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[alsa-devel] [RESEND PATCH v2 1/2] ASoC: samsung: s3c24{xx, 12}-i2s: port to use generic dmaengine API
by Vasily Khoruzhick 23 Jun '14

23 Jun '14

Use dmaengine instead of legacy s3c24xx DMA API for s3c24xx and s3c2412 Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- v2: use hardcoded dma channel number sound/soc/samsung/Kconfig | 9 +++---- sound/soc/samsung/dmaengine.c | 3 +++ sound/soc/samsung/s3c-i2s-v2.c | 17 +------------ sound/soc/samsung/s3c2412-i2s.c | 47 +++++++++++++++++----------------- sound/soc/samsung/s3c24xx-i2s.c | 56 +++++++++++++++++++---------------------- 5 files changed, 56 insertions(+), 76 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 3507574..022c7a4 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,11 +1,10 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" depends on PLAT_SAMSUNG - select S3C2410_DMA if ARCH_S3C24XX + select S3C24XX_DMAC if ARCH_S3C24XX select S3C64XX_PL080 if ARCH_S3C64XX - select SND_S3C_DMA if !ARCH_S3C24XX - select SND_S3C_DMA_LEGACY if ARCH_S3C24XX - select SND_SOC_GENERIC_DMAENGINE_PCM if !ARCH_S3C24XX + select SND_S3C_DMA + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the Samsung SoCs' Audio interfaces. You will also need to @@ -19,7 +18,6 @@ config SND_S3C_DMA_LEGACY config SND_S3C24XX_I2S tristate - select S3C24XX_DMA config SND_S3C_I2SV2_SOC tristate @@ -27,7 +25,6 @@ config SND_S3C_I2SV2_SOC config SND_S3C2412_SOC_I2S tristate select SND_S3C_I2SV2_SOC - select S3C2410_DMA config SND_SAMSUNG_PCM tristate diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index 750ce58..72f27d1 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -17,6 +17,7 @@ #include <linux/module.h> #include <linux/amba/pl08x.h> +#include <linux/platform_data/dma-s3c24xx.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,8 @@ #ifdef CONFIG_ARCH_S3C64XX #define filter_fn pl08x_filter_id +#elif defined(CONFIG_ARCH_S3C24XX) +#define filter_fn s3c24xx_dma_filter #else #define filter_fn NULL #endif diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 79e7efb..1a74051 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -392,8 +392,6 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); pr_debug("Entered %s\n", __func__); @@ -424,13 +422,6 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, local_irq_restore(irqs); - /* - * Load the next buffer to DMA to meet the reqirement - * of the auto reload mechanism of S3C24XX. - * This call won't bother S3C64XX. - */ - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - break; case SNDRV_PCM_TRIGGER_STOP: @@ -644,12 +635,6 @@ int s3c_i2sv2_probe(struct snd_soc_dai *dai, /* record our i2s structure for later use in the callbacks */ snd_soc_dai_set_drvdata(dai, i2s); - i2s->regs = ioremap(base, 0x100); - if (i2s->regs == NULL) { - dev_err(dev, "cannot ioremap registers\n"); - return -ENXIO; - } - i2s->iis_pclk = clk_get(dev, "iis"); if (IS_ERR(i2s->iis_pclk)) { dev_err(dev, "failed to get iis_clock\n"); @@ -729,7 +714,7 @@ int s3c_i2sv2_register_component(struct device *dev, int id, struct snd_soc_component_driver *cmp_drv, struct snd_soc_dai_driver *dai_drv) { - struct snd_soc_dai_ops *ops = dai_drv->ops; + struct snd_soc_dai_ops *ops = (struct snd_soc_dai_ops *)dai_drv->ops; ops->trigger = s3c2412_i2s_trigger; if (!ops->hw_params) diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index d079445..ba084e3 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -33,25 +33,19 @@ #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" -static struct s3c2410_dma_client s3c2412_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c2412_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { - .client = &s3c2412_dma_client_out, + .client = + (struct s3c2410_dma_client *)&s3c2412_i2s_pcm_stereo_out, .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, + .ch_name = "tx", .dma_size = 4, }; static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { - .client = &s3c2412_dma_client_in, + .client = + (struct s3c2410_dma_client *)&s3c2412_i2s_pcm_stereo_in, .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, + .ch_name = "rx", .dma_size = 4, }; @@ -63,6 +57,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) pr_debug("Entered %s\n", __func__); + samsung_asoc_init_dma_data(dai, &s3c2412_i2s_pcm_stereo_out, + &s3c2412_i2s_pcm_stereo_in); + ret = s3c_i2sv2_probe(dai, &s3c2412_i2s, S3C2410_PA_IIS); if (ret) return ret; @@ -70,10 +67,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in; s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; - s3c2412_i2s.iis_cclk = clk_get(dai->dev, "i2sclk"); + s3c2412_i2s.iis_cclk = devm_clk_get(dai->dev, "i2sclk"); if (IS_ERR(s3c2412_i2s.iis_cclk)) { pr_err("failed to get i2sclk clock\n"); - iounmap(s3c2412_i2s.regs); return PTR_ERR(s3c2412_i2s.iis_cclk); } @@ -94,8 +90,6 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) static int s3c2412_i2s_remove(struct snd_soc_dai *dai) { clk_disable(s3c2412_i2s.iis_cclk); - clk_put(s3c2412_i2s.iis_cclk); - iounmap(s3c2412_i2s.regs); return 0; } @@ -105,18 +99,10 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct s3c_i2sv2_info *i2s = snd_soc_dai_get_drvdata(cpu_dai); - struct s3c_dma_params *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - iismod = readl(i2s->regs + S3C2412_IISMOD); pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); @@ -169,6 +155,19 @@ static const struct snd_soc_component_driver s3c2412_i2s_component = { static int s3c2412_iis_dev_probe(struct platform_device *pdev) { int ret = 0; + struct resource *res; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "Can't get IO resource.\n"); + return -ENOENT; + } + s3c2412_i2s.regs = devm_ioremap_resource(&pdev->dev, res); + if (s3c2412_i2s.regs == NULL) + return -ENXIO; + + s3c2412_i2s_pcm_stereo_out.dma_addr = res->start + S3C2412_IISTXD; + s3c2412_i2s_pcm_stereo_in.dma_addr = res->start + S3C2412_IISRXD; ret = s3c_i2sv2_register_component(&pdev->dev, -1, &s3c2412_i2s_component, diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index f31e916..97600df 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -31,25 +31,19 @@ #include "dma.h" #include "s3c24xx-i2s.h" -static struct s3c2410_dma_client s3c24xx_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c24xx_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { - .client = &s3c24xx_dma_client_out, + .client = + (struct s3c2410_dma_client *)&s3c24xx_i2s_pcm_stereo_out, .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .ch_name = "tx", .dma_size = 2, }; static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { - .client = &s3c24xx_dma_client_in, + .client = + (struct s3c2410_dma_client *)&s3c24xx_i2s_pcm_stereo_in, .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .ch_name = "rx", .dma_size = 2, }; @@ -231,18 +225,12 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c24xx_i2s_pcm_stereo_out; - else - dma_data = &s3c24xx_i2s_pcm_stereo_in; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); + dma_data = snd_soc_dai_get_dma_data(dai, substream); /* Working copies of register */ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -251,11 +239,11 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: iismod &= ~S3C2410_IISMOD_16BIT; - dma_data->dma_size = 1; + dma_data->addr_width = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; - dma_data->dma_size = 2; + dma_data->addr_width = 2; break; default: return -EINVAL; @@ -270,8 +258,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { int ret = 0; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(dai, substream); pr_debug("Entered %s\n", __func__); @@ -290,7 +276,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, else s3c24xx_snd_txctrl(1); - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: @@ -380,14 +365,12 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) { pr_debug("Entered %s\n", __func__); - s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); - if (s3c24xx_i2s.regs == NULL) - return -ENXIO; + samsung_asoc_init_dma_data(dai, &s3c24xx_i2s_pcm_stereo_out, + &s3c24xx_i2s_pcm_stereo_in); - s3c24xx_i2s.iis_clk = clk_get(dai->dev, "iis"); + s3c24xx_i2s.iis_clk = devm_clk_get(dai->dev, "iis"); if (IS_ERR(s3c24xx_i2s.iis_clk)) { pr_err("failed to get iis_clock\n"); - iounmap(s3c24xx_i2s.regs); return PTR_ERR(s3c24xx_i2s.iis_clk); } clk_enable(s3c24xx_i2s.iis_clk); @@ -474,6 +457,19 @@ static const struct snd_soc_component_driver s3c24xx_i2s_component = { static int s3c24xx_iis_dev_probe(struct platform_device *pdev) { int ret = 0; + struct resource *res; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "Can't get IO resource.\n"); + return -ENOENT; + } + s3c24xx_i2s.regs = devm_ioremap_resource(&pdev->dev, res); + if (s3c24xx_i2s.regs == NULL) + return -ENXIO; + + s3c24xx_i2s_pcm_stereo_out.dma_addr = res->start + S3C2410_IISFIFO; + s3c24xx_i2s_pcm_stereo_in.dma_addr = res->start + S3C2410_IISFIFO; ret = snd_soc_register_component(&pdev->dev, &s3c24xx_i2s_component, &s3c24xx_i2s_dai, 1); -- 1.9.3

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[alsa-devel] [PATCH 1/2] SOUND: TLV320AIC32x4: Fixed Coding style Issues
by Shahina Shaik 22 Jun '14

22 Jun '14

From: Shahina Shaik <sharab.shaik(a)gmail.com> Fixed a brace coding style issue in the tlv320aic32x4.c Signed-off-by: Shahina Shaik <sharab.shaik(a)gmail.com> --- sound/soc/codecs/tlv320aic32x4.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 1d9b117..1e50f15 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -626,21 +626,18 @@ static int aic32x4_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN | AIC32X4_MICBIAS_2075V); } - if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) { + if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); - } tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ? AIC32X4_LDOCTLEN : 0; snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg); tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE); - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) { + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) tmp_reg |= AIC32X4_LDOIN_18_36; - } - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) { + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) tmp_reg |= AIC32X4_LDOIN2HP; - } snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg); /* Mic PGA routing */ -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: fsl: refine DMA/FIQ dependencies
by Arnd Bergmann 22 Jun '14

22 Jun '14

Commit 31ee2bfd724ab ("ASoC: fsl: select SND_SOC_IMX_PCM_DMA where needed") started selecting SND_SOC_IMX_PCM_DMA and SND_SOC_IMX_PCM_FIQ for two drivers when building for i.MX. This has turned out too aggressive, as FIQ is only available for i.mx2 through i.mx5, but not i.mx6 or vybrid. Further, two more drivers have become user-selectable in the meantime, and they both depend on DMA for the imx platform as well. This changes the selection of FIQ to depend on the TZIC or AVIC interrupt controllers that actually export the imx specific FIQ interfaces, and adds the missing select statements for SAI and ESAI. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3793362..52bbe9f 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -5,6 +5,7 @@ comment "Common SoC Audio options for Freescale CPUs:" config SND_SOC_FSL_SAI tristate "Synchronous Audio Interface (SAI) module support" select REGMAP_MMIO + select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y if you want to add Synchronous Audio Interface (SAI) @@ -15,7 +16,7 @@ config SND_SOC_FSL_SAI config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC + select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) select REGMAP_MMIO help Say Y if you want to add Synchronous Serial Interface (SSI) @@ -27,7 +28,7 @@ config SND_SOC_FSL_SPDIF tristate "Sony/Philips Digital Interface module support" select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC + select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) help Say Y if you want to add Sony/Philips Digital Interface (SPDIF) support for the Freescale CPUs. @@ -37,6 +38,7 @@ config SND_SOC_FSL_SPDIF config SND_SOC_FSL_ESAI tristate "Enhanced Serial Audio Interface (ESAI) module support" select REGMAP_MMIO + select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_FSL_UTILS help Say Y if you want to add Enhanced Synchronous Audio Interface

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[alsa-devel] [PATCH v5 0/3] Add ES8328 audio codec
by Sean Cross 21 Jun '14

21 Jun '14

This patchset adds support for the Everest Semi ES8328 audio codec. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the speaker amps and as well as supporting headphone jacks. This requires support for DT clock reparenting in order to get the codec its 22.5792 MHz supply frequency. Changes since v4: - Simplify clock handling and use assigned-clock-parents - Move the codec regulator from the machine driver to the codec - Specify all four regulators, and require their presence - Fix audio recording, such that it enables the PGA Changes since v3: - Add OF bindings for everest,es8328 - Add back in patch to add everest as a vendor prefix - Give clocks more generic names - Document required clocks in DT bindings document Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 24 + .../devicetree/bindings/sound/imx-audio-es8328.txt | 63 ++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 60 ++ sound/soc/codecs/es8328-spi.c | 49 ++ sound/soc/codecs/es8328.c | 694 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 313 ++++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 263 ++++++++ 12 files changed, 1502 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

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Re: [alsa-devel] [PATCH] ASoC: Fix SOC_DOUBLE_R_SX_TLV volume mixer arguments
by Brian Austin 21 Jun '14

21 Jun '14

On Fri, 20 Jun 2014, Ryan Harvey wrote: > Remove unnecessary bit shifts. Correct min value to match datasheet. #Steps equals number of steps between min and max. > > This patch depends on 8f0367b53d887726b1f4b0b39e0f09b7cca2f79f > ASoC: cs42l56: Fix stereo channel register assignment for Headphone and > LineOut volume mixers. > > Signed-off-by: Ryan Harvey <ryan.harvey(a)cirrus.com> > --- Acked-by: Brian Austin <brian.austin(a)cirrus.com> Thanks,

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[alsa-devel] [PATCH] ASoC: wm5110: Power both channels for differential mono output
by Charles Keepax 21 Jun '14

21 Jun '14

On the wm5110 CODEC both the left and right channel must be powered when an output is being used as a mono output, although no audio is routed to the right output channel. This patch adds additional DAPM routes to link the right channel to the left in the case where an output is marked as mono. Audio must always be brought in on the left channel for mono operation. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 25 +++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 1 + 3 files changed, 27 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 41b56ee..d441d80 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -243,6 +243,31 @@ int arizona_init_spk(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(arizona_init_spk); +static const struct snd_soc_dapm_route arizona_mono_routes[] = { + { "OUT1R", NULL, "OUT1L" }, + { "OUT2R", NULL, "OUT2L" }, + { "OUT3R", NULL, "OUT3L" }, + { "OUT4R", NULL, "OUT4L" }, + { "OUT5R", NULL, "OUT5L" }, + { "OUT6R", NULL, "OUT6L" }, +}; + +int arizona_init_mono(struct snd_soc_codec *codec) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int i; + + for (i = 0; i < ARIZONA_MAX_OUTPUT; ++i) { + if (arizona->pdata.out_mono[i]) + snd_soc_dapm_add_routes(&codec->dapm, + &arizona_mono_routes[i], 1); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_mono); + int arizona_init_gpio(struct snd_soc_codec *codec) { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 05ae17f..942cfb1 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -249,6 +249,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source, extern int arizona_init_spk(struct snd_soc_codec *codec); extern int arizona_init_gpio(struct snd_soc_codec *codec); +extern int arizona_init_mono(struct snd_soc_codec *codec); extern int arizona_init_dai(struct arizona_priv *priv, int dai); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2e5fcb5..62ef544 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1596,6 +1596,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_spk(codec); arizona_init_gpio(codec); + arizona_init_mono(codec); ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8); if (ret != 0) -- 1.7.2.5

2 1

[alsa-devel] [PATCH 1/4] ASoC: rt5677: Replace the string "micbias1" to "MICBIAS1"
by Oder Chiou 21 Jun '14

21 Jun '14

The patch replaces the string "micbias1" to "MICBIAS1". Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 833231e..75268a7 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1685,7 +1685,7 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { /* Input Side */ /* micbias */ - SND_SOC_DAPM_SUPPLY("micbias1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT, + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT, 0, rt5677_set_micbias1_event, SND_SOC_DAPM_POST_PMU), /* Input Lines */ -- 1.8.1.1.439.g50a6b54

2 4

[alsa-devel] [PATCH 1/3] ASoC: rt5640: Remove unneeded goto in rt5640_i2c_probe
by Axel Lin 21 Jun '14

21 Jun '14

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5640.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index de80e89..6bc6efd 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2215,14 +2215,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, rt5640->hp_mute = 1; - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, - rt5640_dai, ARRAY_SIZE(rt5640_dai)); - if (ret < 0) - goto err; - - return 0; -err: - return ret; + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, + rt5640_dai, ARRAY_SIZE(rt5640_dai)); } static int rt5640_i2c_remove(struct i2c_client *i2c) -- 1.8.3.2

2 5

[alsa-devel] [PATCH] ASoC: pcm: fix dpcm_path_put in dpcm runtime update
by Qiao Zhou 21 Jun '14

21 Jun '14

we need to release dapm widget list after dpcm_path_get in soc_dpcm_runtime_update. otherwise, there will be potential memory leak. add dpcm_path_put to fix it. Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/soc-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a391de0..3bb8b80 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2072,6 +2072,7 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); } + dpcm_path_put(&list); capture: /* skip if FE doesn't have capture capability */ if (!fe->cpu_dai->driver->capture.channels_min) -- 1.7.9.5

2 2

[alsa-devel] [PATCH 0/2] Couple of fixes to davinci-mcasp driver
by Jyri Sarha 21 Jun '14

21 Jun '14

These changes target to sii9022 HDMI audio support, but are independent fixes. Jyri Sarha (2): ASoC: davinci-mcasp: Add dependecy to SND_DAVINCI_SOC or SND_OMAP_SOC ASoC: davinci-mcasp: Allow best effort in selecting BCLK divider sound/soc/davinci/Kconfig | 1 + sound/soc/davinci/davinci-mcasp.c | 22 +++++++++++++++++----- 2 files changed, 18 insertions(+), 5 deletions(-) -- 1.7.9.5

2 4

Re: [alsa-devel] [PATCH] ASoC: cs42l56: Fix stereo channel register assignment for Headphone and LineOut volume mixers
by Brian Austin 20 Jun '14

20 Jun '14

On Fri, 20 Jun 2014, Ryan Harvey wrote: > Stereo Headphone and LineOut volume mixers are now attached to HPA+HPB, LOA+LOB. > > Signed-off-by: Ryan Harvey <ryan.harvey(a)cirrus.com> > --- Acked-by: Brian Austin <brian.austin(a)cirrus.com>

1 0

[alsa-devel] [PATCH Resend 1/1] ASoC: mc13783: Add missing of_node_put
by Sachin Kamat 20 Jun '14

20 Jun '14

of_get_child_by_name expects of_node_put be called when done. Signed-off-by: Sachin Kamat <sachin.kamat(a)samsung.com> --- sound/soc/codecs/mc13783.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 9965277b595a..388f90a597fa 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -766,11 +766,11 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port); if (ret) - return ret; + goto out; ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port); if (ret) - return ret; + goto out; } dev_set_drvdata(&pdev->dev, priv); @@ -783,6 +783,8 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); +out: + of_node_put(np); return ret; } -- 1.7.9.5

