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[PATCH] CS4270: bring feature/limitations list in sync
by Daniel Mack 08 Mar '09

08 Mar '09

Removes numbers from the list of features/limitations and makes it reflect recent changes to the code. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/soc/codecs/cs4270.c | 15 +++++++-------- 1 files changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 4c7e546..d16aa81 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -12,14 +12,13 @@ * * Current features/limitations: * - * 1) Software mode is supported. Stand-alone mode is not supported. - * 2) Only I2C is supported, not SPI - * 3) Only Master mode is supported, not Slave. - * 4) The machine driver's 'startup' function must call - * cs4270_set_dai_sysclk() with the value of MCLK. - * 5) Only I2S and left-justified modes are supported - * 6) Power management is not supported - * 7) The only supported control is volume and hardware mute (if enabled) + * - Software mode is supported. Stand-alone mode is not supported. + * - Only I2C is supported, not SPI + * - Support for master and slave mode + * - The machine driver's 'startup' function must call + * cs4270_set_dai_sysclk() with the value of MCLK. + * - Only I2S and left-justified modes are supported + * - Power management is not supported */ #include <linux/module.h> -- 1.6.1.3

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Re: [alsa-devel] [PATCH 1/1] ALSA: ASoC: Davinci: Fix incorrect machine type for SFFSDR board
by Kevin Hilman 08 Mar '09

08 Mar '09

Hugo Villeneuve <hugo(a)hugovil.com> writes: > Signed-off-by: Hugo Villeneuve <hugo(a)hugovil.com> Signed-off-by: Kevin Hilman <khilman(a)deeprootsystems.com> This has also been pushed to DaVinci git until it get merged into the asoc base. Kevin > --- > sound/soc/davinci/Kconfig | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig > index 7d3a85d..411a710 100644 > --- a/sound/soc/davinci/Kconfig > +++ b/sound/soc/davinci/Kconfig > @@ -21,7 +21,7 @@ config SND_DAVINCI_SOC_EVM > > config SND_DAVINCI_SOC_SFFSDR > tristate "SoC Audio support for SFFSDR" > - depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR > + depends on SND_DAVINCI_SOC && MACH_SFFSDR > select SND_DAVINCI_SOC_I2S > select SND_SOC_PCM3008 > select SFFSDR_FPGA > -- > 1.5.4.5 > > > _______________________________________________ > Davinci-linux-open-source mailing list > Davinci-linux-open-source(a)linux.davincidsp.com > http://linux.davincidsp.com/mailman/listinfo/davinci-linux-open-source

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Re: [alsa-devel] [PATCH 1/1] ALSA: ASoC: Davinci: Fix incorrect machine type for SFFSDR board
by Kevin Hilman 08 Mar '09

08 Mar '09

Hugo Villeneuve <hugo(a)hugovil.com> writes: > Signed-off-by: Hugo Villeneuve <hugo(a)hugovil.com> > --- > sound/soc/davinci/Kconfig | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig > index 7d3a85d..411a710 100644 > --- a/sound/soc/davinci/Kconfig > +++ b/sound/soc/davinci/Kconfig > @@ -21,7 +21,7 @@ config SND_DAVINCI_SOC_EVM > > config SND_DAVINCI_SOC_SFFSDR > tristate "SoC Audio support for SFFSDR" > - depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR > + depends on SND_DAVINCI_SOC && MACH_SFFSDR > select SND_DAVINCI_SOC_I2S > select SND_SOC_PCM3008 > select SFFSDR_FPGA Thanks, pushed. Kevin

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[alsa-devel] [PATCH 2.6.30] ASoC: Improve pause/unpause performance in Freescale 8610 drivers
by Timur Tabi 07 Mar '09

07 Mar '09

Add support for true pause and unpause. Without this, mplayer will drop some audio (less than one second, but still noticeable) when pausing playback. Remove support for PM suspend and resume from the trigger function, since the driver doesn't support PM anyway. Optimize the delay after starting capture. Instead of delaying 1ms, the driver now polls the hardware. The new delay is shorter by over 90% yet still effective. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 3 ++- sound/soc/fsl/fsl_ssi.c | 23 ++++++++++++++--------- 2 files changed, 16 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 58a3fa4..b3eb857 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -142,7 +142,8 @@ static const struct snd_pcm_hardware fsl_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_JOINT_DUPLEX, + SNDRV_PCM_INFO_JOINT_DUPLEX | + SNDRV_PCM_INFO_PAUSE, .formats = FSLDMA_PCM_FORMATS, .rates = FSLDMA_PCM_RATES, .rate_min = 5512, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8cb6bcf..b7733e6 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -464,28 +464,33 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); } else { - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + long timeout = jiffies + 10; + setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); - /* - * I think we need this delay to allow time for the SSI - * to put data into its FIFO. Without it, ALSA starts - * to complain about overruns. + /* Wait until the SSI has filled its FIFO. Without this + * delay, ALSA complains about overruns. When the FIFO + * is full, the DMA controller initiates its first + * transfer. Until then, however, the DMA's DAR + * register is zero, which translates to an + * out-of-bounds pointer. This makes ALSA think an + * overrun has occurred. */ - mdelay(1); + while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) && + (jiffies < timeout)); + if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0)) + return -EIO; } break; case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) clrbits32(&ssi->scr, CCSR_SSI_SCR_TE); -- 1.6.1.3

