandita(a)ti.com> > > Cc: Lopez Cruz, Misael <x0052729(a)ti.com> > > Cc: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> > > Cc: Tony Lindgren <tony(a)atomide.com> > > --- > > I don't have this HW so not tested. > Tested on zoom2 using the 2 patches of your series, compilation break is > gone and audio works fine. > > Thanks, > -Misael Applied, and manually added your Tested-by: Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

3 2

[alsa-devel] [PATCH] omap: zoom2: fix build break due to ASoC changes
by Anand Gadiyar 01 Oct '10

01 Oct '10

Commit f0fba2ad1b6 (ASoC: multi-component - ASoC Multi-Component Support) in linux-next as of 20101001 broke omap builds with the following errors. CC arch/arm/mach-omap2/board-zoom2.o arch/arm/mach-omap2/board-zoom2.c:65: error: 'zoom2_bci_data' undeclared here (not in a function) arch/arm/mach-omap2/board-zoom2.c:67: error: 'zoom2_usb_data' undeclared here (not in a function) arch/arm/mach-omap2/board-zoom2.c:68: error: 'zoom2_gpio_data' undeclared here (not in a function) arch/arm/mach-omap2/board-zoom2.c:69: error: 'zoom2_kp_twl4030_data' undeclared here (not in a function) arch/arm/mach-omap2/board-zoom2.c:71: error: 'zoom2_vmmc1' undeclared here (not in a function) arch/arm/mach-omap2/board-zoom2.c:72: error: 'zoom2_vmmc2' undeclared here (not in a function) arch/arm/mach-omap2/board-zoom2.c:73: error: 'zoom2_vsim' undeclared here (not in a function) make[1]: *** [arch/arm/mach-omap2/board-zoom2.o] Error 1 make: *** [arch/arm/mach-omap2] Error 2 For zoom2/3, the i2c peripheral initialization is already done in board-zoom-peripherals.c and should not be done in board-zoom2.c. Revert all of the changes made to board-zoom2.c to fix the build. ASoC-related changes should be done separately if needed. Signed-off-by: Anand Gadiyar <gadiyar(a)ti.com> Cc: Tony Lindgren <tony(a)atomide.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Liam, The only functional change I see here is the twl4030_codec_audio_data. That should probably go into board-zoom-peripherals.c. I'll take a deeper look in a while and fix properly there. - Anand arch/arm/mach-omap2/board-zoom2.c | 28 +++------------------------- 1 file changed, 3 insertions(+), 25 deletions(-) Index: linux-2.6/arch/arm/mach-omap2/board-zoom2.c =================================================================== --- linux-2.6.orig/arch/arm/mach-omap2/board-zoom2.c +++ linux-2.6/arch/arm/mach-omap2/board-zoom2.c @@ -14,7 +14,6 @@ #include <linux/platform_device.h> #include <linux/input.h> #include <linux/gpio.h> -#include <linux/i2c/twl.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> @@ -35,11 +34,8 @@ static void __init omap_zoom2_init_irq(v omap_gpio_init(); } -/* EXTMUTE callback function */ -void zoom2_set_hs_extmute(int mute) -{ - gpio_set_value(ZOOM2_HEADSET_EXTMUTE_GPIO, mute); -} +/* REVISIT: These audio entries can be removed once MFD code is merged */ +#if 0 static struct twl4030_madc_platform_data zoom2_madc_data = { .irq_line = 1, @@ -47,9 +43,6 @@ static struct twl4030_madc_platform_data static struct twl4030_codec_audio_data zoom2_audio_data = { .audio_mclk = 26000000, - .ramp_delay_value = 3, /* 161 ms */ - .hs_extmute = 1, - .set_hs_extmute = zoom2_set_hs_extmute, }; static struct twl4030_codec_data zoom2_codec_data = { @@ -71,24 +64,10 @@ static struct twl4030_platform_data zoom .vmmc1 = &zoom2_vmmc1, .vmmc2 = &zoom2_vmmc2, .vsim = &zoom2_vsim, -}; -static struct i2c_board_info __initdata zoom2_i2c_boardinfo[] = { - { - I2C_BOARD_INFO("twl4030", 0x48), - .flags = I2C_CLIENT_WAKE, - .irq = INT_34XX_SYS_NIRQ, - .platform_data = &zoom2_twldata, - }, }; -static int __init omap3_zoom2_i2c_init(void) -{ - omap_register_i2c_bus(1, 2600, zoom2_i2c_boardinfo, - ARRAY_SIZE(zoom2_i2c_boardinfo)); - return 0; -} - +#endif #ifdef CONFIG_OMAP_MUX static struct omap_board_mux board_mux[] __initdata = { @@ -156,7 +135,6 @@ static void __init omap_zoom2_init(void) { omap3_mux_init(board_mux, OMAP_PACKAGE_CBB); zoom_peripherals_init(); - omap3_zoom2_i2c_init(); board_nand_init(zoom_nand_partitions, ARRAY_SIZE(zoom_nand_partitions), ZOOM_NAND_CS); zoom_debugboard_init();

2 2

[alsa-devel] question on ADAT S/MUX using snd-ice1712 and Alesis AI4
by Niels Mayer 01 Oct '10