1 0

[alsa-devel] [PATCH 1/2] ARM: dts: imx6: remove wrong spdif rxtx2 clock
by Anssi Hannula 20 Jun '14

20 Jun '14

The rxtx2 clock of i.MX6 S/PDIF is currently set to "asrc" clock. However, according to SoC documentation, rxtx2 is connected to ASRC_EXT_CLK, a different external clock. Testing on Cubox-i system seems to confirm that: when fsl_spdif is forced to select rxtx2 as input clock, audio playback fails with an I/O error. Set rxtx2 to the dummy clock by default to prevent fsl_spdif from selecting it. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> Cc: Mark Brown <broonie(a)kernel.org> --- This one seems clear enough. arch/arm/boot/dts/imx6qdl.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/boot/dts/imx6qdl.dtsi b/arch/arm/boot/dts/imx6qdl.dtsi index ce0599134a69..3aef9a2f5fda 100644 --- a/arch/arm/boot/dts/imx6qdl.dtsi +++ b/arch/arm/boot/dts/imx6qdl.dtsi @@ -181,7 +181,7 @@ <&sdma 15 18 0>; dma-names = "rx", "tx"; clocks = <&clks 197>, <&clks 3>, - <&clks 197>, <&clks 107>, + <&clks 197>, <&clks 0>, <&clks 0>, <&clks 118>, <&clks 0>, <&clks 139>, <&clks 0>; -- 1.8.4.5

3 4

[alsa-devel] [PATCH] *** ASoC: core: refine of node id parse of cpu/codec dai ***
by Qiao Zhou 20 Jun '14

20 Jun '14

Hi Mark, Liam This patch is to add another check besides cpu_dai/codec_dai name during dai_link bind. currently if the cpu_dai/codec_dai name match corresponding dai_link cpu/codec name, then a match is found. in this patch, it also checks whether cpu/codec dai id match dailink cpu_dai_id/codec_dai_id. Still check name first. 1. if it doesn't match, it will keep checking whether cpu_id/codec_id match corresponding dai_link cpu_dai_id/codec_dai_id. if the ids are equal, then a match is found. 2. if it does match, then a match is already found. no need to further check. By this way, it's convenient to maintain dai_link in DT files, and machine driver can parse related node point to get dai id info. Then there is no need to parse cpu_dai/codec_dai name for each dai. Like below: In DT files: sound { compatible = "sound card name xx"; dai_link_1:dai_link_1 { compatible = "xx-dailink-1"; cpu-dai = <&cpu-dai-node CPU_DAI_ID_1>; codec-dai = <&codec-dai-node CODEC_DAI_ID_1>; ...... } dai_link_2:dai_link_2 { compatible = "xx-dailink-2"; cpu-dai = <&cpu-dai-node CPU_DAI_ID_2>; codec-dai = <&codec-dai-node CODEC_DAI_ID_2>; ...... } ...... } In machine driver: static xx_sound_card_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; i = 0; for_each_child_of_node(np, dailink_np) { of_parse_phandle_with_args(dailink_np, "cpu-dai", "#dailink-cells", 0, &out_args); dailink[i].cpu_of_node = out_args.np; dailink[i].cpu_dai_id = out_args.args[0]; dailink[i].platform_of_node = out_args.np; of_parse_phandle_with_args(dailink_np, "codec-dai", "#dailink-cells", 0, &out_args); dailink[i].codec_of_node = out_args.np; dailink[i].codec_dai_id = out_args.args[0]; ...... i++ } ...... ret = snd_soc_register_card(card); ...... } If some drivers don't have DT support, then we have no choice but to add correct cpu/codec(dai) name in DT and parse name info in machine driver. Could you help share your opinions? Thanks in advance. Qiao Zhou (1): ASoC: core: refine of node id parse of cpu/codec dai include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 32 +++++++++++++++++++++++--------- 2 files changed, 25 insertions(+), 9 deletions(-) -- 1.7.9.5

2 5

[alsa-devel] [PATCH] ALSA: hda - Fix usage of "model" module parameter
by David Henningsson 19 Jun '14

19 Jun '14

A recent refactoring broke the possibility to manually specify model name as a module parameter. This patch restores the desired functionality. Reported-by: Kent Baxley <kent.baxley(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index b684c6e..dabe419 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -898,6 +898,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, if (!strcmp(codec->modelname, models->name)) { codec->fixup_id = models->id; codec->fixup_name = models->name; + codec->fixup_list = fixlist; codec->fixup_forced = 1; return; } -- 1.9.1

1 0

[alsa-devel] [PATCH v4 0/3] Add ES8328 audio codec
by Sean Cross 19 Jun '14

19 Jun '14

This patchset adds support for the Everest Semi ES8328 audio codec. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the codec itself and speaker amps, have headphone support, and must supply an unusual frequency to the codec chip. Changes since v3: - Add OF bindings for everest,es8328 - Add back in patch to add everest as a vendor prefix - Give clocks more generic names - Document required clocks in DT bindings document Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 18 + .../devicetree/bindings/sound/imx-audio-es8328.txt | 68 +++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 62 ++ sound/soc/codecs/es8328-spi.c | 51 ++ sound/soc/codecs/es8328.c | 625 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 313 +++++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 359 ++++++++++++ 12 files changed, 1532 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

3 11

[alsa-devel] [PATCH] ASoC: max98090: Remove needless defines and line feeds
by Jarkko Nikula 19 Jun '14

19 Jun '14

Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/codecs/max98090.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index d97f1ce7ff7d..6ea0fdd42eb5 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -26,10 +26,6 @@ #include <sound/max98090.h> #include "max98090.h" -#define DEBUG -#define EXTMIC_METHOD -#define EXTMIC_METHOD_TEST - /* Allows for sparsely populated register maps */ static struct reg_default max98090_reg[] = { { 0x00, 0x00 }, /* 00 Software Reset */ @@ -820,7 +816,6 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w, else val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT; - if (val >= 1) { if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) { max98090->pa1en = val - 1; /* Update for volatile */ @@ -1140,7 +1135,6 @@ static const struct snd_kcontrol_new max98090_mixhprsel_mux = SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_INPUT("DMICL"), @@ -1304,7 +1298,6 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { }; static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("DMIC3"), SND_SOC_DAPM_INPUT("DMIC4"), @@ -1315,7 +1308,6 @@ static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { }; static const struct snd_soc_dapm_route max98090_dapm_routes[] = { - {"MIC1 Input", NULL, "MIC1"}, {"MIC2 Input", NULL, "MIC2"}, @@ -1493,17 +1485,14 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"SPKR", NULL, "SPK Right Out"}, {"RCVL", NULL, "RCV Left Out"}, {"RCVR", NULL, "RCV Right Out"}, - }; static const struct snd_soc_dapm_route max98091_dapm_routes[] = { - /* DMIC inputs */ {"DMIC3", NULL, "DMIC3_ENA"}, {"DMIC4", NULL, "DMIC4_ENA"}, {"DMIC3", NULL, "AHPF"}, {"DMIC4", NULL, "AHPF"}, - }; static int max98090_add_widgets(struct snd_soc_codec *codec) @@ -1531,7 +1520,6 @@ static int max98090_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, max98091_dapm_routes, ARRAY_SIZE(max98091_dapm_routes)); - } return 0; @@ -2221,7 +2209,6 @@ static void max98090_handle_pdata(struct snd_soc_codec *codec) dev_err(codec->dev, "No platform data\n"); return; } - } static int max98090_probe(struct snd_soc_codec *codec) -- 2.0.0

2 1

[alsa-devel] [PATCH] ASoC: max98090: Fix missing free_irq
by Jarkko Nikula 19 Jun '14

19 Jun '14

max98090.c doesn't free the threaded interrupt it requests. This causes an oops when doing "cat /proc/interrupts" after snd-soc-max98090.ko is unloaded. Fix this by requesting the interrupt by using devm_request_threaded_irq(). Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Cc: Stable <stable(a)vger.kernel.org> # 3.10+ --- This issue has been in max98090.c since v3.8 commit 685e42154dcf ("ASoC: Replace max98090 Device Driver"). However this patch applies only after v3.10 which has 3d15aacbb802 ("ASoC: max98090: request IRQF_ONESHOT interrupt"). I'm not sure is there need to go before that since first upstream max98090 user was introduced in a v3.14 commit 7637af2e17f1 ("ASoC: tegra: add tegra+MAX98090 machine driver"). However I'm aware there is an out-of-tree 3.10 user. --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f5fccc7a8e89..d97f1ce7ff7d 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2284,7 +2284,7 @@ static int max98090_probe(struct snd_soc_codec *codec) /* Register for interrupts */ dev_dbg(codec->dev, "irq = %d\n", max98090->irq); - ret = request_threaded_irq(max98090->irq, NULL, + ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL, max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "max98090_interrupt", codec); if (ret < 0) { -- 2.0.0

2 1

[alsa-devel] [PATCH] ASoC: wm5100/wm8903/wm8996: Replace open-coded snd_soc_dapm_to_codec()
by Lars-Peter Clausen 19 Jun '14

19 Jun '14

We now have a generic helper function to cast from a DAPM context to a CODEC. Make use of it in the places which previously open-coded it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm5100.c | 3 +-- sound/soc/codecs/wm8903.c | 3 +-- sound/soc/codecs/wm8996.c | 3 +-- 3 files changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 91a9ea2..7bb0d36 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -735,8 +735,7 @@ WM5100_MIXER_CONTROLS("LHPF4", WM5100_HPLP4MIX_INPUT_1_SOURCE), static void wm5100_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); u16 val, expect, i; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b84940c..ec3250d 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -281,8 +281,7 @@ static int wm8903_dcs_event(struct snd_soc_dapm_widget *w, static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int dcs_mode = WM8903_DCS_MODE_WRITE_STOP; int i, val; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 6926633..9304a91 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -690,8 +690,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); u16 val, mask; -- 1.8.0

3 2

[alsa-devel] [PATCH 1/2] tinycompress: update compress headers for sample rate field
by Vinod Koul 19 Jun '14

19 Jun '14

commit "929559b: ALSA: compress: add num_sample_rates in snd_codec_desc" adds array for passing sample rate instead of bit map. Update the local header as well Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/compress_params.h | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h index deda425..378417a 100644 --- a/include/sound/compress_params.h +++ b/include/sound/compress_params.h @@ -23,6 +23,7 @@ #define MAX_NUM_CODECS 32 #define MAX_NUM_CODEC_DESCRIPTORS 32 #define MAX_NUM_BITRATES 32 +#define MAX_NUM_SAMPLE_RATES 32 #define SND_AUDIOCODEC_PCM ((__u32) 0x00000001) #define SND_AUDIOCODEC_MP3 ((__u32) 0x00000002) @@ -209,7 +210,8 @@ union snd_codec_options { struct snd_codec_desc { __u32 max_ch; - __u32 sample_rates; + __u32 sample_rates[MAX_NUM_SAMPLE_RATES]; + __u32 num_sample_rates; __u32 bit_rate[MAX_NUM_BITRATES]; __u32 num_bitrates; __u32 rate_control; -- 1.7.0.4

1 1

[alsa-devel] Enable HP machine to enable capture mute led and output mute led
by Kailang 19 Jun '14

19 Jun '14

Hi Takashi, Attach was HP mute led support patch. It use Line1 vref and Mic1 vref to control mute led for ALC282. Line1 vref to control output mute led. Mic1 vref to control input mute led. It use GPIO3 and Mic1 vref to control mute led for ALC280. GPIO3 to control output mute led. Mic1 vref to control input mute led. For the DMI string for output mute led application. I already request from HP to support it in next release. HP_Mute_LED_0_A => use Mic1 vref to control mute led. Is the string right? I have tested upper string for Mic1 mute led. BR, Kailang

2 2

Re: [alsa-devel] [PATCH v11 2/8] ASoC: core: Generic ac97 link reset functions
by Markus Pargmann 18 Jun '14

18 Jun '14

Hi Silvio, On Wed, Jun 18, 2014 at 08:42:34PM +0200, S. Fricke wrote: > > Hi Markus, > > I'm fighting with your ac97 pinctrl configuration per dts. > > > [...] > > +ssi { > > + ... > > + > > + pinctrl-names = "default", "ac97-running", "ac97-reset", "ac97-warm-reset"; > > + pinctrl-0 = <&ac97link_running>; > > + pinctrl-1 = <&ac97link_running>; > > + pinctrl-2 = <&ac97link_reset>; > > + pinctrl-3 = <&ac97link_warm_reset>; > > + ac97-gpios = <&gpio3 20 0 &gpio3 22 0 &gpio3 28 0>; > > I see here the gpios for sync, sdata and reset. On my board I have two sdata > lines (sdata-in and sdata-out) which line should I specify here for this setting? Try to use sdata-out. This is the only data output line. sdata-in is the input to the SoC so we don't have to use this for an AC97 reset. Regards, Markus -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

1 0

Re: [alsa-devel] [RFC PATCH] ASoC: tas2552: Support TI TAS2552 Amplifier
by Mark Brown 18 Jun '14

18 Jun '14

On Wed, Jun 18, 2014 at 12:45:46PM -0500, Dan Murphy wrote: > +static int tas2552_i2c_read(struct tas2552_data *tas_data, int reg) > +{ > + int val; > + > + if (WARN_ON(!tas_data->tas2552_client)) > + return -EINVAL; > + > + /* If powered off, return the cached value */ > + mutex_lock(&tas_data->mutex); > + if (tas_data->power_state) { > + val = i2c_smbus_read_byte_data(tas_data->tas2552_client, reg); Don't open code your I/O, use regmap like any other CODEC driver. This will remove a very large proportion of the code in the driver which replicates features of the regmap framework. > +static int tas2552_startup(struct snd_pcm_substream *substream, > + struct snd_soc_dai *dai) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); > + > + tas2552_sw_shutdown(tas2552, 1); > + tas2552_power(tas2552, 1); > + /* Turn on Class D amplifier */ > + tas2552_i2c_write(tas2552, TAS2552_CFG_2, 0x80); > + tas2552_sw_shutdown(tas2552, 0); I'd expect this to be done using DAPM. > +static const struct regmap_config tas2552_regmap_config = { > + .reg_bits = 8, > + .val_bits = 8, > + > + .max_register = TAS2552_MAX_REG, > + .reg_defaults = NULL, > + .num_reg_defaults = 0, No need to set things to zero or NULL in static variables, that's the default. > + dev = &client->dev; > + data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); > + if (data == NULL) { > + dev_err(dev, "Can not allocate memory\n"); No need to print an error, OOM messages are already quite noisy.

1 0

[alsa-devel] [PATCH v3] ASoC: davinci-mcasp: set up channel status bits for S/PDIF mode
by Daniel Mack 18 Jun '14

18 Jun '14

In DIT (S/PDIF) mode, program the transmitted user bits to reflect the configured sample rate, along with some other details. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- v3 just changes variable names. sound/soc/davinci/davinci-mcasp.c | 49 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 47 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 4500cf0..a21a7d4 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include <linux/of_platform.h> #include <linux/of_device.h> +#include <sound/asoundef.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -576,8 +577,12 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) } /* S/PDIF */ -static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) +static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, + unsigned int rate) { + u32 cs_value = 0; + u8 *cs_bytes = (u8*) &cs_value; + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); @@ -599,6 +604,46 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) /* Enable the DIT */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); + /* Set S/PDIF channel status bits */ + cs_bytes[0] = IEC958_AES0_CON_NOT_COPYRIGHT; + cs_bytes[1] = IEC958_AES1_CON_PCM_CODER; + + switch (rate) { + case 22050: + cs_bytes[3] |= IEC958_AES3_CON_FS_22050; + break; + case 24000: + cs_bytes[3] |= IEC958_AES3_CON_FS_24000; + break; + case 32000: + cs_bytes[3] |= IEC958_AES3_CON_FS_32000; + break; + case 44100: + cs_bytes[3] |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + cs_bytes[3] |= IEC958_AES3_CON_FS_48000; + break; + case 88200: + cs_bytes[3] |= IEC958_AES3_CON_FS_88200; + break; + case 96000: + cs_bytes[3] |= IEC958_AES3_CON_FS_96000; + break; + case 176400: + cs_bytes[3] |= IEC958_AES3_CON_FS_176400; + break; + case 192000: + cs_bytes[3] |= IEC958_AES3_CON_FS_192000; + break; + default: + printk(KERN_WARNING "unsupported sampling rate: %d\n", rate); + return -EINVAL; + } + + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, cs_value); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, cs_value); + return 0; } @@ -634,7 +679,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return ret; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - ret = mcasp_dit_hw_param(mcasp); + ret = mcasp_dit_hw_param(mcasp, params_rate(params)); else ret = mcasp_i2s_hw_param(mcasp, substream->stream); -- 1.8.5.3

4 4

[alsa-devel] [PATCH 1/2] ASoC: ak5386: Add regulators to documentation and fix sparse warning
by Daniel Mack 18 Jun '14

18 Jun '14

Document the newly added regulators to the DT binding document. Also, "static const char const *x" is not identical to "static const char * const x", which sparse now complains about. Fix it. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Documentation/devicetree/bindings/sound/ak5386.txt | 4 ++++ sound/soc/codecs/ak5386.c | 2 +- 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/ak5386.txt b/Documentation/devicetree/bindings/sound/ak5386.txt index dc3914f..ec3df3a 100644 --- a/Documentation/devicetree/bindings/sound/ak5386.txt +++ b/Documentation/devicetree/bindings/sound/ak5386.txt @@ -10,10 +10,14 @@ Optional properties: - reset-gpio : a GPIO spec for the reset/power down pin. If specified, it will be deasserted at probe time. + - va-supply : a regulator spec, providing 5.0V + - vd-supply : a regulator spec, providing 3.3V Example: spdif: ak5386@0 { compatible = "asahi-kasei,ak5386"; reset-gpio = <&gpio0 23>; + va-supply = <&vdd_5v0_reg>; + vd-supply = <&vdd_3v3_reg>; }; diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index a30be5c..8107a1c 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -19,7 +19,7 @@ #include <sound/pcm.h> #include <sound/initval.h> -static const char const *supply_names[] = { +static const char * const supply_names[] = { "va", "vd" }; -- 1.8.5.3

3 5

Re: [alsa-devel] [v3.13.y][SRU][PATCH 0/1] ALSA: usb-audio: Prevent printk ratelimiting from spamming kernel log while DEBUG not defined
by Greg KH 18 Jun '14