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[alsa-devel] No headphone output on HT Omega Claro Halo
by Christopher J. Atherton 07 Mar '09

07 Mar '09

I have an HT Omega Claro Halo (CMI8878) card that is not entirely working. This card has two analog outputs, a pair of RCA connectors and also a separate headphone output (3.5mm jack). The RCA outputs work, but the headphone output, which notably passes a signal from a built in headphone amp, does not. Interestingly, when I turn up the master volume really high, I can hear pops/cracks on the headphone output. Thanks for any assistance. -- Kristoff

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[alsa-devel] [PATCH 1/1] ALSA: ASoC: Davinci: Fix incorrect machine type for SFFSDR board
by Hugo Villeneuve 06 Mar '09

06 Mar '09

Signed-off-by: Hugo Villeneuve <hugo(a)hugovil.com> --- sound/soc/davinci/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 7d3a85d..411a710 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -21,7 +21,7 @@ config SND_DAVINCI_SOC_EVM config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR + depends on SND_DAVINCI_SOC && MACH_SFFSDR select SND_DAVINCI_SOC_I2S select SND_SOC_PCM3008 select SFFSDR_FPGA -- 1.5.4.5

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[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 06 Mar '09

06 Mar '09

The following changes since commit 5f2a9384a9291d898b4bf85c4fbf497eef582977: Philipp Zabel (1): ASoC: UDA1380: DATAI is slave only are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Ben Dooks (6): S3C24XX: Move and update IIS headers S3C: Move <mach/audio.h> to <plat/audio.h> ASoC: Add JIVE audio support ASoC: Split s3c2412-i2s.c into core and SoC specific parts ASoC: Add s3c64xx-i2s support ASoC: Select DMA if I2S is configured Daniel Mack (2): ASoC: Fix name of register bit in pxa-ssp ASoC: add two more bitfields for PXA SSP Jonas Andersson (1): ASoC: wm8510 pll settings Lopez Cruz, Misael (3): ASoC: Add GPIO support for jack reporting interface ASoC: Add DAPM machine widgets to SDP3430 driver ASoC: Add headset jack detection for SDP3430 machine driver Mark Brown (5): ASoC: Refresh JIVE driver ASoC: Factor out DAPM widget power check into separate function ASoC: Fix memory allocation for snd_soc_dapm_switch names ASoC: Update Kconfig for Samsung CPUs to reflect S3C64xx support ASoC: Staticise workqueue function for GPIO jack detection Mike Frysinger (2): ASoC: Blackfin: move gpio_err behind the define that is only user of it ASoC: Blackfin: fix typo in MUTE definition Peter Ujfalusi (1): ASoC: TWL4030: Make the HS ramp delay configurable Timur Tabi (1): ASoC: add support for SSI asynchronous mode to the Freescale SSI drivers arch/arm/mach-pxa/include/mach/regs-ssp.h | 7 +- arch/arm/mach-s3c2410/dma.c | 2 +- arch/arm/mach-s3c2410/include/mach/hardware.h | 3 - arch/arm/mach-s3c2410/include/mach/io.h | 2 +- arch/arm/mach-s3c2412/dma.c | 4 +- arch/arm/mach-s3c2440/dma.c | 2 +- arch/arm/mach-s3c2443/dma.c | 2 +- arch/arm/mach-shark/include/mach/io.h | 2 +- .../include/mach => plat-s3c/include/plat}/audio.h | 0 .../arm/plat-s3c/include/plat}/regs-s3c2412-iis.h | 3 + arch/arm/plat-s3c24xx/clock-dclk.c | 1 + .../arm/plat-s3c24xx/include/plat}/regs-iis.h | 0 include/sound/soc.h | 32 + sound/soc/atmel/playpaq_wm8510.c | 24 +- sound/soc/blackfin/bf5xx-ac97.c | 2 +- sound/soc/codecs/ad73311.h | 2 +- sound/soc/codecs/twl4030.c | 18 +- sound/soc/codecs/wm8510.c | 4 +- sound/soc/fsl/fsl_ssi.c | 42 +- sound/soc/fsl/fsl_ssi.h | 2 + sound/soc/fsl/mpc8610_hpcd.c | 5 + sound/soc/omap/sdp3430.c | 105 ++++- sound/soc/pxa/pxa-ssp.c | 6 +- sound/soc/s3c24xx/Kconfig | 29 +- sound/soc/s3c24xx/Makefile | 6 + sound/soc/s3c24xx/jive_wm8750.c | 201 ++++++ sound/soc/s3c24xx/neo1973_wm8753.c | 4 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 645 ++++++++++++++++++++ sound/soc/s3c24xx/s3c-i2s-v2.h | 90 +++ sound/soc/s3c24xx/s3c2412-i2s.c | 604 +------------------ sound/soc/s3c24xx/s3c2412-i2s.h | 17 +- sound/soc/s3c24xx/s3c2443-ac97.c | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 4 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 220 +++++++ sound/soc/s3c24xx/s3c64xx-i2s.h | 31 + sound/soc/soc-dapm.c | 265 +++++---- sound/soc/soc-jack.c | 129 ++++ 39 files changed, 1745 insertions(+), 776 deletions(-) rename arch/arm/{mach-s3c2410/include/mach => plat-s3c/include/plat}/audio.h (100%) rename {include/asm-arm/plat-s3c24xx => arch/arm/plat-s3c/include/plat}/regs-s3c2412-iis.h (96%) rename {include/asm-arm/plat-s3c24xx => arch/arm/plat-s3c24xx/include/plat}/regs-iis.h (100%) create mode 100644 sound/soc/s3c24xx/jive_wm8750.c create mode 100644 sound/soc/s3c24xx/s3c-i2s-v2.c create mode 100644 sound/soc/s3c24xx/s3c-i2s-v2.h create mode 100644 sound/soc/s3c24xx/s3c64xx-i2s.c create mode 100644 sound/soc/s3c24xx/s3c64xx-i2s.h