01 Oct '10

Now that I have an ADAT-capable card ($20 ebay ice1712-based terratec ews88d) I'm curious... if I combine it with something like http://www.kellyindustries.com/computer/alesis_ai4.html ( http://www.alesis.com/ai4 ) and use the S/MUX mode built in to the AI4 across eight channels to create four 24/96 channels. //// //// //// //// In order for the AI-4 to operate at the 96 kHz samplerate it has to be run in S/MUX mode or sample split mode which means that you get 4 channels of conversion and not 8. This is standard and perfectly acceptable. The first two channels will be routed out to the ADAT lightpipe outputs 1 through 4 and the 2nd two channels with be routed out ADAT lightpipe outputs 5 through 8. //// //// //// //// Would I be able to "transparently" use these "S/MUX"d channels in a linux DAW, by simply recording/playing-back a higher channel count per track (e.g. 4 for a stereo track, 2 for a mono)? This seems like a good way to achieve "studio quality" 24/96 record&playback at a distance from the computer -- via ADAT cable -- using high quality outboard A/D and D/A or interfacing to external equipment already presenting AES/EBU format I/O. If a device like the AI4 actually does all the bit-splitting and other fu, both for input and output -- wouldn't it not matter that the actual 2-track or 4-track contents are essentially "noise" because nothing in Linux-land would understand the S/MUX format. Next question: to avoid the hack suggested above, is there some kind of ALSA plugin that would reconstitute/create synchronized pairs of S/MUX data on the same soundcard into single 24/96 streams, both for input and output? How is S/Mux handled in Linux & ALSA? Thanks. Niels http://nielsmayer.com PS: The Alesis AI4 seems nicer than the http://www.aphex.com/144.htm -- for one I won't need to make 110 ohm cables with XLR connectors to DB25. The AI4's support for S/MUX mode is especially nice-sounding -- if there was a way for it to work in linux. Any other suggestions for converting ADAT to AES/EBU (or spdif) w/ decent synchronization options for input?

1 0

[alsa-devel] [PATCH] ASoC: WM8940: incorrect mask for BCLKDIV
by Grzegorz Daniluk 01 Oct '10

01 Oct '10

corrected bitmask in BCLKDIV setting (wm8940_set_dai_clkdiv) and bias_level in wm8940 register function. best regards, Grzegorz Daniluk ---- diff -urN a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c --- a/sound/soc/codecs/wm8940.c 2010-04-16 11:10:10.000000000 +0000 +++ b/sound/soc/codecs/wm8940.c 2010-09-29 14:12:17.263497290 +0000 @@ -514,7 +514,7 @@ if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING - "WM8940 N value %d outwith recommended range!d\n", + "WM8940 N value %d outwith recommended range!\n", Ndiv); pll_div.n = Ndiv; @@ -604,7 +604,7 @@ switch (div_id) { case WM8940_BCLKDIV: - reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFFEF3; + reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFFE3; ret = snd_soc_write(codec, WM8940_CLOCK, reg | (div << 2)); break; case WM8940_MCLKDIV: @@ -800,9 +800,10 @@ wm8940_dai.dev = codec->dev; - wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8940_set_bias_level(codec, SND_SOC_BIAS_PREPARE); - ret = snd_soc_write(codec, WM8940_POWER1, 0x180); + reg = snd_soc_read(codec, WM8940_POWER1); + ret = snd_soc_write(codec, WM8940_POWER1, reg | 0x180); if (ret < 0) return ret;

2 1

[alsa-devel] Creative Sound Blaster X-Fi Titanium PCIe support
by nidujay 01 Oct '10

01 Oct '10

Hi all, I've got a Creative Sound Blaster X-Fi Titanium PCIe sound card which I purchased for the following 2 reasons: 1. It supports sound fonts 2. I require an SPDIF optical input My OS is UbuntuStudio 10.04 which I believe uses alsa version 1.0.22. Unfortunately, I haven't been able to get either of the above working (Ok I haven't investigated (2) very deeply yet). Any idea what I can do? P.S. I'm willing to get my hands dirty if there's actual development effort required. Thanks D

3 3

[alsa-devel] namehint: Evaluate more possibilities for hw devices
by Stas Sergeev 01 Oct '10

01 Oct '10

Hi Jaroslav. I updated the alsa-lib from git, and "aplay -L" no longer shows some PCMs it used to list before. I tracked this down to this commit: --- commit 97d25c0766405453d3c298bd51dd4916121a4f6a Author: Jaroslav Kysela <perex(a)perex.cz> Date: Fri Aug 20 10:31:17 2010 +0200 namehint: Evaluate more possibilities for hw devices --- Attached are the outputs before and after the commit. So why some devices are now missing?

3 10

[alsa-devel] refreshing the reg cache in asoc driver
by lin_g 01 Oct '10

01 Oct '10

Hi, I'm writing the driver for wm8804 s/pdif transceiver. However, my problem is that I would like to refresh my register cache inside the driver by periodically reading the values stored in device's registers. There is an interrupt handler fsi_interrupt() (in fsi.c) called when codec receives some data, but I suppose I need to find something closer to the 'codec layer'. Or maybe I'm just thinking wrongly and there is much simpler way of doing this. Appreciate any advice. Thank you very much, best regards, Greg

3 2