18 Jun '14

On Wed, Jun 18, 2014 at 01:14:05PM -0400, Joseph Salisbury wrote: > BugLink: http://bugs.launchpad.net/bugs/1319457 > > Hello, > > Please consider including upstream commit b7a77235 in the next v3.13.y release. It was included upstream as of v3.15-rc5. It has been tested and confirmed to resolve http://bugs.launchpad.net/bugs/1319457 . Please get a good email client and wrap your lines :( > commit b7a7723513dc89f83d6df13206df55d4dc26e825 > Author: Sander Eikelenboom <linux(a)eikelenboom.it> > Date: Fri May 2 15:09:27 2014 +0200 > > ALSA: usb-audio: Prevent printk ratelimiting from spamming kernel log while DEBUG not defined > > > This commit does not apply cleanly to v3.13, so I performed a backport, which is in email 1/1. Any reason this shouldn't also be in 3.14-stable? thanks, greg k-h

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - Remove the obsoleted static quirk codes from patch_conexant.c
by Takashi Iwai 18 Jun '14

18 Jun '14

The static quirk code has been disabled for a while and it seems working fine, so it's time to actually get rid of it. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 2631 +--------------------------------------- 1 file changed, 9 insertions(+), 2622 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1dc7e974f3b1..7627a69ca6d7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -34,27 +34,6 @@ #include "hda_jack.h" #include "hda_generic.h" -#undef ENABLE_CXT_STATIC_QUIRKS - -#define CXT_PIN_DIR_IN 0x00 -#define CXT_PIN_DIR_OUT 0x01 -#define CXT_PIN_DIR_INOUT 0x02 -#define CXT_PIN_DIR_IN_NOMICBIAS 0x03 -#define CXT_PIN_DIR_INOUT_NOMICBIAS 0x04 - -#define CONEXANT_HP_EVENT 0x37 -#define CONEXANT_MIC_EVENT 0x38 -#define CONEXANT_LINE_EVENT 0x39 - -/* Conexant 5051 specific */ - -#define CXT5051_SPDIF_OUT 0x12 -#define CXT5051_PORTB_EVENT 0x38 -#define CXT5051_PORTC_EVENT 0x39 - -#define AUTO_MIC_PORTB (1 << 1) -#define AUTO_MIC_PORTC (1 << 2) - struct conexant_spec { struct hda_gen_spec gen; @@ -72,64 +51,6 @@ struct conexant_spec { bool dc_enable; unsigned int dc_input_bias; /* offset into olpc_xo_dc_bias */ struct nid_path *dc_mode_path; - -#ifdef ENABLE_CXT_STATIC_QUIRKS - const struct snd_kcontrol_new *mixers[5]; - int num_mixers; - hda_nid_t vmaster_nid; - - const struct hda_verb *init_verbs[5]; /* initialization verbs - * don't forget NULL - * termination! - */ - unsigned int num_init_verbs; - - /* playback */ - struct hda_multi_out multiout; /* playback set-up - * max_channels, dacs must be set - * dig_out_nid and hp_nid are optional - */ - unsigned int cur_eapd; - unsigned int hp_present; - unsigned int line_present; - unsigned int auto_mic; - - /* capture */ - unsigned int num_adc_nids; - const hda_nid_t *adc_nids; - hda_nid_t dig_in_nid; /* digital-in NID; optional */ - - unsigned int cur_adc_idx; - hda_nid_t cur_adc; - unsigned int cur_adc_stream_tag; - unsigned int cur_adc_format; - - const struct hda_pcm_stream *capture_stream; - - /* capture source */ - const struct hda_input_mux *input_mux; - const hda_nid_t *capsrc_nids; - unsigned int cur_mux[3]; - - /* channel model */ - const struct hda_channel_mode *channel_mode; - int num_channel_mode; - - /* PCM information */ - struct hda_pcm pcm_rec[2]; /* used in build_pcms() */ - - unsigned int spdif_route; - - unsigned int port_d_mode; - unsigned int dell_automute:1; - unsigned int dell_vostro:1; - unsigned int ideapad:1; - unsigned int thinkpad:1; - unsigned int hp_laptop:1; - unsigned int asus:1; - - unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */ -#endif /* ENABLE_CXT_STATIC_QUIRKS */ }; @@ -173,2533 +94,6 @@ static int add_beep_ctls(struct hda_codec *codec) #define add_beep_ctls(codec) 0 #endif - -#ifdef ENABLE_CXT_STATIC_QUIRKS -static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, - hinfo); -} - -static int conexant_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, - stream_tag, - format, substream); -} - -static int conexant_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - -/* - * Digital out - */ -static int conexant_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_dig_open(codec, &spec->multiout); -} - -static int conexant_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_dig_close(codec, &spec->multiout); -} - -static int conexant_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, - stream_tag, - format, substream); -} - -/* - * Analog capture - */ -static int conexant_capture_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - stream_tag, 0, format); - return 0; -} - -static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); - return 0; -} - - - -static const struct hda_pcm_stream conexant_pcm_analog_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .open = conexant_playback_pcm_open, - .prepare = conexant_playback_pcm_prepare, - .cleanup = conexant_playback_pcm_cleanup - }, -}; - -static const struct hda_pcm_stream conexant_pcm_analog_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .prepare = conexant_capture_pcm_prepare, - .cleanup = conexant_capture_pcm_cleanup - }, -}; - - -static const struct hda_pcm_stream conexant_pcm_digital_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .open = conexant_dig_playback_pcm_open, - .close = conexant_dig_playback_pcm_close, - .prepare = conexant_dig_playback_pcm_prepare - }, -}; - -static const struct hda_pcm_stream conexant_pcm_digital_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - /* NID is set in alc_build_pcms */ -}; - -static int cx5051_capture_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - spec->cur_adc = spec->adc_nids[spec->cur_adc_idx]; - spec->cur_adc_stream_tag = stream_tag; - spec->cur_adc_format = format; - snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format); - return 0; -} - -static int cx5051_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - snd_hda_codec_cleanup_stream(codec, spec->cur_adc); - spec->cur_adc = 0; - return 0; -} - -static const struct hda_pcm_stream cx5051_pcm_analog_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .prepare = cx5051_capture_pcm_prepare, - .cleanup = cx5051_capture_pcm_cleanup - }, -}; - -static int conexant_build_pcms(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; - - codec->num_pcms = 1; - codec->pcm_info = info; - - info->name = "CONEXANT Analog"; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = conexant_pcm_analog_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = - spec->multiout.max_channels; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = - spec->multiout.dac_nids[0]; - if (spec->capture_stream) - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *spec->capture_stream; - else { - if (codec->vendor_id == 0x14f15051) - info->stream[SNDRV_PCM_STREAM_CAPTURE] = - cx5051_pcm_analog_capture; - else { - info->stream[SNDRV_PCM_STREAM_CAPTURE] = - conexant_pcm_analog_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = - spec->num_adc_nids; - } - } - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; - - if (spec->multiout.dig_out_nid) { - info++; - codec->num_pcms++; - info->name = "Conexant Digital"; - info->pcm_type = HDA_PCM_TYPE_SPDIF; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = - conexant_pcm_digital_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = - spec->multiout.dig_out_nid; - if (spec->dig_in_nid) { - info->stream[SNDRV_PCM_STREAM_CAPTURE] = - conexant_pcm_digital_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = - spec->dig_in_nid; - } - } - - return 0; -} - -static int conexant_mux_enum_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - - return snd_hda_input_mux_info(spec->input_mux, uinfo); -} - -static int conexant_mux_enum_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - - ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx]; - return 0; -} - -static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - spec->capsrc_nids[adc_idx], - &spec->cur_mux[adc_idx]); -} - -static void conexant_set_power(struct hda_codec *codec, hda_nid_t fg, - unsigned int power_state) -{ - if (power_state == AC_PWRST_D3) - msleep(100); - snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, - power_state); - /* partial workaround for "azx_get_response timeout" */ - if (power_state == AC_PWRST_D0) - msleep(10); - snd_hda_codec_set_power_to_all(codec, fg, power_state); -} - -static int conexant_init(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_init_verbs; i++) - snd_hda_sequence_write(codec, spec->init_verbs[i]); - return 0; -} - -static void conexant_free(struct hda_codec *codec) -{ - kfree(codec->spec); -} - -static const struct snd_kcontrol_new cxt_capture_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, - {} -}; - -static const char * const slave_pfxs[] = { - "Headphone", "Speaker", "Bass Speaker", "Front", "Surround", "CLFE", - NULL -}; - -static int conexant_build_controls(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int i; - int err; - - for (i = 0; i < spec->num_mixers; i++) { - err = snd_hda_add_new_ctls(codec, spec->mixers[i]); - if (err < 0) - return err; - } - if (spec->multiout.dig_out_nid) { - err = snd_hda_create_spdif_out_ctls(codec, - spec->multiout.dig_out_nid, - spec->multiout.dig_out_nid); - if (err < 0) - return err; - err = snd_hda_create_spdif_share_sw(codec, - &spec->multiout); - if (err < 0) - return err; - spec->multiout.share_spdif = 1; - } - if (spec->dig_in_nid) { - err = snd_hda_create_spdif_in_ctls(codec,spec->dig_in_nid); - if (err < 0) - return err; - } - - /* if we have no master control, let's create it */ - if (spec->vmaster_nid && - !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { - unsigned int vmaster_tlv[4]; - snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, - HDA_OUTPUT, vmaster_tlv); - err = snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, slave_pfxs, - "Playback Volume"); - if (err < 0) - return err; - } - if (spec->vmaster_nid && - !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { - err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, slave_pfxs, - "Playback Switch"); - if (err < 0) - return err; - } - - if (spec->input_mux) { - err = snd_hda_add_new_ctls(codec, cxt_capture_mixers); - if (err < 0) - return err; - } - - err = add_beep_ctls(codec); - if (err < 0) - return err; - - return 0; -} - -static const struct hda_codec_ops conexant_patch_ops = { - .build_controls = conexant_build_controls, - .build_pcms = conexant_build_pcms, - .init = conexant_init, - .free = conexant_free, - .set_power_state = conexant_set_power, -}; - -static int patch_conexant_auto(struct hda_codec *codec); -/* - * EAPD control - * the private value = nid | (invert << 8) - */ - -#define cxt_eapd_info snd_ctl_boolean_mono_info - -static int cxt_eapd_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - int invert = (kcontrol->private_value >> 8) & 1; - if (invert) - ucontrol->value.integer.value[0] = !spec->cur_eapd; - else - ucontrol->value.integer.value[0] = spec->cur_eapd; - return 0; - -} - -static int cxt_eapd_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - int invert = (kcontrol->private_value >> 8) & 1; - hda_nid_t nid = kcontrol->private_value & 0xff; - unsigned int eapd; - - eapd = !!ucontrol->value.integer.value[0]; - if (invert) - eapd = !eapd; - if (eapd == spec->cur_eapd) - return 0; - - spec->cur_eapd = eapd; - snd_hda_codec_write_cache(codec, nid, - 0, AC_VERB_SET_EAPD_BTLENABLE, - eapd ? 0x02 : 0x00); - return 1; -} - -/* controls for test mode */ -#ifdef CONFIG_SND_DEBUG - -#define CXT_EAPD_SWITCH(xname, nid, mask) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .info = cxt_eapd_info, \ - .get = cxt_eapd_get, \ - .put = cxt_eapd_put, \ - .private_value = nid | (mask<<16) } - - - -static int conexant_ch_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode, - spec->num_channel_mode); -} - -static int conexant_ch_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - spec->multiout.max_channels); -} - -static int conexant_ch_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - &spec->multiout.max_channels); - return err; -} - -#define CXT_PIN_MODE(xname, nid, dir) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .info = conexant_ch_mode_info, \ - .get = conexant_ch_mode_get, \ - .put = conexant_ch_mode_put, \ - .private_value = nid | (dir<<16) } - -#endif /* CONFIG_SND_DEBUG */ - -/* Conexant 5045 specific */ - -static const hda_nid_t cxt5045_dac_nids[1] = { 0x19 }; -static const hda_nid_t cxt5045_adc_nids[1] = { 0x1a }; -static const hda_nid_t cxt5045_capsrc_nids[1] = { 0x1a }; -#define CXT5045_SPDIF_OUT 0x18 - -static const struct hda_channel_mode cxt5045_modes[1] = { - { 2, NULL }, -}; - -static const struct hda_input_mux cxt5045_capture_source = { - .num_items = 2, - .items = { - { "Internal Mic", 0x1 }, - { "Mic", 0x2 }, - } -}; - -static const struct hda_input_mux cxt5045_capture_source_benq = { - .num_items = 4, - .items = { - { "Internal Mic", 0x1 }, - { "Mic", 0x2 }, - { "Line", 0x3 }, - { "Mixer", 0x0 }, - } -}; - -/* turn on/off EAPD (+ mute HP) as a master switch */ -static int cxt5045_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - if (!cxt_eapd_put(kcontrol, ucontrol)) - return 0; - - /* toggle internal speakers mute depending of presence of - * the headphone jack - */ - bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - - bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, 0x11, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - return 1; -} - -/* bind volumes of both NID 0x10 and 0x11 */ -static const struct hda_bind_ctls cxt5045_hp_bind_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x10, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -/* toggle input of built-in and mic jack appropriately */ -static void cxt5045_hp_automic(struct hda_codec *codec) -{ - static const struct hda_verb mic_jack_on[] = { - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {} - }; - static const struct hda_verb mic_jack_off[] = { - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {} - }; - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x12); - if (present) - snd_hda_sequence_write(codec, mic_jack_on); - else - snd_hda_sequence_write(codec, mic_jack_off); -} - - -/* mute internal speaker if HP is plugged */ -static void cxt5045_hp_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - spec->hp_present = snd_hda_jack_detect(codec, 0x11); - - bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} - -/* unsolicited event for HP jack sensing */ -static void cxt5045_hp_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - res >>= 26; - switch (res) { - case CONEXANT_HP_EVENT: - cxt5045_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: - cxt5045_hp_automic(codec); - break; - - } -} - -static const struct snd_kcontrol_new cxt5045_mixers[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x17, 0x2, HDA_INPUT), - HDA_BIND_VOL("Master Playback Volume", &cxt5045_hp_bind_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5045_hp_master_sw_put, - .private_value = 0x10, - }, - - {} -}; - -static const struct snd_kcontrol_new cxt5045_benq_mixers[] = { - HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT), - - {} -}; - -static const struct hda_verb cxt5045_init_verbs[] = { - /* Line in, Mic */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, - /* HP, Amp */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x10, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x11, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Record selector: Internal mic */ - {0x1a, AC_VERB_SET_CONNECT_SEL,0x1}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, - /* SPDIF route: PCM */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - { 0x13, AC_VERB_SET_CONNECT_SEL, 0x0 }, - /* EAPD */ - {0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2 }, /* default on */ - { } /* end */ -}; - -static const struct hda_verb cxt5045_benq_init_verbs[] = { - /* Internal Mic, Mic */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, - /* Line In,HP, Amp */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x10, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x11, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Record selector: Internal mic */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, - /* SPDIF route: PCM */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* EAPD */ - {0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - { } /* end */ -}; - -static const struct hda_verb cxt5045_hp_sense_init_verbs[] = { - /* pin sensing on HP jack */ - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5045_mic_sense_init_verbs[] = { - /* pin sensing on HP jack */ - {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -#ifdef CONFIG_SND_DEBUG -/* Test configuration for debugging, modelled after the ALC260 test - * configuration. - */ -static const struct hda_input_mux cxt5045_test_capture_source = { - .num_items = 5, - .items = { - { "MIXER", 0x0 }, - { "MIC1 pin", 0x1 }, - { "LINE1 pin", 0x2 }, - { "HP-OUT pin", 0x3 }, - { "CD pin", 0x4 }, - }, -}; - -static const struct snd_kcontrol_new cxt5045_test_mixer[] = { - - /* Output controls */ - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("HP-OUT Playback Volume", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("HP-OUT Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("LINE1 Playback Switch", 0x12, 0x0, HDA_OUTPUT), - - /* Modes for retasking pin widgets */ - CXT_PIN_MODE("HP-OUT pin mode", 0x11, CXT_PIN_DIR_INOUT), - CXT_PIN_MODE("LINE1 pin mode", 0x12, CXT_PIN_DIR_INOUT), - - /* EAPD Switch Control */ - CXT_EAPD_SWITCH("External Amplifier", 0x10, 0x0), - - /* Loopback mixer controls */ - - HDA_CODEC_VOLUME("PCM Volume", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("PCM Switch", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("MIC1 pin Volume", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("MIC1 pin Switch", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("LINE1 pin Volume", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("LINE1 pin Switch", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("HP-OUT pin Volume", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("HP-OUT pin Switch", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("CD pin Volume", 0x17, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD pin Switch", 0x17, 0x4, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Input Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put, - }, - /* Audio input controls */ - HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_verb cxt5045_test_init_verbs[] = { - /* Set connections */ - { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 }, - { 0x11, AC_VERB_SET_CONNECT_SEL, 0x0 }, - { 0x12, AC_VERB_SET_CONNECT_SEL, 0x0 }, - /* Enable retasking pins as output, initially without power amp */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* Disable digital (SPDIF) pins initially, but users can enable - * them via a mixer switch. In the case of SPDIF-out, this initverb - * payload also sets the generation to 0, output to be in "consumer" - * PCM format, copyright asserted, no pre-emphasis and no validity - * control. - */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Unmute retasking pin widget output buffers since the default - * state appears to be output. As the pin mode is changed by the - * user the pin mode control will take care of enabling the pin's - * input/output buffers as needed. - */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Mute capture amp left and right */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - - /* Set ADC connection select to match default mixer setting (mic1 - * pin) - */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x17, AC_VERB_SET_CONNECT_SEL, 0x01}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Mic1 pin */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* Line pin */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* HP pin */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - - { } -}; -#endif - - -/* initialize jack-sensing, too */ -static int cxt5045_init(struct hda_codec *codec) -{ - conexant_init(codec); - cxt5045_hp_automute(codec); - return 0; -} - - -enum { - CXT5045_LAPTOP_HPSENSE, - CXT5045_LAPTOP_MICSENSE, - CXT5045_LAPTOP_HPMICSENSE, - CXT5045_BENQ, -#ifdef CONFIG_SND_DEBUG - CXT5045_TEST, -#endif - CXT5045_AUTO, - CXT5045_MODELS -}; - -static const char * const cxt5045_models[CXT5045_MODELS] = { - [CXT5045_LAPTOP_HPSENSE] = "laptop-hpsense", - [CXT5045_LAPTOP_MICSENSE] = "laptop-micsense", - [CXT5045_LAPTOP_HPMICSENSE] = "laptop-hpmicsense", - [CXT5045_BENQ] = "benq", -#ifdef CONFIG_SND_DEBUG - [CXT5045_TEST] = "test", -#endif - [CXT5045_AUTO] = "auto", -}; - -static const struct snd_pci_quirk cxt5045_cfg_tbl[] = { - SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), - SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), - SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505", - CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK_MASK(0x1631, 0xff00, 0xc100, "Packard Bell", - CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP_HPSENSE), - {} -}; - -static int patch_cxt5045(struct hda_codec *codec) -{ - struct conexant_spec *spec; - int board_config; - - board_config = snd_hda_check_board_config(codec, CXT5045_MODELS, - cxt5045_models, - cxt5045_cfg_tbl); - if (board_config < 0) - board_config = CXT5045_AUTO; /* model=auto as default */ - if (board_config == CXT5045_AUTO) - return patch_conexant_auto(codec); - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; - codec->spec = spec; - codec->single_adc_amp = 1; - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids); - spec->multiout.dac_nids = cxt5045_dac_nids; - spec->multiout.dig_out_nid = CXT5045_SPDIF_OUT; - spec->num_adc_nids = 1; - spec->adc_nids = cxt5045_adc_nids; - spec->capsrc_nids = cxt5045_capsrc_nids; - spec->input_mux = &cxt5045_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = cxt5045_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = cxt5045_init_verbs; - spec->spdif_route = 0; - spec->num_channel_mode = ARRAY_SIZE(cxt5045_modes); - spec->channel_mode = cxt5045_modes; - - set_beep_amp(spec, 0x16, 0, 1); - - codec->patch_ops = conexant_patch_ops; - - switch (board_config) { - case CXT5045_LAPTOP_HPSENSE: - codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; - spec->input_mux = &cxt5045_capture_source; - spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5045_hp_sense_init_verbs; - spec->mixers[0] = cxt5045_mixers; - codec->patch_ops.init = cxt5045_init; - break; - case CXT5045_LAPTOP_MICSENSE: - codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; - spec->input_mux = &cxt5045_capture_source; - spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5045_mic_sense_init_verbs; - spec->mixers[0] = cxt5045_mixers; - codec->patch_ops.init = cxt5045_init; - break; - default: - case CXT5045_LAPTOP_HPMICSENSE: - codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; - spec->input_mux = &cxt5045_capture_source; - spec->num_init_verbs = 3; - spec->init_verbs[1] = cxt5045_hp_sense_init_verbs; - spec->init_verbs[2] = cxt5045_mic_sense_init_verbs; - spec->mixers[0] = cxt5045_mixers; - codec->patch_ops.init = cxt5045_init; - break; - case CXT5045_BENQ: - codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; - spec->input_mux = &cxt5045_capture_source_benq; - spec->num_init_verbs = 1; - spec->init_verbs[0] = cxt5045_benq_init_verbs; - spec->mixers[0] = cxt5045_mixers; - spec->mixers[1] = cxt5045_benq_mixers; - spec->num_mixers = 2; - codec->patch_ops.init = cxt5045_init; - break; -#ifdef CONFIG_SND_DEBUG - case CXT5045_TEST: - spec->input_mux = &cxt5045_test_capture_source; - spec->mixers[0] = cxt5045_test_mixer; - spec->init_verbs[0] = cxt5045_test_init_verbs; - break; - -#endif - } - - switch (codec->subsystem_id >> 16) { - case 0x103c: - case 0x1631: - case 0x1734: - case 0x17aa: - /* HP, Packard Bell, Fujitsu-Siemens & Lenovo laptops have - * really bad sound over 0dB on NID 0x17. Fix max PCM level to - * 0 dB (originally it has 0x2b steps with 0dB offset 0x14) - */ - snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT, - (0x14 << AC_AMPCAP_OFFSET_SHIFT) | - (0x14 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - } - - if (spec->beep_amp) - snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); - - return 0; -} - - -/* Conexant 5047 specific */ -#define CXT5047_SPDIF_OUT 0x11 - -static const hda_nid_t cxt5047_dac_nids[1] = { 0x10 }; /* 0x1c */ -static const hda_nid_t cxt5047_adc_nids[1] = { 0x12 }; -static const hda_nid_t cxt5047_capsrc_nids[1] = { 0x1a }; - -static const struct hda_channel_mode cxt5047_modes[1] = { - { 2, NULL }, -}; - -static const struct hda_input_mux cxt5047_toshiba_capture_source = { - .num_items = 2, - .items = { - { "ExtMic", 0x2 }, - { "Line-In", 0x1 }, - } -}; - -/* turn on/off EAPD (+ mute HP) as a master switch */ -static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - if (!cxt_eapd_put(kcontrol, ucontrol)) - return 0; - - /* toggle internal speakers mute depending of presence of - * the headphone jack - */ - bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE; - /* NOTE: Conexat codec needs the index for *OUTPUT* amp of - * pin widgets unlike other codecs. In this case, we need to - * set index 0x01 for the volume from the mixer amp 0x19. - */ - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, - HDA_AMP_MUTE, bits); - bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - return 1; -} - -/* mute internal speaker if HP is plugged */ -static void cxt5047_hp_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - spec->hp_present = snd_hda_jack_detect(codec, 0x13); - - bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; - /* See the note in cxt5047_hp_master_sw_put */ - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, - HDA_AMP_MUTE, bits); -} - -/* toggle input of built-in and mic jack appropriately */ -static void cxt5047_hp_automic(struct hda_codec *codec) -{ - static const struct hda_verb mic_jack_on[] = { - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {} - }; - static const struct hda_verb mic_jack_off[] = { - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {} - }; - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x15); - if (present) - snd_hda_sequence_write(codec, mic_jack_on); - else - snd_hda_sequence_write(codec, mic_jack_off); -} - -/* unsolicited event for HP jack sensing */ -static void cxt5047_hp_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5047_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: - cxt5047_hp_automic(codec); - break; - } -} - -static const struct snd_kcontrol_new cxt5047_base_mixers[] = { - HDA_CODEC_VOLUME("Mic Playback Volume", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5047_hp_master_sw_put, - .private_value = 0x13, - }, - - {} -}; - -static const struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = { - /* See the note in cxt5047_hp_master_sw_put */ - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x01, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5047_hp_only_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x13, 0x00, HDA_OUTPUT), - { } /* end */ -}; - -static const struct hda_verb cxt5047_init_verbs[] = { - /* Line in, Mic, Built-in Mic */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 }, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 }, - /* HP, Speaker */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, /* mixer(0x19) */ - {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mixer(0x19) */ - /* Record selector: Mic */ - {0x12, AC_VERB_SET_CONNECT_SEL,0x03}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, - {0x1A, AC_VERB_SET_CONNECT_SEL,0x02}, - {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_OUTPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x00}, - {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_OUTPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x03}, - /* SPDIF route: PCM */ - { 0x18, AC_VERB_SET_CONNECT_SEL, 0x0 }, - /* Enable unsolicited events */ - {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -/* configuration for Toshiba Laptops */ -static const struct hda_verb cxt5047_toshiba_init_verbs[] = { - {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0}, /* default off */ - {} -}; - -/* Test configuration for debugging, modelled after the ALC260 test - * configuration. - */ -#ifdef CONFIG_SND_DEBUG -static const struct hda_input_mux cxt5047_test_capture_source = { - .num_items = 4, - .items = { - { "LINE1 pin", 0x0 }, - { "MIC1 pin", 0x1 }, - { "MIC2 pin", 0x2 }, - { "CD pin", 0x3 }, - }, -}; - -static const struct snd_kcontrol_new cxt5047_test_mixer[] = { - - /* Output only controls */ - HDA_CODEC_VOLUME("OutAmp-1 Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("OutAmp-1 Switch", 0x10,0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("OutAmp-2 Volume", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("OutAmp-2 Switch", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("HeadPhone Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("HeadPhone Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line1-Out Playback Volume", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line1-Out Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line2-Out Playback Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line2-Out Playback Switch", 0x15, 0x0, HDA_OUTPUT), - - /* Modes for retasking pin widgets */ - CXT_PIN_MODE("LINE1 pin mode", 0x14, CXT_PIN_DIR_INOUT), - CXT_PIN_MODE("MIC1 pin mode", 0x15, CXT_PIN_DIR_INOUT), - - /* EAPD Switch Control */ - CXT_EAPD_SWITCH("External Amplifier", 0x13, 0x0), - - /* Loopback mixer controls */ - HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x12, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("MIC1 Playback Switch", 0x12, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x12, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("MIC2 Playback Switch", 0x12, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("LINE Playback Volume", 0x12, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("LINE Playback Switch", 0x12, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x12, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x12, 0x04, HDA_INPUT), - - HDA_CODEC_VOLUME("Capture-1 Volume", 0x19, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture-1 Switch", 0x19, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture-2 Volume", 0x19, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Capture-2 Switch", 0x19, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture-3 Volume", 0x19, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Capture-3 Switch", 0x19, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Capture-4 Volume", 0x19, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Capture-4 Switch", 0x19, 0x3, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Input Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put, - }, - HDA_CODEC_VOLUME("Mic Boost Volume", 0x1a, 0x0, HDA_OUTPUT), - - { } /* end */ -}; - -static const struct hda_verb cxt5047_test_init_verbs[] = { - /* Enable retasking pins as output, initially without power amp */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* Disable digital (SPDIF) pins initially, but users can enable - * them via a mixer switch. In the case of SPDIF-out, this initverb - * payload also sets the generation to 0, output to be in "consumer" - * PCM format, copyright asserted, no pre-emphasis and no validity - * control. - */ - {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Ensure mic1, mic2, line1 pin widgets take input from the - * OUT1 sum bus when acting as an output. - */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0}, - - /* Start with output sum widgets muted and their output gains at min */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - - /* Unmute retasking pin widget output buffers since the default - * state appears to be output. As the pin mode is changed by the - * user the pin mode control will take care of enabling the pin's - * input/output buffers as needed. - */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Mute capture amp left and right */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - - /* Set ADC connection select to match default mixer setting (mic1 - * pin) - */ - {0x12, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ - - { } -}; -#endif - - -/* initialize jack-sensing, too */ -static int cxt5047_hp_init(struct hda_codec *codec) -{ - conexant_init(codec); - cxt5047_hp_automute(codec); - return 0; -} - - -enum { - CXT5047_LAPTOP, /* Laptops w/o EAPD support */ - CXT5047_LAPTOP_HP, /* Some HP laptops */ - CXT5047_LAPTOP_EAPD, /* Laptops with EAPD support */ -#ifdef CONFIG_SND_DEBUG - CXT5047_TEST, -#endif - CXT5047_AUTO, - CXT5047_MODELS -}; - -static const char * const cxt5047_models[CXT5047_MODELS] = { - [CXT5047_LAPTOP] = "laptop", - [CXT5047_LAPTOP_HP] = "laptop-hp", - [CXT5047_LAPTOP_EAPD] = "laptop-eapd", -#ifdef CONFIG_SND_DEBUG - [CXT5047_TEST] = "test", -#endif - [CXT5047_AUTO] = "auto", -}; - -static const struct snd_pci_quirk cxt5047_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP), - SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series", - CXT5047_LAPTOP), - SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD), - {} -}; - -static int patch_cxt5047(struct hda_codec *codec) -{ - struct conexant_spec *spec; - int board_config; - - board_config = snd_hda_check_board_config(codec, CXT5047_MODELS, - cxt5047_models, - cxt5047_cfg_tbl); - if (board_config < 0) - board_config = CXT5047_AUTO; /* model=auto as default */ - if (board_config == CXT5047_AUTO) - return patch_conexant_auto(codec); - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; - codec->spec = spec; - codec->pin_amp_workaround = 1; - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(cxt5047_dac_nids); - spec->multiout.dac_nids = cxt5047_dac_nids; - spec->multiout.dig_out_nid = CXT5047_SPDIF_OUT; - spec->num_adc_nids = 1; - spec->adc_nids = cxt5047_adc_nids; - spec->capsrc_nids = cxt5047_capsrc_nids; - spec->num_mixers = 1; - spec->mixers[0] = cxt5047_base_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = cxt5047_init_verbs; - spec->spdif_route = 0; - spec->num_channel_mode = ARRAY_SIZE(cxt5047_modes), - spec->channel_mode = cxt5047_modes, - - codec->patch_ops = conexant_patch_ops; - - switch (board_config) { - case CXT5047_LAPTOP: - spec->num_mixers = 2; - spec->mixers[1] = cxt5047_hp_spk_mixers; - codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; - break; - case CXT5047_LAPTOP_HP: - spec->num_mixers = 2; - spec->mixers[1] = cxt5047_hp_only_mixers; - codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; - codec->patch_ops.init = cxt5047_hp_init; - break; - case CXT5047_LAPTOP_EAPD: - spec->input_mux = &cxt5047_toshiba_capture_source; - spec->num_mixers = 2; - spec->mixers[1] = cxt5047_hp_spk_mixers; - spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5047_toshiba_init_verbs; - codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; - break; -#ifdef CONFIG_SND_DEBUG - case CXT5047_TEST: - spec->input_mux = &cxt5047_test_capture_source; - spec->mixers[0] = cxt5047_test_mixer; - spec->init_verbs[0] = cxt5047_test_init_verbs; - codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; -#endif - } - spec->vmaster_nid = 0x13; - - switch (codec->subsystem_id >> 16) { - case 0x103c: - /* HP laptops have really bad sound over 0 dB on NID 0x10. - * Fix max PCM level to 0 dB (originally it has 0x1e steps - * with 0 dB offset 0x17) - */ - snd_hda_override_amp_caps(codec, 0x10, HDA_INPUT, - (0x17 << AC_AMPCAP_OFFSET_SHIFT) | - (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - } - - return 0; -} - -/* Conexant 5051 specific */ -static const hda_nid_t cxt5051_dac_nids[1] = { 0x10 }; -static const hda_nid_t cxt5051_adc_nids[2] = { 0x14, 0x15 }; - -static const struct hda_channel_mode cxt5051_modes[1] = { - { 2, NULL }, -}; - -static void cxt5051_update_speaker(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int pinctl; - /* headphone pin */ - pinctl = (spec->hp_present && spec->cur_eapd) ? PIN_HP : 0; - snd_hda_set_pin_ctl(codec, 0x16, pinctl); - /* speaker pin */ - pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; - snd_hda_set_pin_ctl(codec, 0x1a, pinctl); - /* on ideapad there is an additional speaker (subwoofer) to mute */ - if (spec->ideapad) - snd_hda_set_pin_ctl(codec, 0x1b, pinctl); -} - -/* turn on/off EAPD (+ mute HP) as a master switch */ -static int cxt5051_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - - if (!cxt_eapd_put(kcontrol, ucontrol)) - return 0; - cxt5051_update_speaker(codec); - return 1; -} - -/* toggle input of built-in and mic jack appropriately */ -static void cxt5051_portb_automic(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int present; - - if (!(spec->auto_mic & AUTO_MIC_PORTB)) - return; - present = snd_hda_jack_detect(codec, 0x17); - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_CONNECT_SEL, - present ? 0x01 : 0x00); -} - -/* switch the current ADC according to the jack state */ -static void cxt5051_portc_automic(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int present; - hda_nid_t new_adc; - - if (!(spec->auto_mic & AUTO_MIC_PORTC)) - return; - present = snd_hda_jack_detect(codec, 0x18); - if (present) - spec->cur_adc_idx = 1; - else - spec->cur_adc_idx = 0; - new_adc = spec->adc_nids[spec->cur_adc_idx]; - if (spec->cur_adc && spec->cur_adc != new_adc) { - /* stream is running, let's swap the current ADC */ - __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1); - spec->cur_adc = new_adc; - snd_hda_codec_setup_stream(codec, new_adc, - spec->cur_adc_stream_tag, 0, - spec->cur_adc_format); - } -} - -/* mute internal speaker if HP is plugged */ -static void cxt5051_hp_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - - spec->hp_present = snd_hda_jack_detect(codec, 0x16); - cxt5051_update_speaker(codec); -} - -/* unsolicited event for HP jack sensing */ -static void cxt5051_hp_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5051_hp_automute(codec); - break; - case CXT5051_PORTB_EVENT: - cxt5051_portb_automic(codec); - break; - case CXT5051_PORTC_EVENT: - cxt5051_portc_automic(codec); - break; - } -} - -static const struct snd_kcontrol_new cxt5051_playback_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5051_hp_master_sw_put, - .private_value = 0x1a, - }, - {} -}; - -static const struct snd_kcontrol_new cxt5051_capture_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Dock Mic Volume", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Dock Mic Switch", 0x15, 0x00, HDA_INPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5051_hp_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Volume", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Mic Switch", 0x15, 0x00, HDA_INPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x14, 0x00, HDA_INPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5051_f700_mixers[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x14, 0x01, HDA_INPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5051_toshiba_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), - {} -}; - -static const struct hda_verb cxt5051_init_verbs[] = { - /* Line in, Mic */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - /* SPK */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP, Amp */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Record selector: Internal mic */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, - /* SPDIF route: PCM */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* EAPD */ - {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5051_hp_dv6736_init_verbs[] = { - /* Line in, Mic */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, - /* SPK */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP, Amp */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Record selector: Internal mic */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* SPDIF route: PCM */ - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* EAPD */ - {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5051_f700_init_verbs[] = { - /* Line in, Mic */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, - /* SPK */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP, Amp */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Record selector: Internal mic */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* SPDIF route: PCM */ - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* EAPD */ - {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - { } /* end */ -}; - -static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid, - unsigned int event) -{ - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | event); -} - -static const struct hda_verb cxt5051_ideapad_init_verbs[] = { - /* Subwoofer */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - { } /* end */ -}; - -/* initialize jack-sensing, too */ -static int cxt5051_init(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - - conexant_init(codec); - - if (spec->auto_mic & AUTO_MIC_PORTB) - cxt5051_init_mic_port(codec, 0x17, CXT5051_PORTB_EVENT); - if (spec->auto_mic & AUTO_MIC_PORTC) - cxt5051_init_mic_port(codec, 0x18, CXT5051_PORTC_EVENT); - - if (codec->patch_ops.unsol_event) { - cxt5051_hp_automute(codec); - cxt5051_portb_automic(codec); - cxt5051_portc_automic(codec); - } - return 0; -} - - -enum { - CXT5051_LAPTOP, /* Laptops w/ EAPD support */ - CXT5051_HP, /* no docking */ - CXT5051_HP_DV6736, /* HP without mic switch */ - CXT5051_F700, /* HP Compaq Presario F700 */ - CXT5051_TOSHIBA, /* Toshiba M300 & co */ - CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */ - CXT5051_AUTO, /* auto-parser */ - CXT5051_MODELS -}; - -static const char *const cxt5051_models[CXT5051_MODELS] = { - [CXT5051_LAPTOP] = "laptop", - [CXT5051_HP] = "hp", - [CXT5051_HP_DV6736] = "hp-dv6736", - [CXT5051_F700] = "hp-700", - [CXT5051_TOSHIBA] = "toshiba", - [CXT5051_IDEAPAD] = "ideapad", - [CXT5051_AUTO] = "auto", -}; - -static const struct snd_pci_quirk cxt5051_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), - SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP), - SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700), - SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba M30x", CXT5051_TOSHIBA), - SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", - CXT5051_LAPTOP), - SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), - SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD), - {} -}; - -static int patch_cxt5051(struct hda_codec *codec) -{ - struct conexant_spec *spec; - int board_config; - - board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, - cxt5051_models, - cxt5051_cfg_tbl); - if (board_config < 0) - board_config = CXT5051_AUTO; /* model=auto as default */ - if (board_config == CXT5051_AUTO) - return patch_conexant_auto(codec); - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; - codec->spec = spec; - codec->pin_amp_workaround = 1; - - codec->patch_ops = conexant_patch_ops; - codec->patch_ops.init = cxt5051_init; - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(cxt5051_dac_nids); - spec->multiout.dac_nids = cxt5051_dac_nids; - spec->multiout.dig_out_nid = CXT5051_SPDIF_OUT; - spec->num_adc_nids = 1; /* not 2; via auto-mic switch */ - spec->adc_nids = cxt5051_adc_nids; - spec->num_mixers = 2; - spec->mixers[0] = cxt5051_capture_mixers; - spec->mixers[1] = cxt5051_playback_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = cxt5051_init_verbs; - spec->spdif_route = 0; - spec->num_channel_mode = ARRAY_SIZE(cxt5051_modes); - spec->channel_mode = cxt5051_modes; - spec->cur_adc = 0; - spec->cur_adc_idx = 0; - - set_beep_amp(spec, 0x13, 0, HDA_OUTPUT); - - codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; - - spec->auto_mic = AUTO_MIC_PORTB | AUTO_MIC_PORTC; - switch (board_config) { - case CXT5051_HP: - spec->mixers[0] = cxt5051_hp_mixers; - break; - case CXT5051_HP_DV6736: - spec->init_verbs[0] = cxt5051_hp_dv6736_init_verbs; - spec->mixers[0] = cxt5051_hp_dv6736_mixers; - spec->auto_mic = 0; - break; - case CXT5051_F700: - spec->init_verbs[0] = cxt5051_f700_init_verbs; - spec->mixers[0] = cxt5051_f700_mixers; - spec->auto_mic = 0; - break; - case CXT5051_TOSHIBA: - spec->mixers[0] = cxt5051_toshiba_mixers; - spec->auto_mic = AUTO_MIC_PORTB; - break; - case CXT5051_IDEAPAD: - spec->init_verbs[spec->num_init_verbs++] = - cxt5051_ideapad_init_verbs; - spec->ideapad = 1; - break; - } - - if (spec->beep_amp) - snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); - - return 0; -} - -/* Conexant 5066 specific */ - -static const hda_nid_t cxt5066_dac_nids[1] = { 0x10 }; -static const hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 }; -static const hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 }; -static const hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 }; - -static const struct hda_channel_mode cxt5066_modes[1] = { - { 2, NULL }, -}; - -#define HP_PRESENT_PORT_A (1 << 0) -#define HP_PRESENT_PORT_D (1 << 1) -#define hp_port_a_present(spec) ((spec)->hp_present & HP_PRESENT_PORT_A) -#define hp_port_d_present(spec) ((spec)->hp_present & HP_PRESENT_PORT_D) - -static void cxt5066_update_speaker(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int pinctl; - - codec_dbg(codec, - "CXT5066: update speaker, hp_present=%d, cur_eapd=%d\n", - spec->hp_present, spec->cur_eapd); - - /* Port A (HP) */ - pinctl = (hp_port_a_present(spec) && spec->cur_eapd) ? PIN_HP : 0; - snd_hda_set_pin_ctl(codec, 0x19, pinctl); - - /* Port D (HP/LO) */ - pinctl = spec->cur_eapd ? spec->port_d_mode : 0; - if (spec->dell_automute || spec->thinkpad) { - /* Mute if Port A is connected */ - if (hp_port_a_present(spec)) - pinctl = 0; - } else { - /* Thinkpad/Dell doesn't give pin-D status */ - if (!hp_port_d_present(spec)) - pinctl = 0; - } - snd_hda_set_pin_ctl(codec, 0x1c, pinctl); - - /* CLASS_D AMP */ - pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; - snd_hda_set_pin_ctl(codec, 0x1f, pinctl); -} - -/* turn on/off EAPD (+ mute HP) as a master switch */ -static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - - if (!cxt_eapd_put(kcontrol, ucontrol)) - return 0; - - cxt5066_update_speaker(codec); - return 1; -} - -/* toggle input of built-in digital mic and mic jack appropriately */ -static void cxt5066_vostro_automic(struct hda_codec *codec) -{ - unsigned int present; - - struct hda_verb ext_mic_present[] = { - /* enable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - - /* switch to external mic input */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0}, - - /* disable internal digital mic */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static const struct hda_verb ext_mic_absent[] = { - /* enable internal mic, port C */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - /* switch to internal mic input */ - {0x14, AC_VERB_SET_CONNECT_SEL, 2}, - - /* disable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - - present = snd_hda_jack_detect(codec, 0x1a); - if (present) { - codec_dbg(codec, "CXT5066: external microphone detected\n"); - snd_hda_sequence_write(codec, ext_mic_present); - } else { - codec_dbg(codec, "CXT5066: external microphone absent\n"); - snd_hda_sequence_write(codec, ext_mic_absent); - } -} - -/* toggle input of built-in digital mic and mic jack appropriately */ -static void cxt5066_ideapad_automic(struct hda_codec *codec) -{ - unsigned int present; - - struct hda_verb ext_mic_present[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static const struct hda_verb ext_mic_absent[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 2}, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - - present = snd_hda_jack_detect(codec, 0x1b); - if (present) { - codec_dbg(codec, "CXT5066: external microphone detected\n"); - snd_hda_sequence_write(codec, ext_mic_present); - } else { - codec_dbg(codec, "CXT5066: external microphone absent\n"); - snd_hda_sequence_write(codec, ext_mic_absent); - } -} - - -/* toggle input of built-in digital mic and mic jack appropriately */ -static void cxt5066_asus_automic(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x1b); - codec_dbg(codec, "CXT5066: external microphone present=%d\n", present); - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, - present ? 1 : 0); -} - - -/* toggle input of built-in digital mic and mic jack appropriately */ -static void cxt5066_hp_laptop_automic(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x1b); - codec_dbg(codec, "CXT5066: external microphone present=%d\n", present); - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, - present ? 1 : 3); -} - - -/* toggle input of built-in digital mic and mic jack appropriately - order is: external mic -> dock mic -> interal mic */ -static void cxt5066_thinkpad_automic(struct hda_codec *codec) -{ - unsigned int ext_present, dock_present; - - static const struct hda_verb ext_mic_present[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0}, - {0x17, AC_VERB_SET_CONNECT_SEL, 1}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static const struct hda_verb dock_mic_present[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0}, - {0x17, AC_VERB_SET_CONNECT_SEL, 0}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static const struct hda_verb ext_mic_absent[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 2}, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - - ext_present = snd_hda_jack_detect(codec, 0x1b); - dock_present = snd_hda_jack_detect(codec, 0x1a); - if (ext_present) { - codec_dbg(codec, "CXT5066: external microphone detected\n"); - snd_hda_sequence_write(codec, ext_mic_present); - } else if (dock_present) { - codec_dbg(codec, "CXT5066: dock microphone detected\n"); - snd_hda_sequence_write(codec, dock_mic_present); - } else { - codec_dbg(codec, "CXT5066: external microphone absent\n"); - snd_hda_sequence_write(codec, ext_mic_absent); - } -} - -/* mute internal speaker if HP is plugged */ -static void cxt5066_hp_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int portA, portD; - - /* Port A */ - portA = snd_hda_jack_detect(codec, 0x19); - - /* Port D */ - portD = snd_hda_jack_detect(codec, 0x1c); - - spec->hp_present = portA ? HP_PRESENT_PORT_A : 0; - spec->hp_present |= portD ? HP_PRESENT_PORT_D : 0; - codec_dbg(codec, "CXT5066: hp automute portA=%x portD=%x present=%d\n", - portA, portD, spec->hp_present); - cxt5066_update_speaker(codec); -} - -/* Dispatch the right mic autoswitch function */ -static void cxt5066_automic(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - - if (spec->dell_vostro) - cxt5066_vostro_automic(codec); - else if (spec->ideapad) - cxt5066_ideapad_automic(codec); - else if (spec->thinkpad) - cxt5066_thinkpad_automic(codec); - else if (spec->hp_laptop) - cxt5066_hp_laptop_automic(codec); - else if (spec->asus) - cxt5066_asus_automic(codec); -} - -/* unsolicited event for jack sensing */ -static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) -{ - codec_dbg(codec, "CXT5066: unsol event %x (%x)\n", res, res >> 26); - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5066_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: - cxt5066_automic(codec); - break; - } -} - - -static const struct hda_input_mux cxt5066_analog_mic_boost = { - .num_items = 5, - .items = { - { "0dB", 0 }, - { "10dB", 1 }, - { "20dB", 2 }, - { "30dB", 3 }, - { "40dB", 4 }, - }, -}; - -static void cxt5066_set_mic_boost(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - snd_hda_codec_write_cache(codec, 0x17, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | - cxt5066_analog_mic_boost.items[spec->mic_boost].index); - if (spec->ideapad || spec->thinkpad) { - /* adjust the internal mic as well...it is not through 0x17 */ - snd_hda_codec_write_cache(codec, 0x23, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_INPUT | - cxt5066_analog_mic_boost. - items[spec->mic_boost].index); - } -} - -static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - return snd_hda_input_mux_info(&cxt5066_analog_mic_boost, uinfo); -} - -static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - ucontrol->value.enumerated.item[0] = spec->mic_boost; - return 0; -} - -static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; - unsigned int idx; - idx = ucontrol->value.enumerated.item[0]; - if (idx >= imux->num_items) - idx = imux->num_items - 1; - - spec->mic_boost = idx; - cxt5066_set_mic_boost(codec); - return 1; -} - -static void conexant_check_dig_outs(struct hda_codec *codec, - const hda_nid_t *dig_pins, - int num_pins) -{ - struct conexant_spec *spec = codec->spec; - hda_nid_t *nid_loc = &spec->multiout.dig_out_nid; - int i; - - for (i = 0; i < num_pins; i++, dig_pins++) { - unsigned int cfg = snd_hda_codec_get_pincfg(codec, *dig_pins); - if (get_defcfg_connect(cfg) == AC_JACK_PORT_NONE) - continue; - if (snd_hda_get_connections(codec, *dig_pins, nid_loc, 1) != 1) - continue; - } -} - -static const struct hda_input_mux cxt5066_capture_source = { - .num_items = 4, - .items = { - { "Mic B", 0 }, - { "Mic C", 1 }, - { "Mic E", 2 }, - { "Mic F", 3 }, - }, -}; - -static const struct hda_bind_ctls cxt5066_bind_capture_vol_others = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT), - 0 - }, -}; - -static const struct hda_bind_ctls cxt5066_bind_capture_sw_others = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT), - 0 - }, -}; - -static const struct snd_kcontrol_new cxt5066_mixer_master[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5066_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5066_hp_master_sw_put, - .private_value = 0x1d, - }, - - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Mic Boost Capture Enum", - .info = cxt5066_mic_boost_mux_enum_info, - .get = cxt5066_mic_boost_mux_enum_get, - .put = cxt5066_mic_boost_mux_enum_put, - }, - - HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others), - HDA_BIND_SW("Capture Switch", &cxt5066_bind_capture_sw_others), - {} -}; - -static const struct snd_kcontrol_new cxt5066_vostro_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Internal Mic Boost Capture Enum", - .info = cxt5066_mic_boost_mux_enum_info, - .get = cxt5066_mic_boost_mux_enum_get, - .put = cxt5066_mic_boost_mux_enum_put, - .private_value = 0x23 | 0x100, - }, - {} -}; - -static const struct hda_verb cxt5066_init_verbs[] = { - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ - {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ - - /* Speakers */ - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* HP, Amp */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - - /* no digital microphone support yet */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Audio input selector */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, - - /* SPDIF route: PCM */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, - - {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* EAPD */ - {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - - /* not handling these yet */ - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - { } /* end */ -}; - -static const struct hda_verb cxt5066_init_verbs_vostro[] = { - /* Port A: headphones */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* Port B: external microphone */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port C: unused */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port D: unused */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port E: unused, but has primary EAPD */ - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - - /* Port F: unused */ - {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port G: internal speakers */ - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* DAC2: unused */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - - /* Digital microphone port */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - /* Audio input selectors */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - - /* Disable SPDIF */ - {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* enable unsolicited events for Port A and B */ - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5066_init_verbs_ideapad[] = { - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ - {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ - - /* Speakers */ - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* HP, Amp */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x14, AC_VERB_SET_CONNECT_SEL, 2}, /* default to internal mic */ - - /* Audio input selector */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x2}, - {0x17, AC_VERB_SET_CONNECT_SEL, 1}, /* route ext mic */ - - /* SPDIF route: PCM */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, - - {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* internal microphone */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* enable internal mic */ - - /* EAPD */ - {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5066_init_verbs_thinkpad[] = { - {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ - - /* Port G: internal speakers */ - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* Port A: HP, Amp */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* Port B: Mic Dock */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port C: Mic */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port D: HP Dock, Amp */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x14, AC_VERB_SET_CONNECT_SEL, 2}, /* default to internal mic */ - - /* Audio input selector */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x2}, - {0x17, AC_VERB_SET_CONNECT_SEL, 1}, /* route ext mic */ - - /* SPDIF route: PCM */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, - - {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* internal microphone */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* enable internal mic */ - - /* EAPD */ - {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - - /* enable unsolicited events for Port A, B, C and D */ - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5066_init_verbs_portd_lo[] = { - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - { } /* end */ -}; - - -static const struct hda_verb cxt5066_init_verbs_hp_laptop[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -/* initialize jack-sensing, too */ -static int cxt5066_init(struct hda_codec *codec) -{ - codec_dbg(codec, "CXT5066: init\n"); - conexant_init(codec); - if (codec->patch_ops.unsol_event) { - cxt5066_hp_automute(codec); - cxt5066_automic(codec); - } - cxt5066_set_mic_boost(codec); - return 0; -} - -enum { - CXT5066_LAPTOP, /* Laptops w/ EAPD support */ - CXT5066_DELL_LAPTOP, /* Dell Laptop */ - CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */ - CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ - CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ - CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ - CXT5066_HP_LAPTOP, /* HP Laptop */ - CXT5066_AUTO, /* BIOS auto-parser */ - CXT5066_MODELS -}; - -static const char * const cxt5066_models[CXT5066_MODELS] = { - [CXT5066_LAPTOP] = "laptop", - [CXT5066_DELL_LAPTOP] = "dell-laptop", - [CXT5066_DELL_VOSTRO] = "dell-vostro", - [CXT5066_IDEAPAD] = "ideapad", - [CXT5066_THINKPAD] = "thinkpad", - [CXT5066_ASUS] = "asus", - [CXT5066_HP_LAPTOP] = "hp-laptop", - [CXT5066_AUTO] = "auto", -}; - -static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { - SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), - SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), - SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), - SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS), - SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), - SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS), - SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", - CXT5066_LAPTOP), - SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), - SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), - SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), - {} -}; - -static int patch_cxt5066(struct hda_codec *codec) -{ - struct conexant_spec *spec; - int board_config; - - board_config = snd_hda_check_board_config(codec, CXT5066_MODELS, - cxt5066_models, cxt5066_cfg_tbl); - if (board_config < 0) - board_config = CXT5066_AUTO; /* model=auto as default */ - if (board_config == CXT5066_AUTO) - return patch_conexant_auto(codec); - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; - codec->spec = spec; - - codec->patch_ops = conexant_patch_ops; - codec->patch_ops.init = conexant_init; - - spec->dell_automute = 0; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids); - spec->multiout.dac_nids = cxt5066_dac_nids; - conexant_check_dig_outs(codec, cxt5066_digout_pin_nids, - ARRAY_SIZE(cxt5066_digout_pin_nids)); - spec->num_adc_nids = 1; - spec->adc_nids = cxt5066_adc_nids; - spec->capsrc_nids = cxt5066_capsrc_nids; - spec->input_mux = &cxt5066_capture_source; - - spec->port_d_mode = PIN_HP; - - spec->num_init_verbs = 1; - spec->init_verbs[0] = cxt5066_init_verbs; - spec->num_channel_mode = ARRAY_SIZE(cxt5066_modes); - spec->channel_mode = cxt5066_modes; - spec->cur_adc = 0; - spec->cur_adc_idx = 0; - - set_beep_amp(spec, 0x13, 0, HDA_OUTPUT); - - switch (board_config) { - default: - case CXT5066_LAPTOP: - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - break; - case CXT5066_DELL_LAPTOP: - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - - spec->port_d_mode = PIN_OUT; - spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_portd_lo; - spec->num_init_verbs++; - spec->dell_automute = 1; - break; - case CXT5066_ASUS: - case CXT5066_HP_LAPTOP: - codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_unsol_event; - spec->init_verbs[spec->num_init_verbs] = - cxt5066_init_verbs_hp_laptop; - spec->num_init_verbs++; - spec->hp_laptop = board_config == CXT5066_HP_LAPTOP; - spec->asus = board_config == CXT5066_ASUS; - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - /* no S/PDIF out */ - if (board_config == CXT5066_HP_LAPTOP) - spec->multiout.dig_out_nid = 0; - /* input source automatically selected */ - spec->input_mux = NULL; - spec->port_d_mode = 0; - spec->mic_boost = 3; /* default 30dB gain */ - break; - - case CXT5066_DELL_VOSTRO: - codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_unsol_event; - spec->init_verbs[0] = cxt5066_init_verbs_vostro; - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers; - spec->port_d_mode = 0; - spec->dell_vostro = 1; - spec->mic_boost = 3; /* default 30dB gain */ - - /* no S/PDIF out */ - spec->multiout.dig_out_nid = 0; - - /* input source automatically selected */ - spec->input_mux = NULL; - break; - case CXT5066_IDEAPAD: - codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_unsol_event; - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - spec->init_verbs[0] = cxt5066_init_verbs_ideapad; - spec->port_d_mode = 0; - spec->ideapad = 1; - spec->mic_boost = 2; /* default 20dB gain */ - - /* no S/PDIF out */ - spec->multiout.dig_out_nid = 0; - - /* input source automatically selected */ - spec->input_mux = NULL; - break; - case CXT5066_THINKPAD: - codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_unsol_event; - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - spec->init_verbs[0] = cxt5066_init_verbs_thinkpad; - spec->thinkpad = 1; - spec->port_d_mode = PIN_OUT; - spec->mic_boost = 2; /* default 20dB gain */ - - /* no S/PDIF out */ - spec->multiout.dig_out_nid = 0; - - /* input source automatically selected */ - spec->input_mux = NULL; - break; - } - - if (spec->beep_amp) - snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); - - return 0; -} - -#endif /* ENABLE_CXT_STATIC_QUIRKS */ - - /* * Automatic parser for CX20641 & co */ @@ -3487,35 +881,28 @@ static int patch_conexant_auto(struct hda_codec *codec) return err; } -#ifndef ENABLE_CXT_STATIC_QUIRKS -#define patch_cxt5045 patch_conexant_auto -#define patch_cxt5047 patch_conexant_auto -#define patch_cxt5051 patch_conexant_auto -#define patch_cxt5066 patch_conexant_auto -#endif - /* */ static const struct hda_codec_preset snd_hda_preset_conexant[] = { { .id = 0x14f15045, .name = "CX20549 (Venice)", - .patch = patch_cxt5045 }, + .patch = patch_conexant_auto }, { .id = 0x14f15047, .name = "CX20551 (Waikiki)", - .patch = patch_cxt5047 }, + .patch = patch_conexant_auto }, { .id = 0x14f15051, .name = "CX20561 (Hermosa)", - .patch = patch_cxt5051 }, + .patch = patch_conexant_auto }, { .id = 0x14f15066, .name = "CX20582 (Pebble)", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f15067, .name = "CX20583 (Pebble HSF)", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f15068, .name = "CX20584", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f15069, .name = "CX20585", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f1506c, .name = "CX20588", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f1506e, .name = "CX20590", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f15097, .name = "CX20631", .patch = patch_conexant_auto }, { .id = 0x14f15098, .name = "CX20632", -- 1.9.3