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[alsa-devel] [PATCH] ASoC: add support for SSI asynchronous mode to the Freescale SSI drivers
by Timur Tabi 06 Mar '09

06 Mar '09

Add a new device tree property for the SSI node: "fsl,ssi-asynchronous". If defined, the SSI is programmed into asynchronous mode, otherwise it is programmed into synchronous mode. In asynchronous mode, pin SRCK must be connected to the same clock source as STFS, and pin SRFS must be connected to the same signal as STFS. Asynchronous mode allows playback and capture to use different sample sizes. It also technically allows different sample rates, but the driver does not support that. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 42 ++++++++++++++++++++++++++++++++---------- sound/soc/fsl/fsl_ssi.h | 2 ++ sound/soc/fsl/mpc8610_hpcd.c | 5 +++++ 3 files changed, 39 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6844009..8cb6bcf 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -72,6 +72,7 @@ * @dev: struct device pointer * @playback: the number of playback streams opened * @capture: the number of capture streams opened + * @asynchronous: 0=synchronous mode, 1=asynchronous mode * @cpu_dai: the CPU DAI for this device * @dev_attr: the sysfs device attribute structure * @stats: SSI statistics @@ -86,6 +87,7 @@ struct fsl_ssi_private { struct device *dev; unsigned int playback; unsigned int capture; + int asynchronous; struct snd_soc_dai cpu_dai; struct device_attribute dev_attr; @@ -301,9 +303,10 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * * FIXME: Little-endian samples require a different shift dir */ - clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK, - CCSR_SSI_SCR_TFR_CLK_DIS | - CCSR_SSI_SCR_I2S_MODE_SLAVE | CCSR_SSI_SCR_SYN); + clrsetbits_be32(&ssi->scr, + CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, + CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE + | (ssi_private->asynchronous ? 0 : CCSR_SSI_SCR_SYN)); out_be32(&ssi->stcr, CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -382,10 +385,15 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_RATE, first_runtime->rate, first_runtime->rate); - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - first_runtime->sample_bits, - first_runtime->sample_bits); + /* If we're in synchronous mode, then we need to constrain + * the sample size as well. We don't support independent sample + * rates in asynchronous mode. + */ + if (!ssi_private->asynchronous) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + first_runtime->sample_bits, + first_runtime->sample_bits); ssi_private->second_stream = substream; } @@ -421,13 +429,18 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, struct ccsr_ssi __iomem *ssi = ssi_private->ssi; unsigned int sample_size = snd_pcm_format_width(params_format(hw_params)); - u32 wl; + u32 wl = CCSR_SSI_SxCCR_WL(sample_size); /* The SSI should always be disabled at this points (SSIEN=0) */ - wl = CCSR_SSI_SxCCR_WL(sample_size); /* In synchronous mode, the SSI uses STCCR for capture */ - clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); + if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || + !ssi_private->asynchronous) + clrsetbits_be32(&ssi->stccr, + CCSR_SSI_SxCCR_WL_MASK, wl); + else + clrsetbits_be32(&ssi->srccr, + CCSR_SSI_SxCCR_WL_MASK, wl); } return 0; @@ -653,6 +666,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) ssi_private->ssi_phys = ssi_info->ssi_phys; ssi_private->irq = ssi_info->irq; ssi_private->dev = ssi_info->dev; + ssi_private->asynchronous = ssi_info->asynchronous; ssi_private->dev->driver_data = fsl_ssi_dai; @@ -703,6 +717,14 @@ void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai) } EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai); +static int __init fsl_ssi_init(void) +{ + printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n"); + + return 0; +} +module_init(fsl_ssi_init); + MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 83b44d7..eade01f 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -208,6 +208,7 @@ struct ccsr_ssi { * ssi_phys: physical address of the SSI registers * irq: IRQ of this SSI * dev: struct device, used to create the sysfs statistics file + * asynchronous: 0=synchronous mode, 1=asynchronous mode */ struct fsl_ssi_info { unsigned int id; @@ -215,6 +216,7 @@ struct fsl_ssi_info { dma_addr_t ssi_phys; unsigned int irq; struct device *dev; + int asynchronous; }; struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index acf39a6..ef67d1c 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -353,6 +353,11 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, } ssi_info.irq = machine_data->ssi_irq; + /* Do we want to use asynchronous mode? */ + ssi_info.asynchronous = + of_find_property(np, "fsl,ssi-asynchronous", NULL) ? 1 : 0; + if (ssi_info.asynchronous) + dev_info(&ofdev->dev, "using asynchronous mode\n"); /* Map the global utilities registers. */ guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); -- 1.6.1.3