1 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Kill snd_printd*() in HDMI debug / info prints
by Takashi Iwai 18 Jun '14

18 Jun '14

Pass codec instance to each function that still prints info and debug outputs via snd_printd*(). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_eld.c | 46 ++++++++++++++++++++++------------------------ sound/pci/hda/hda_local.h | 4 ++-- sound/pci/hda/patch_hdmi.c | 11 ++++++----- 3 files changed, 30 insertions(+), 31 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 46690a7f48f6..e1cd34d9011d 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -167,7 +167,8 @@ static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid, (buf[byte] >> (lowbit)) & ((1 << (bits)) - 1); \ }) -static void hdmi_update_short_audio_desc(struct cea_sad *a, +static void hdmi_update_short_audio_desc(struct hda_codec *codec, + struct cea_sad *a, const unsigned char *buf) { int i; @@ -188,8 +189,7 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, a->format = GRAB_BITS(buf, 0, 3, 4); switch (a->format) { case AUDIO_CODING_TYPE_REF_STREAM_HEADER: - snd_printd(KERN_INFO - "HDMI: audio coding type 0 not expected\n"); + codec_info(codec, "HDMI: audio coding type 0 not expected\n"); break; case AUDIO_CODING_TYPE_LPCM: @@ -233,9 +233,9 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, a->format = GRAB_BITS(buf, 2, 3, 5); if (a->format == AUDIO_CODING_XTYPE_HE_REF_CT || a->format >= AUDIO_CODING_XTYPE_FIRST_RESERVED) { - snd_printd(KERN_INFO - "HDMI: audio coding xtype %d not expected\n", - a->format); + codec_info(codec, + "HDMI: audio coding xtype %d not expected\n", + a->format); a->format = 0; } else a->format += AUDIO_CODING_TYPE_HE_AAC - @@ -247,7 +247,7 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, /* * Be careful, ELD buf could be totally rubbish! */ -int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e, +int snd_hdmi_parse_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e, const unsigned char *buf, int size) { int mnl; @@ -256,8 +256,7 @@ int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e, e->eld_ver = GRAB_BITS(buf, 0, 3, 5); if (e->eld_ver != ELD_VER_CEA_861D && e->eld_ver != ELD_VER_PARTIAL) { - snd_printd(KERN_INFO "HDMI: Unknown ELD version %d\n", - e->eld_ver); + codec_info(codec, "HDMI: Unknown ELD version %d\n", e->eld_ver); goto out_fail; } @@ -280,20 +279,20 @@ int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e, e->product_id = get_unaligned_le16(buf + 18); if (mnl > ELD_MAX_MNL) { - snd_printd(KERN_INFO "HDMI: MNL is reserved value %d\n", mnl); + codec_info(codec, "HDMI: MNL is reserved value %d\n", mnl); goto out_fail; } else if (ELD_FIXED_BYTES + mnl > size) { - snd_printd(KERN_INFO "HDMI: out of range MNL %d\n", mnl); + codec_info(codec, "HDMI: out of range MNL %d\n", mnl); goto out_fail; } else strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl + 1); for (i = 0; i < e->sad_count; i++) { if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) { - snd_printd(KERN_INFO "HDMI: out of range SAD %d\n", i); + codec_info(codec, "HDMI: out of range SAD %d\n", i); goto out_fail; } - hdmi_update_short_audio_desc(e->sad + i, + hdmi_update_short_audio_desc(codec, e->sad + i, buf + ELD_FIXED_BYTES + mnl + 3 * i); } @@ -394,7 +393,8 @@ static void hdmi_print_pcm_rates(int pcm, char *buf, int buflen) #define SND_PRINT_RATES_ADVISED_BUFSIZE 80 -static void hdmi_show_short_audio_desc(struct cea_sad *a) +static void hdmi_show_short_audio_desc(struct hda_codec *codec, + struct cea_sad *a) { char buf[SND_PRINT_RATES_ADVISED_BUFSIZE]; char buf2[8 + SND_PRINT_BITS_ADVISED_BUFSIZE] = ", bits ="; @@ -412,12 +412,10 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a) else buf2[0] = '\0'; - _snd_printd(SND_PR_VERBOSE, "HDMI: supports coding type %s:" - " channels = %d, rates =%s%s\n", - cea_audio_coding_type_names[a->format], - a->channels, - buf, - buf2); + codec_dbg(codec, + "HDMI: supports coding type %s: channels = %d, rates =%s%s\n", + cea_audio_coding_type_names[a->format], + a->channels, buf, buf2); } void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen) @@ -432,22 +430,22 @@ void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen) buf[j] = '\0'; /* necessary when j == 0 */ } -void snd_hdmi_show_eld(struct parsed_hdmi_eld *e) +void snd_hdmi_show_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e) { int i; - _snd_printd(SND_PR_VERBOSE, "HDMI: detected monitor %s at connection type %s\n", + codec_dbg(codec, "HDMI: detected monitor %s at connection type %s\n", e->monitor_name, eld_connection_type_names[e->conn_type]); if (e->spk_alloc) { char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf)); - _snd_printd(SND_PR_VERBOSE, "HDMI: available speakers:%s\n", buf); + codec_dbg(codec, "HDMI: available speakers:%s\n", buf); } for (i = 0; i < e->sad_count; i++) - hdmi_show_short_audio_desc(e->sad + i); + hdmi_show_short_audio_desc(codec, e->sad + i); } #ifdef CONFIG_PROC_FS diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index e51d15529215..7820fd2d9ba8 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -738,9 +738,9 @@ struct hdmi_eld { int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid); int snd_hdmi_get_eld(struct hda_codec *codec, hda_nid_t nid, unsigned char *buf, int *eld_size); -int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e, +int snd_hdmi_parse_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e, const unsigned char *buf, int size); -void snd_hdmi_show_eld(struct parsed_hdmi_eld *e); +void snd_hdmi_show_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e); void snd_hdmi_eld_update_pcm_info(struct parsed_hdmi_eld *e, struct hda_pcm_stream *hinfo); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b4218a19df22..39f86fad0330 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -648,7 +648,8 @@ static int get_channel_allocation_order(int ca) * * TODO: it could select the wrong CA from multiple candidates. */ -static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels) +static int hdmi_channel_allocation(struct hda_codec *codec, + struct hdmi_eld *eld, int channels) { int i; int ca = 0; @@ -694,7 +695,7 @@ static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels) } snd_print_channel_allocation(eld->info.spk_alloc, buf, sizeof(buf)); - snd_printdd("HDMI: select CA 0x%x for %d-channel allocation: %s\n", + codec_dbg(codec, "HDMI: select CA 0x%x for %d-channel allocation: %s\n", ca, channels, buf); return ca; @@ -1135,7 +1136,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (!non_pcm && per_pin->chmap_set) ca = hdmi_manual_channel_allocation(channels, per_pin->chmap); else - ca = hdmi_channel_allocation(eld, channels); + ca = hdmi_channel_allocation(codec, eld, channels); if (ca < 0) ca = 0; @@ -1561,13 +1562,13 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) eld->eld_valid = false; else { memset(&eld->info, 0, sizeof(struct parsed_hdmi_eld)); - if (snd_hdmi_parse_eld(&eld->info, eld->eld_buffer, + if (snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer, eld->eld_size) < 0) eld->eld_valid = false; } if (eld->eld_valid) { - snd_hdmi_show_eld(&eld->info); + snd_hdmi_show_eld(codec, &eld->info); update_eld = true; } else if (repoll) { -- 1.9.3

1 1

[alsa-devel] ALSA 1.0.28 release
by Jaroslav Kysela 18 Jun '14

18 Jun '14

Hello all, the ALSA 1.0.28 packages are available on the main ALSA project web site: http://www.alsa-project.org . Thank you to all authors of code updates. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

1 0

Re: [alsa-devel] Plug In Creation - aplay can't find existing .so
by Takashi Iwai 18 Jun '14

18 Jun '14

At Tue, 17 Jun 2014 15:08:31 -0400, Brandon Yates wrote: > > Interestingly strace shows the failure occurs on the open call (not > dlopen) > > open("/usr/lib/i386-linux-gnu/alsa-lib/libasound_module_pcm_adt_dsp.so", > O_RDONLY|O_CLOEXEC) = 3 This succeeded and returned the proper fd number 3. > running ldd on my shared library shows undefined symbols that are part > of libasound, for example > undefined symbol: snd_pcm_ioplug_create > > I wonder if I have to build the .so a different way so these can be > resolved? Any advice would be greatly appreciated. Maybe you forgot to link your shared object with -lasound -lm? Takashi > > > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Friday, June 13, 2014 8:42 AM > To: Brandon Yates > Cc: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Plug In Creation - aplay can't find existing > .so > > At Tue, 10 Jun 2014 13:55:39 -0400, > Brandon Yates wrote: > > > > Hello, > > > > I am working on a new pcm plugin. At this point it is basically just a > > > Hello World plugin, to make sure I have the tools working and > > understand the workflow. I created a simple file which is just > basically: > > > > #include <alsa/asoundlib.h> > > #include <alsa/pcm_external.h> > > SND_PCM_PLUGIN_DEFINE_FUNC(my_mod) > > { > > printf("hello world\n"); > > return 0; > > } > > SND_PCM_PLUGIN_SYMBOL(my_mod); > > > > > > I compiled this as a shared lib named > 'libasound_module_pcm_my_mod.so' > > and copied it to '/usr/lib/i386-linux-gnu/alsa-lib/' > > > > I edited my ~/.asoundrc to create a new device that uses this plugin. > > Then I try to use it with `aplay --device=my_mod_test out.pcm` > > > > I get the error message > > > > ALSA lib dlmisc.c:254:(snd1_dlobj_cache_get) Cannot open shared > > library > > /usr/lib/i386-linux-gnu/alsa-lib/libasound_module_pcm_my_mod.so > > aplay: main:682: audio open error: No such device or address > > > > > > But this file actually does exist, I just copied it there. > > > > > > Any idea why this happens? > > Try strace to see what happens. If it fails in dlopen(), it's likely > because your shared object is invalid. > > > Takashi >

1 0

[alsa-devel] Plug In Creation - aplay can't find existing .so
by Brandon Yates 18 Jun '14

18 Jun '14

Hello, I am working on a new pcm plugin. At this point it is basically just a Hello World plugin, to make sure I have the tools working and understand the workflow. I created a simple file which is just basically: #include <alsa/asoundlib.h> #include <alsa/pcm_external.h> SND_PCM_PLUGIN_DEFINE_FUNC(my_mod) { printf("hello world\n"); return 0; } SND_PCM_PLUGIN_SYMBOL(my_mod); I compiled this as a shared lib named 'libasound_module_pcm_my_mod.so' and copied it to '/usr/lib/i386-linux-gnu/alsa-lib/' I edited my ~/.asoundrc to create a new device that uses this plugin. Then I try to use it with `aplay --device=my_mod_test out.pcm` I get the error message ALSA lib dlmisc.c:254:(snd1_dlobj_cache_get) Cannot open shared library /usr/lib/i386-linux-gnu/alsa-lib/libasound_module_pcm_my_mod.so aplay: main:682: audio open error: No such device or address But this file actually does exist, I just copied it there. Any idea why this happens? Thanks much, Brandon

2 3

[alsa-devel] Enable more HP machine to enable mute led
by Kailang 18 Jun '14

18 Jun '14

Hi Takashi, Add more HP machine to support mute led functions. Thanks. BR, Kailang

2 4

[alsa-devel] Possible regression caused by "ASoC: rsnd: add rsnd_gen_dma_addr() for DMAC addr"
by Simon Horman 18 Jun '14

18 Jun '14

Hi Morimoto-san, Hi All, ad32d0c7b0e99343 ("ASoC: rsnd: add rsnd_gen_dma_addr() for DMAC addr"), which is included in v3.16-rc1, appears to cause the following panic on the bockw board when booted using the bockw_defconfig. Booting Linux on physical CPU 0x0 Linux version 3.15.0-rc1-00017-gad32d0c (horms(a)ayumi.isobedori.kobe.vergenet.net) (gcc version 4.6.3 (GCC) ) #123 Tue Jun 17 20:56:53 JST 2014 CPU: ARMv7 Processor [412fc094] revision 4 (ARMv7), cr=10c53c7d CPU: PIPT / VIPT nonaliasing data cache, VIPT aliasing instruction cache Machine model: bockw debug: ignoring loglevel setting. Memory policy: Data cache writeback On node 0 totalpages: 65536 free_area_init_node: node 0, pgdat c03f7f88, node_mem_map cfdfb000 Normal zone: 512 pages used for memmap Normal zone: 0 pages reserved Normal zone: 65536 pages, LIFO batch:15 DT missing boot CPU MPIDR[23:0], fall back to default cpu_logical_map CPU: All CPU(s) started in SVC mode. pcpu-alloc: s0 r0 d32768 u32768 alloc=1*32768 pcpu-alloc: [0] 0 Built 1 zonelists in Zone order, mobility grouping on. Total pages: 65024 Kernel command line: console=ttySC0,115200 ignore_loglevel ip=dhcp root=/dev/nfs rw PID hash table entries: 1024 (order: 0, 4096 bytes) Dentry cache hash table entries: 32768 (order: 5, 131072 bytes) Inode-cache hash table entries: 16384 (order: 4, 65536 bytes) Memory: 255600K/262144K available (3123K kernel code, 146K rwdata, 628K rodata, 130K init, 165K bss, 6544K reserved, 0K highmem) Virtual kernel memory layout: vector : 0xffff0000 - 0xffff1000 ( 4 kB) fixmap : 0xfff00000 - 0xfffe0000 ( 896 kB) vmalloc : 0xd0800000 - 0xff000000 ( 744 MB) lowmem : 0xc0000000 - 0xd0000000 ( 256 MB) pkmap : 0xbfe00000 - 0xc0000000 ( 2 MB) .text : 0xc0008000 - 0xc03b1e14 (3752 kB) .init : 0xc03b2000 - 0xc03d2a38 ( 131 kB) .data : 0xc03d4000 - 0xc03f8a20 ( 147 kB) .bss : 0xc03f8a20 - 0xc0422100 ( 166 kB) NR_IRQS:16 nr_irqs:16 16 GIC CPU mask not found - kernel will fail to boot. GIC CPU mask not found - kernel will fail to boot. sched_clock: 32 bits at 1kHz, resolution 976562ns, wraps every 2097152000000000ns Calibrating delay loop (skipped) preset value.. 532.00 BogoMIPS (lpj=260000) pid_max: default: 32768 minimum: 301 Mount-cache hash table entries: 1024 (order: 0, 4096 bytes) Mountpoint-cache hash table entries: 1024 (order: 0, 4096 bytes) CPU: Testing write buffer coherency: ok Setting up static identity map for 0x60312b28 - 0x60312b5c devtmpfs: initialized VFP support v0.3: implementor 41 architecture 3 part 30 variant 9 rev 4 pinctrl core: initialized pinctrl subsystem regulator-dummy: no parameters NET: Registered protocol family 16 DMA: preallocated 256 KiB pool for atomic coherent allocations renesas_intc_irqpin renesas_intc_irqpin: driving 4 irqs L310 cache controller enabled l2x0: 16 ways, CACHE_ID 0x410000c6, AUX_CTRL 0x42460000, Cache size: 1024 kB sh-pfc pfc-r8a7778: r8a7778_pfc support registered fixed-dummy: no parameters fixed-3.3V: at 3300 mV SCSI subsystem initialized usbcore: registered new interface driver usbfs usbcore: registered new interface driver hub usbcore: registered new device driver usb Linux video capture interface: v2.00 sh_tmu sh_tmu.0: used for clock events sh_tmu sh_tmu.0: used for periodic clock events sh_tmu sh_tmu.1: used as clock source Advanced Linux Sound Architecture Driver Initialized. Switched to clocksource sh_tmu.1 NET: Registered protocol family 2 TCP established hash table entries: 2048 (order: 1, 8192 bytes) TCP bind hash table entries: 2048 (order: 1, 8192 bytes) TCP: Hash tables configured (established 2048 bind 2048) TCP: reno registered UDP hash table entries: 256 (order: 0, 4096 bytes) UDP-Lite hash table entries: 256 (order: 0, 4096 bytes) NET: Registered protocol family 1 RPC: Registered named UNIX socket transport module. RPC: Registered udp transport module. RPC: Registered tcp transport module. RPC: Registered tcp NFSv4.1 backchannel transport module. futex hash table entries: 256 (order: -1, 3072 bytes) NFS: Registering the id_resolver key type Key type id_resolver registered Key type id_legacy registered nfs4filelayout_init: NFSv4 File Layout Driver Registering... Block layer SCSI generic (bsg) driver version 0.4 loaded (major 253) io scheduler noop registered io scheduler deadline registered (default) SuperH (H)SCI(F) driver initialized sh-sci.0: ttySC0 at MMIO 0xffe40000 (irq = 102, base_baud = 0) is a scif console [ttySC0] enabled sh-sci.1: ttySC1 at MMIO 0xffe41000 (irq = 103, base_baud = 0) is a scif sh-sci.2: ttySC2 at MMIO 0xffe42000 (irq = 104, base_baud = 0) is a scif sh-sci.3: ttySC3 at MMIO 0xffe43000 (irq = 105, base_baud = 0) is a scif sh-sci.4: ttySC4 at MMIO 0xffe44000 (irq = 106, base_baud = 0) is a scif sh-sci.5: ttySC5 at MMIO 0xffe45000 (irq = 107, base_baud = 0) is a scif m25p80 spi0.0: found at26df081a, expected s25fl008k m25p80 spi0.0: at26df081a (1024 Kbytes) Creating 1 MTD partitions on "m25p80": 0x000000000000-0x000000100000 : "data(spi)" libphy: smsc911x-mdio: probed smsc911x smsc911x eth0: attached PHY driver [Generic PHY] (mii_bus:phy_addr=smsc911x-fffffff:01, irq=-1) smsc911x smsc911x eth0: MAC Address: 2e:09:0a:00:2f:04 ehci_hcd: USB 2.0 'Enhanced' Host Controller (EHCI) Driver ehci-platform: EHCI generic platform driver ohci_hcd: USB 1.1 'Open' Host Controller (OHCI) Driver ohci-platform: OHCI generic platform driver usbcore: registered new interface driver usb-storage rtc-rx8581 0-0051: chip found, driver version 0.1 rtc-rx8581 0-0051: low voltage detected, date/time is not reliable. rtc-rx8581 0-0051: rtc core: registered rtc-rx8581 as rtc0 i2c-rcar i2c-rcar.0: probed i2c-rcar i2c-rcar.1: probed i2c-rcar i2c-rcar.2: probed i2c-rcar i2c-rcar.3: probed soc-camera-pdrv soc-camera-pdrv.0: Probing soc-camera-pdrv.0 ml86v7667 0-0041: chip found @ 0x41 (i2c-rcar) soc-camera-pdrv soc-camera-pdrv.1: Probing soc-camera-pdrv.1 ml86v7667 0-0043: chip found @ 0x43 (i2c-rcar) sh_mobile_sdhi.0 supply vmmc not found, using dummy regulator sh_mobile_sdhi sh_mobile_sdhi.0: Failed getting OCR mask: -22 sh_mobile_sdhi sh_mobile_sdhi.0: could not set regulator OCR (-22) sh_mobile_sdhi sh_mobile_sdhi.0: mmc0 base at 0xffe4c000 clock rate 66 MHz sh_mmcif sh_mmcif: driver version 2010-04-28 asoc-simple-card asoc-simple-card.0: ASoC: CPU DAI rsnd-dai.0 not registered platform asoc-simple-card.0: Driver asoc-simple-card requests probe deferral asoc-simple-card asoc-simple-card.1: ASoC: CPU DAI rsnd-dai.1 not registered platform asoc-simple-card.1: Driver asoc-simple-card requests probe deferral asoc-simple-card asoc-simple-card.2: ASoC: CPU DAI rsnd-dai.2 not registered platform asoc-simple-card.2: Driver asoc-simple-card requests probe deferral asoc-simple-card asoc-simple-card.3: ASoC: CPU DAI rsnd-dai.3 not registered platform asoc-simple-card.3: Driver asoc-simple-card requests probe deferral asoc-simple-card asoc-simple-card.4: ASoC: CPU DAI rsnd-dai.4 not registered platform asoc-simple-card.4: Driver asoc-simple-card requests probe deferral Unable to handle kernel NULL pointer dereference at virtual address 00000000 pgd = c0004000 [00000000] *pgd=00000000 Internal error: Oops: 5 [#1] ARM CPU: 0 PID: 1 Comm: swapper Not tainted 3.15.0-rc1-00017-gad32d0c #123 task: cf827a40 ti: cf828000 task.ti: cf828000 PC is at rsnd_gen_dma_addr+0x40/0x1dc LR is at rsnd_gen_dma_addr+0x38/0x1dc pc : [<c026402c>] lr : [<c0264024>] psr: 60000113 sp : cf829c38 ip : 00000003 fp : cf829cdc r10: cf8e5600 r9 : 00000000 r8 : cf8e5610 r7 : 00000001 r6 : cf9a3e0c r5 : cf9a7c24 r4 : cf829ce8 r3 : 00000003 r2 : 00000000 r1 : 00000200 r0 : 00000000 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c53c7d Table: 60004059 DAC: 00000015 Process swapper (pid: 1, stack limit = 0xcf828238) Stack: (0xcf829c38 to 0xcf82a000) 9c20: c0088644 c00886f8 9c40: cf9b01e0 c032a990 00000000 cf9b5990 cf829c8c cf829c60 c016f7c8 c016e798 9c60: c016f70c cf9b01e0 c03ea34c c03ea35c 0000001f c016fab8 00200200 cf829d2c 9c80: cf829ca4 cf829c90 c016d628 c016f718 00000003 cf9b04d0 cf829cdc cf829ca8 9ca0: c016dc04 c0311854 00000000 cf9b01e0 cf829cd4 cf9a3e0c 00000001 cf9b5990 9cc0: cf829d1c cf8e5610 00000000 0000001f cf829d5c cf829ce0 c0263348 c0263ff8 9ce0: 0000001f ffffffff cf829d14 cf8e1f80 cf9a3630 cf9a7d90 00000060 00000004 9d00: 00000000 c03e0fd8 c03e1098 cf9a7dc8 cf829d34 cf829d20 c0262c14 c0007874 9d20: cf8e1f80 cf9a3630 cf829d7c 00000200 c0262e98 cf8e5600 c03e0fb4 00000005 9d40: cf9a7c1c 00000000 00000000 cf9b5990 cf829d7c cf829d60 c0265d08 c0263208 9d60: c03f40e0 cf9a7c18 cf9a7c10 cf9a7c10 cf829dcc cf829d80 c0262850 c0265c90 9d80: cf829db4 cf8e5610 c02641c8 c0265f6c c0264c7c c026663c c0265630 c0262cd8 9da0: 00000000 cf8e5610 c03f3d14 c0416bb8 00000000 00000000 c03d29e0 c03f3d14 9dc0: cf829de4 cf829dd0 c018fd88 c026274c cf8e5610 c0416bb0 cf829e0c cf829de8 9de0: c018ecb8 c018fd74 cf8e5610 c03f3d14 cf8e5644 00000000 c03b24c0 00000000 9e00: cf829e2c cf829e10 c018ee5c c018ebac cf8e8e3c c03f3d14 00000000 c018edc8 9e20: cf829e54 cf829e30 c018d32c c018edd4 cf818ecc cf8e8e30 cf9b5a34 c03f3d14 9e40: cf9b5a00 c03eb348 cf829e64 cf829e58 c018e990 c018d2dc cf829e8c cf829e68 9e60: c018db54 c018e978 c0380c40 cf829e78 c03f3d14 00000006 c03cd314 c03c69dc 9e80: cf829ea4 cf829e90 c018f27c c018da80 c03cd334 00000006 cf829eb4 cf829ea8 9ea0: c0190590 c018f208 cf829ec4 cf829eb8 c03c69f4 c019054c cf829f54 cf829ec8 9ec0: c03b2af4 c03c69e8 c040c9b4 c03f8a40 00000000 c03b0410 cf829f04 cf829ee8 9ee0: cf829f0c cf829ef0 cf829f00 cf829ef8 c03b24c0 c03b0410 cfdf7a3c 0000005c 9f00: cf829f54 cf829f10 c0030398 c03b24cc 00000000 00000006 00000006 c03afe50 9f20: 00000000 c039d11c cf829f54 c03cd334 00000006 c03cd314 c03f8a40 c03b24c0 9f40: c03d29e0 0000005c cf829f94 cf829f58 c03b2c98 c03b2a60 00000006 00000006 9f60: c03b24c0 3e9b7ef7 dfdf4fed 00000000 c030ce98 00000000 00000000 00000000 9f80: 00000000 00000000 cf829fac cf829f98 c030cea8 c03b2bb4 00000000 00000000 9fa0: 00000000 cf829fb0 c000e298 c030cea4 00000000 00000000 00000000 00000000 9fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 9fe0: 00000000 00000000 00000000 00000000 00000013 00000000 5e3decfa e74e5bde Backtrace: [<c0263fec>] (rsnd_gen_dma_addr) from [<c0263348>] (rsnd_dma_init+0x14c/0x24c) r10:0000001f r9:00000000 r8:cf8e5610 r7:cf829d1c r6:cf9b5990 r5:00000001 r4:cf9a3e0c [<c02631fc>] (rsnd_dma_init) from [<c0265d08>] (rsnd_ssi_dma_probe+0x84/0xa4) r10:cf9b5990 r9:00000000 r8:00000000 r7:cf9a7c1c r6:00000005 r5:c03e0fb4 r4:cf8e5600 [<c0265c84>] (rsnd_ssi_dma_probe) from [<c0262850>] (rsnd_probe+0x110/0x258) r6:cf9a7c10 r5:cf9a7c10 r4:cf9a7c18 r3:c03f40e0 [<c0262740>] (rsnd_probe) from [<c018fd88>] (platform_drv_probe+0x20/0x50) r10:c03f3d14 r9:c03d29e0 r8:00000000 r7:00000000 r6:c0416bb8 r5:c03f3d14 r4:cf8e5610 [<c018fd68>] (platform_drv_probe) from [<c018ecb8>] (driver_probe_device+0x118/0x228) r5:c0416bb0 r4:cf8e5610 [<c018eba0>] (driver_probe_device) from [<c018ee5c>] (__driver_attach+0x94/0x98) r10:00000000 r8:c03b24c0 r7:00000000 r6:cf8e5644 r5:c03f3d14 r4:cf8e5610 [<c018edc8>] (__driver_attach) from [<c018d32c>] (bus_for_each_dev+0x5c/0x90) r6:c018edc8 r5:00000000 r4:c03f3d14 r3:cf8e8e3c [<c018d2d0>] (bus_for_each_dev) from [<c018e990>] (driver_attach+0x24/0x28) r6:c03eb348 r5:cf9b5a00 r4:c03f3d14 [<c018e96c>] (driver_attach) from [<c018db54>] (bus_add_driver+0xe0/0x1c8) [<c018da74>] (bus_add_driver) from [<c018f27c>] (driver_register+0x80/0xfc) r7:c03c69dc r6:c03cd314 r5:00000006 r4:c03f3d14 [<c018f1fc>] (driver_register) from [<c0190590>] (__platform_driver_register+0x50/0x64) r5:00000006 r4:c03cd334 [<c0190540>] (__platform_driver_register) from [<c03c69f4>] (rsnd_driver_init+0x18/0x20) [<c03c69dc>] (rsnd_driver_init) from [<c03b2af4>] (do_one_initcall+0xa0/0x154) [<c03b2a54>] (do_one_initcall) from [<c03b2c98>] (kernel_init_freeable+0xf0/0x1b8) r10:0000005c r9:c03d29e0 r8:c03b24c0 r7:c03f8a40 r6:c03cd314 r5:00000006 r4:c03cd334 [<c03b2ba8>] (kernel_init_freeable) from [<c030cea8>] (kernel_init+0x10/0xec) r10:00000000 r9:00000000 r8:00000000 r7:00000000 r6:00000000 r5:c030ce98 r4:00000000 [<c030ce98>] (kernel_init) from [<c000e298>] (ret_from_fork+0x14/0x3c) r4:00000000 r3:00000000 Code: e1a07003 ebfcaf68 e3a01c02 e3a02000 (e5908000) ---[ end trace 421d78826f73f05e ]--- Kernel panic - not syncing: Attempted to kill init! exitcode=0x0000000b ---[ end Kernel panic - not syncing: Attempted to kill init! exitcode=0x0000000b

3 3

[alsa-devel] [PATCH 0/4] ASoC: fsi/rsnd: cyclic transfer support
by Kuninori Morimoto 18 Jun '14

18 Jun '14

Hi Mark Current Renesas sound driver (= FSI/RSND) is not using ALSA common DMAEngine method for some reasons. The 1st reason was that Renesas DMA didn't support cyclic transfer mode. But now, it is supported ! These patches support cyclic transfer for FSI/RSND It needs this branch git://git.infradead.org/users/vkoul/slave-dma.git :: next especially dfbb85cab5f0819d0424a3637b03e7892704fa42 (DMA: shdma: add cyclic transfer support) These are based on mark/topic/rcar branch But, #4 patch needs mark/fix/rcar merge Kuninori Morimoto (4): ASoC: fsi: use SNDRV_DMA_TYPE_DEV for sound buffer ASoC: fsi: add fsi_pointer_update() for common pointer method ASoC: fsi: use dmaengine_prep_dma_cyclic() for DMA transfer ASoC: rsnd: use dmaengine_prep_dma_cyclic() instead of original method arch/arm/mach-shmobile/board-armadillo800eva.c | 4 + arch/arm/mach-shmobile/board-kzm9g.c | 2 + arch/arm/mach-shmobile/board-mackerel.c | 4 + arch/sh/boards/mach-ecovec24/setup.c | 2 + sound/soc/sh/fsi.c | 189 +++++++----------------- sound/soc/sh/rcar/core.c | 76 +++------- sound/soc/sh/rcar/rsnd.h | 4 - 7 files changed, 82 insertions(+), 199 deletions(-) Best regards --- Kuninori Morimoto

2 5

[alsa-devel] [PATCH 0/2] ASoC: rsnd: add BUSIF support for DMA
by Kuninori Morimoto 18 Jun '14

18 Jun '14

Hi Mark These patches are for R-Car sound BUSIF support for DMA transfer. If we use BUSIF via DMA, it will work as FIFO. It can reduce underrun error. Kuninori Morimoto (2): ASoC: rsnd: fixup loop exit timing of dma name search ASoC: rsnd: SSI + DMA can select BUSIF .../devicetree/bindings/sound/renesas,rsnd.txt | 1 + include/sound/rcar_snd.h | 1 + sound/soc/sh/rcar/core.c | 26 +++++--- sound/soc/sh/rcar/gen.c | 64 +++++++++++++------- sound/soc/sh/rcar/rsnd.h | 10 ++- sound/soc/sh/rcar/src.c | 37 ++++++++--- sound/soc/sh/rcar/ssi.c | 33 +++++++++- 7 files changed, 126 insertions(+), 46 deletions(-) Best regards --- Kuninori Morimoto

2 3

[alsa-devel] [PATCH 0/3] Auvitek au0828 fixups
by Mauro Carvalho Chehab 18 Jun '14

18 Jun '14

Testing auvitek au0828-based devices with xawtv caused lots of overruns, as the period_bytes is set to a too low value, with causes troubles when syncing the audio capture from au0828 card with the audio playback, at the main audio card. It turns that we need a period_bytes size that it is big enough to receive data from two consecutive URB intervals. While here, also add the missing USB IDs from other au0828-based devices. Mauro Carvalho Chehab (3): sound: Add a quirk to enforce period_bytes sound: simplify au0828 quirk table sound: Update au0828 quirks table sound/usb/card.h | 1 + sound/usb/pcm.c | 34 ++++++++++ sound/usb/quirks-table.h | 167 ++++++++++++----------------------------------- sound/usb/quirks.c | 14 ++-- sound/usb/stream.c | 1 + sound/usb/usbaudio.h | 3 +- 6 files changed, 88 insertions(+), 132 deletions(-) -- 1.9.3

4 15

[alsa-devel] Trying to enable digital audio output - Intel HDA - AD1983
by Scott D'Vileskis 18 Jun '14

18 Jun '14

Hi-- I have a Lenovo M55 SFF desktop I use for a media center computer (XBMCbuntu). It has the AD1983 chip on it. From the factory, it had 2 ports in the back (mic + speaker) , and 2 ports in the front (mic + speaker). It also has an on-board speaker. It never shipped with a SPDIF output, but I soldered a wire to the AD1983 and installed an RCA jack on the back of the case. Earlier versions of XBMC shipped with a different version of alsa, and I was able to use this custom SPDIF output without issue for the last year or two, but the current version only enables the 'proper' inputs/outputs I have tried editing the module options and specifying several different models, but the documentation doesn't list any 'model' options for the AD1983. I suspect I need to overwrite the pin configs. cat init_pin_configs: 0x05 0x01014010 0x06 0x0221401f 0x07 0x991301f0 0x08 0x02a19020 0x09 0x0181302e 0x0a 0x414511f0 [ 2421.907510] hda-codec: reconfiguring [ 2421.980324] autoconfig: line_outs=1 (0x5/0x0/0x0/0x0/0x0) type:line [ 2421.980328] speaker_outs=1 (0x7/0x0/0x0/0x0/0x0) [ 2421.980330] hp_outs=1 (0x6/0x0/0x0/0x0/0x0) [ 2421.980332] mono: mono_out=0x0 [ 2421.980333] inputs: [ 2421.980335] Mic=0x8 [ 2421.980336] Line=0x9 [ 2421.981858] input: HDA Intel Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input34 [ 2421.984829] input: HDA Intel Line Out as /devices/pci0000:00/0000:00:1b.0/sound/card0/input33 [ 2421.985848] input: HDA Intel Line as /devices/pci0000:00/0000:00:1b.0/sound/card0/input32 Does anyone have any guidance for how I may re-enable the SPDIF on the AD1983 when my BIOS disables it by default? Where else should I look for information on pin configs? Thanks!