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[alsa-devel] [PATCH 2/2] ASoC: Add headset jack detection for SDP3430 machine driver
by Lopez Cruz, Misael 06 Mar '09

06 Mar '09

Add headset jack detection for SDP3430 boards using SoC jack reporting interface. Headset detection on SDP3430 board is achieved through TWL4030 GPIO_2 pin. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/omap/sdp3430.c | 43 +++++++++++++++++++++++++++++++++++++++++-- 1 files changed, 41 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 4eab4b4..715c648 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -28,6 +28,7 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/jack.h> #include <asm/mach-types.h> #include <mach/hardware.h> @@ -122,7 +123,7 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) /* SDP3430 connected pins */ snd_soc_dapm_enable_pin(codec, "Ext Mic"); snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_enable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); /* TWL4030 not connected pins */ snd_soc_dapm_nc_pin(codec, "AUXL"); @@ -144,6 +145,27 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) return ret; } +/* Headset jack */ +static struct snd_soc_jack hs_jack; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Jack", + .mask = SND_JACK_HEADSET, + }, +}; + +/* Headset jack detection gpios */ +static struct snd_soc_jack_gpio hs_jack_gpios[] = { + { + .gpio = (OMAP_MAX_GPIO_LINES + 2), + .name = "hsdet-gpio", + .report = SND_JACK_HEADSET, + .debounce_time = 200, + }, +}; + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", @@ -194,7 +216,21 @@ static int __init sdp3430_soc_init(void) if (ret) goto err1; - return 0; + /* Headset jack detection */ + ret = snd_soc_jack_new(&snd_soc_sdp3430, "SDP3430 headset jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + + return ret; err1: printk(KERN_ERR "Unable to add platform device\n"); @@ -206,6 +242,9 @@ module_init(sdp3430_soc_init); static void __exit sdp3430_soc_exit(void) { + snd_soc_jack_free_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + platform_device_unregister(sdp3430_snd_device); } module_exit(sdp3430_soc_exit); -- 1.5.4.3

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[alsa-devel] ALC889A (Gigabyte EX58-UD5) pin support
by Jo Shields 06 Mar '09

06 Mar '09

I've got a pretty new Gigabyte board, using snd-hda-intel and an ALC889A chip. Whilst the chip seems to be detected fine, I have no functional audio out - no sound on any of the ports. dmesg shows that ALSA has no idea about the specific model: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 hda_codec: Unknown model for ALC883, trying auto-probe from BIOS... If I'm reading the source correctly, the default pin layout is 3stack-digout. I tried 6stack-digout, but it didn't help. Linked is output from the diagnostic script: http://www.alsa-project.org/db/?f=3938bc16481f549203e373e303c4b1c1dfb303dd I'm happy enough to try things out in an effort to get the onboard sound to behave

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