2 2

[alsa-devel] [PATCH 1/3] ASoC: Add SND_SOC_BYTES_EXT control type
by Charles Keepax 17 Jun '14

17 Jun '14

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7..92c1b99 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -256,6 +256,13 @@ ((unsigned long)&(struct soc_bytes) \ {.base = xbase, .num_regs = xregs }) } +#define SND_SOC_BYTES_EXT(xname, xbase, xregs, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_bytes_info, .get = xhandler_get, \ + .put = xhandler_put, .private_value = \ + ((unsigned long)&(struct soc_bytes) \ + {.base = xbase, .num_regs = xregs }) } + #define SND_SOC_BYTES_MASK(xname, xbase, xregs, xmask) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \ -- 1.7.2.5

3 6

[alsa-devel] alsa-lib: pcm_plug: fix float conversion for user specified ttable
by Sergey 17 Jun '14

17 Jun '14

Move custom ttable with equal channels case from a separate ttable_last exception into a common plugins insertion loop. Fixes plug with ttable for float pcms (jack, ladspa). Example: aplay -fFLOAT_LE /dev/zero pcm.!default { type plug slave.pcm { type null } ttable.0.0 1 } --- src/pcm/pcm_plug.c | 36 +++++++++++------------------------- 1 file changed, 11 insertions(+), 25 deletions(-) diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index ede9c15..6d0454f 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -53,7 +53,7 @@ typedef struct { const snd_config_t *rate_converter; enum snd_pcm_plug_route_policy route_policy; snd_pcm_route_ttable_entry_t *ttable; - int ttable_ok, ttable_last; + int ttable_ok; unsigned int tt_ssize, tt_cused, tt_sused; } snd_pcm_plug_t; @@ -380,7 +380,7 @@ static int snd_pcm_plug_change_channels(snd_pcm_t *pcm, snd_pcm_t **new, snd_pcm snd_pcm_route_ttable_entry_t *ttable; int err; if (clt->channels == slv->channels && - (!plug->ttable || !plug->ttable_last)) + (!plug->ttable || plug->ttable_ok)) return 0; if (clt->rate != slv->rate && clt->channels > slv->channels) @@ -485,13 +485,15 @@ static int snd_pcm_plug_change_format(snd_pcm_t *pcm, snd_pcm_t **new, snd_pcm_p /* No conversion is needed */ if (clt->format == slv->format && clt->rate == slv->rate && - clt->channels == slv->channels) + clt->channels == slv->channels && + (!plug->ttable || plug->ttable_ok)) return 0; if (snd_pcm_format_linear(slv->format)) { /* Conversion is done in another plugin */ if (clt->rate != slv->rate || - clt->channels != slv->channels) + clt->channels != slv->channels || + (plug->ttable && !plug->ttable_ok)) return 0; cfmt = clt->format; switch (clt->format) { @@ -525,7 +527,8 @@ static int snd_pcm_plug_change_format(snd_pcm_t *pcm, snd_pcm_t **new, snd_pcm_p if (snd_pcm_format_linear(clt->format)) { cfmt = clt->format; f = snd_pcm_lfloat_open; - } else if (clt->rate != slv->rate || clt->channels != slv->channels) { + } else if (clt->rate != slv->rate || clt->channels != slv->channels || + (plug->ttable && !plug->ttable_ok)) { cfmt = SND_PCM_FORMAT_S16; f = snd_pcm_lfloat_open; } else @@ -641,11 +644,12 @@ static int snd_pcm_plug_insert_plugins(snd_pcm_t *pcm, }; snd_pcm_plug_params_t p = *slave; unsigned int k = 0; - plug->ttable_ok = plug->ttable_last = 0; + plug->ttable_ok = 0; while (client->format != p.format || client->channels != p.channels || client->rate != p.rate || - client->access != p.access) { + client->access != p.access || + (plug->ttable && !plug->ttable_ok)) { snd_pcm_t *new; int err; if (k >= sizeof(funcs)/sizeof(*funcs)) @@ -660,24 +664,6 @@ static int snd_pcm_plug_insert_plugins(snd_pcm_t *pcm, } k++; } -#ifdef BUILD_PCM_PLUGIN_ROUTE - /* it's exception, user specified ttable, but no reduction/expand */ - if (plug->ttable && !plug->ttable_ok) { - snd_pcm_t *new; - int err; - plug->ttable_last = 1; - err = snd_pcm_plug_change_channels(pcm, &new, client, &p); - if (err < 0) { - snd_pcm_plug_clear(pcm); - return err; - } - assert(err); - assert(plug->ttable_ok); - plug->gen.slave = new; - pcm->fast_ops = new->fast_ops; - pcm->fast_op_arg = new->fast_op_arg; - } -#endif return 0; } -- 1.7.10.4

3 3

[alsa-devel] [PATCH v3 0/3] Add ES8328 audio codec
by Sean Cross 17 Jun '14

17 Jun '14

This patchset adds support for the Everest Semi ES8328 audio codec. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the codec itself and speaker amps, have headphone support, and must supply an unusual frequency to the codec chip. Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver *** BLURB HERE *** Sean Cross (2): ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 18 + .../devicetree/bindings/sound/imx-audio-es8328.txt | 60 ++ sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 55 ++ sound/soc/codecs/es8328-spi.c | 44 ++ sound/soc/codecs/es8328.c | 625 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 313 +++++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 359 ++++++++++++ 11 files changed, 1509 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

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[alsa-devel] ASoC updates for v3.16
by Mark Brown 17 Jun '14

17 Jun '14

The following changes since commit 7171511eaec5bf23fb06078f59784a3a0626b38f: Linux 3.16-rc1 (2014-06-15 17:45:28 -1000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.16-rc1 for you to fetch changes up to 28e48f0e262fd94a613613d06ac5eaf6745b18a8: Merge remote-tracking branches 'asoc/fix/fsl-dma', 'asoc/fix/fsl-spdif', 'asoc/fix/pxa', 'asoc/fix/rcar' and 'asoc/fix/sigmadsp' into asoc-linus (2014-06-16 16:05:16 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.16 Quite a few build coverage fixes in here among the usual small driver fixes includling the sigmadsp change from Lars - moving the driver to separate modules per bus (which is basically just code motion) avoids issues with some combinations of buses being enabled. ---------------------------------------------------------------- Anssi Hannula (1): ASoC: fsl_spdif: Fix integer overflow when calculating divisors Arnd Bergmann (2): ASoC: pxa: add I2C dependencies as needed ASoC: MMP audio needs sram support Guenter Roeck (1): ASoC: fsl: Fix build problem Jarkko Nikula (1): ASoC: dapm: Make sure register value is in sync with DAPM kcontrol state Kuninori Morimoto (1): ASoC: rsnd: fixup index of src/dst mod when capture Lars-Peter Clausen (1): ASoC: sigmadsp: Split regmap and I2C support into separate modules Mark Brown (2): Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branches 'asoc/fix/fsl-dma', 'asoc/fix/fsl-spdif', 'asoc/fix/pxa', 'asoc/fix/rcar' and 'asoc/fix/sigmadsp' into asoc-linus Nicolin Chen (1): ASoC: fsl_spdif: Fix incorrect usage of regmap_read() sound/soc/codecs/Kconfig | 12 +++++-- sound/soc/codecs/Makefile | 4 +++ sound/soc/codecs/sigmadsp-i2c.c | 35 ++++++++++++++++++++ sound/soc/codecs/sigmadsp-regmap.c | 36 +++++++++++++++++++++ sound/soc/codecs/sigmadsp.c | 65 ++------------------------------------ sound/soc/codecs/sigmadsp.h | 20 ++++++++++++ sound/soc/fsl/fsl_dma.c | 4 +-- sound/soc/fsl/fsl_spdif.c | 6 ++-- sound/soc/pxa/Kconfig | 11 ++++--- sound/soc/sh/rcar/core.c | 2 +- sound/soc/soc-dapm.c | 29 +++++++++-------- 11 files changed, 135 insertions(+), 89 deletions(-) create mode 100644 sound/soc/codecs/sigmadsp-i2c.c create mode 100644 sound/soc/codecs/sigmadsp-regmap.c

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[alsa-devel] TLV320AIC32x4 driver configuration for playing wav file using aplay
by Rahul Panwar 17 Jun '14

17 Jun '14

Hi, We integrated TLV320AIC32x4 codec to OMAP-L138 Soc board. We are using Linux kernel 3.3.0 with driver of TLV320AIC32x4. Driver is successfully detecting. But we are not able to play a wav file using aplay. When we are trying to play any WAV file using aplay, it is not enabling the DAC and file failed to play. When we try to debug that using debugfs, it look like, it is only setting "LOR" ==> "LOR Powe" ==> "LOR Output Mixer". But after that there is no further Input / path / output configuration found. As per the driver it should enable the "Right DAC" via "R_DAC Switch". When it will enable "Right DAC", DAC will be enabled and it may start playing WAV file after that. We are not able the get the cause, what we are doing wrong. What should we need to do to enable DAC. I am attaching the debugfs log for TLV320AIC32x4 dapm, it may be helpful to understand the cause, which we are unable to get. -- Thanks and Regards, Rahul Panwar www.linuxexplore.com

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[alsa-devel] [PATCH] ASoC: wm8994: Prevent double lock of accdet_lock mutex on wm1811
by Charles Keepax 17 Jun '14

17 Jun '14

wm1811_micd_stop takes the accdet_lock mutex, and is called from two places, one of which is already holding the accdet_lock. This obviously causes a lock up. This patch fixes this issue by removing the lock from wm1811_micd_stop and ensuring that it is always locked externally. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247b390..9719d3c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3505,6 +3505,7 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) return IRQ_HANDLED; } +/* Should be called with accdet_lock held */ static void wm1811_micd_stop(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -3512,14 +3513,10 @@ static void wm1811_micd_stop(struct snd_soc_codec *codec) if (!wm8994->jackdet) return; - mutex_lock(&wm8994->accdet_lock); - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_JACK); - mutex_unlock(&wm8994->accdet_lock); - if (wm8994->wm8994->pdata.jd_ext_cap) snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); @@ -3560,10 +3557,10 @@ static void wm8958_open_circuit_work(struct work_struct *work) open_circuit_work.work); struct device *dev = wm8994->wm8994->dev; - wm1811_micd_stop(wm8994->hubs.codec); - mutex_lock(&wm8994->accdet_lock); + wm1811_micd_stop(wm8994->hubs.codec); + dev_dbg(dev, "Reporting open circuit\n"); wm8994->jack_mic = false; -- 1.7.9.5

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[alsa-devel] [PATCH v2] ASoC: fsl-ssi: fix do_div build warning in fsl_ssi_set_bclk()
by Timur Tabi 17 Jun '14

17 Jun '14

do_div() requires that the first parameter is a 64-bit integer, which but clkrate was defined as an unsigned long. This caused the following warnings: CC sound/soc/fsl/fsl_ssi.o sound/soc/fsl/fsl_ssi.c: In function 'fsl_ssi_set_bclk': sound/soc/fsl/fsl_ssi.c:593:3: warning: comparison of distinct pointer types lacks a cast sound/soc/fsl/fsl_ssi.c:593:3: warning: right shift count >= width of type sound/soc/fsl/fsl_ssi.c:593:3: warning: passing argument 1 of '__div64_32' from incompatible pointer type include/asm-generic/div64.h:35:17: note: expected 'uint64_t *' but argument is of type 'long unsigned int *' Signed-off-by: Timur Tabi <timur(a)tabi.org> --- sound/soc/fsl/fsl_ssi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9bfef55..3043d57 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -590,8 +590,8 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, else clkrate = clk_round_rate(ssi_private->baudclk, tmprate); - do_div(clkrate, factor); - afreq = (u32)clkrate / (i + 1); + clkrate /= factor; + afreq = clkrate / (i + 1); if (freq == afreq) sub = 0; -- 1.7.11.7

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[alsa-devel] [PATCH] ASoC: wm8985: Remove unused pointer in wm8985_remove()
by Christian Engelmayer 17 Jun '14

17 Jun '14

Commit a0b148b4 (ASoC: wm8985: Use devm_regulator_bulk_get()) removed the last user of pointer wm8985 to struct wm8985_priv. Thus remove it. Detected by Coverity CID 1222150. Signed-off-by: Christian Engelmayer <cengelma(a)gmx.at> --- Compile tested. Applies against branch for-next in tree git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git --- sound/soc/codecs/wm8985.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 0f5780c..cbbdfbb 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -980,9 +980,6 @@ static int wm8985_resume(struct snd_soc_codec *codec) static int wm8985_remove(struct snd_soc_codec *codec) { - struct wm8985_priv *wm8985; - - wm8985 = snd_soc_codec_get_drvdata(codec); wm8985_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: spdif_transmitter: Allow 192kHz sample rate
by Anssi Hannula 17 Jun '14

17 Jun '14

Transmitters and receivers may support a 192kHz sample rate. Tested with a Cubox-i imx6 system and an Onkyo TX-SR607 receiver. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- sound/soc/codecs/spdif_transmitter.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c index a078aa31052a..e0df537dd4b7 100644 --- a/sound/soc/codecs/spdif_transmitter.c +++ b/sound/soc/codecs/spdif_transmitter.c @@ -24,7 +24,7 @@ #define DRV_NAME "spdif-dit" -#define STUB_RATES SNDRV_PCM_RATE_8000_96000 +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) -- 1.8.4.5

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[alsa-devel] [PATCH] ASoC: MMP audio needs sram support
by Arnd Bergmann 17 Jun '14

17 Jun '14

>From e7a94bb7fb871c73cc85712d89c1f48d0271c1be Mon Sep 17 00:00:00 2001 From: Arnd Bergmann <arnd(a)arndb.de> Date: Thu, 5 Jun 2014 12:31:28 +0200 Subject: [PATCH] ASoC: MMP audio needs sram support Building the pxa/mmp audio driver without support for the mmp sram driver enabled results in this link error: sound/built-in.o: In function `mmp_pcm_free_dma_buffers': :(.text+0x3e734): undefined reference to `sram_get_gpool' sound/built-in.o: In function `mmp_pcm_new': :(.text+0x3e7c0): undefined reference to `sram_get_gpool' The sram driver is cannot be manually enabled and needs to be turned on by selecting MMP_SRAM from each module that needs it, which is what this patch does. Ideally, MMP should move over to the generic SRAM support, but for the moment, we can avoid the build error. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Eric Miao <eric.y.miao(a)gmail.com> Cc: Haojian Zhuang <haojian.zhuang(a)gmail.com> Cc: Qiao Zhou <zhouqiao(a)marvell.com> diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 42b305e..12a6da8 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -11,6 +11,7 @@ config SND_PXA2XX_SOC config SND_MMP_SOC bool "Soc Audio for Marvell MMP chips" depends on ARCH_MMP + select MMP_SRAM select SND_SOC_GENERIC_DMAENGINE_PCM select SND_ARM help

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[alsa-devel] Sound issue in IMx6
by Manish Badarkhe 17 Jun '14

17 Jun '14

Hi All, In our android project, We are using IMx6 chip which is having SSI module to send data to codec. We are facing issues with PCM data which is getting passed to codec getting changed sometimes (even though android is filling up same data). Our use case is as follows: Press key on keypad (PCM open function gets called) and after 3secs of inactivity PCM close function get called. After doing this above activity,quite number of times we have seen sometimes sound gets changed abruptly. I have seen your commit in mainline which is fix to almost similar behavior as I explained above in usecase. Commit: ASoC: fsl: imx-ssi: omit ssi counter to avoid harm in unbalanced situation But still after applying above change issue gets seen sometimes. After this I put some delay in "imx_ssi_trigger" function as below: @@ -360,9 +360,9 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, scr |= SSI_SCR_RE; sier |= sier_bits; - if (++ssi->enabled == 1) - scr |= SSI_SCR_SSIEN; - + scr |= SSI_SCR_SSIEN; + mdelay(1); break; case SNDRV_PCM_TRIGGER_STOP: @@ -374,9 +374,10 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, scr &= ~SSI_SCR_RE; sier &= ~sier_bits; - if (--ssi->enabled == 0) + if (!(scr & (SSI_SCR_TE | SSI_SCR_RE))) scr &= ~SSI_SCR_SSIEN; - + mdelay(1); break; and with this change I never seen issue. But looking like delay to put is just a workaround. Even in datasheet it is not mentioned to put any delay after SSI enabled/disabled. Please let me know if anybody have any proper solution/hints for this issue. Also, let me know if any more information requires. We are using 3.0.35 kernel. Regards Manish Badarkhe

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[alsa-devel] [PATCH v2 0/3] Add ES8328 audio codec
by Sean Cross 17 Jun '14

17 Jun '14

This patchset adds support for the Everest Semi ES8328 audio codec. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the codec itself and speaker amps, have headphone support, and must supply an unusual frequency to the codec chip. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 18 + .../devicetree/bindings/sound/imx-audio-es8328.txt | 60 ++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8328.c | 711 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 305 +++++++++ sound/soc/fsl/Kconfig | 13 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 359 +++++++++++ 10 files changed, 1476 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

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[alsa-devel] [PATCH 0/4] arizona: Improvements to codec DVFS control
by Richard Fitzgerald 16 Jun '14

16 Jun '14

This set of changes cleans up the way the drivers handle the control of internal subsystem clocks for codecs with DVFS. This previously relied on the clock control happening as a side-effect of changing the regulator voltage, rather than explicitly, which assumed that you were uusing the built-int regulator. Also DVFS control was being done from the ADSP driver despite that driver supposedly being codec- agnostic. There are other cases on some codecs where DVFS control is needed so this sequence of patches cleans up the way DVFS is implemented and adds the extra case of DVFS boost for higher sample rates on WM8997 and WM5102 Richard Fitzgerald (4): mfd: arizona: Export function to control subsystem DVFS ASoC: wm_adsp: Move DVFS control into codec driver ASoC: arizona: Add DVFS handling for sample rate control regulator: arizona-ldo1: Do not control clocking from regulator drivers/mfd/arizona-core.c | 84 ++++++++++++++++++++++++++++++++++++++ drivers/regulator/arizona-ldo1.c | 5 -- include/linux/mfd/arizona/core.h | 12 +++++ sound/soc/codecs/arizona.c | 22 +++++++++- sound/soc/codecs/wm5102.c | 47 ++++++++++++++++++++- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm_adsp.c | 76 +--------------------------------- sound/soc/codecs/wm_adsp.h | 11 +++-- 8 files changed, 170 insertions(+), 89 deletions(-) -- 1.7.2.5

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[alsa-devel] [PATCH RESEND] ASoC: rt5677: Convert to use rl6231_get_clk_info
by Axel Lin 16 Jun '14

16 Jun '14

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> Tested-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 2 ++ sound/soc/codecs/rt5677.c | 18 ++---------------- 2 files changed, 4 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cbfa1e1..9d88845 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -445,9 +445,11 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5640=y default y if SND_SOC_RT5645=y default y if SND_SOC_RT5651=y + default y if SND_SOC_RT5677=y default m if SND_SOC_RT5640=m default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m + default m if SND_SOC_RT5677=m config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 60fff7f..1ffc689 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -27,6 +27,7 @@ #include <sound/initval.h> #include <sound/tlv.h> +#include "rl6231.h" #include "rt5677.h" #define RT5677_DEVICE_ID 0x6327 @@ -2798,21 +2799,6 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "PDM2R", NULL, "PDM2 R Mux" }, }; -static int get_clk_info(int sclk, int rate) -{ - int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; - - if (sclk <= 0 || rate <= 0) - return -EINVAL; - - rate = rate << 8; - for (i = 0; i < ARRAY_SIZE(pd); i++) - if (sclk == rate * pd[i]) - return i; - - return -EINVAL; -} - static int rt5677_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -2822,7 +2808,7 @@ static int rt5677_hw_params(struct snd_pcm_substream *substream, int pre_div, bclk_ms, frame_size; rt5677->lrck[dai->id] = params_rate(params); - pre_div = get_clk_info(rt5677->sysclk, rt5677->lrck[dai->id]); + pre_div = rl6231_get_clk_info(rt5677->sysclk, rt5677->lrck[dai->id]); if (pre_div < 0) { dev_err(codec->dev, "Unsupported clock setting\n"); return -EINVAL; -- 1.8.3.2

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[alsa-devel] [PATCH -next 00/26] treewide: Use dma_zalloc_coherent
by Joe Perches 16 Jun '14

16 Jun '14

Use the zeroing function instead of dma_alloc_coherent & memset(,0,) Joe Perches (26): powerpc: Use dma_zalloc_coherent sh: Use dma_zalloc_coherent ata: Use dma_zalloc_coherent block: Use dma_zalloc_coherent crypto: Use dma_zalloc_coherent dma: Use dma_zalloc_coherent gpu: Use dma_zalloc_coherent infiniband: Use dma_zalloc_coherent mmc: Use dma_zalloc_coherent broadcom: Use dma_zalloc_coherent hisilicon: Use dma_zalloc_coherent intel: Use dma_zalloc_coherent ath: Use dma_zalloc_coherent rt2x00: Use dma_zalloc_coherent bfa: Use dma_zalloc_coherent bnx2fc: Use dma_zalloc_coherent bnx2i: Use dma_zalloc_coherent dpt_i2o: Use dma_zalloc_coherent lpfc: Use dma_zalloc_coherent megaraid: Use dma_zalloc_coherent mvsas: Use dma_zalloc_coherent qla2xxx: Use dma_zalloc_coherent qla4xxx: Use dma_zalloc_coherent usb: Use dma_zalloc_coherent fbdev: Use dma_zalloc_coherent sound: Use dma_zalloc_coherent arch/powerpc/platforms/pasemi/dma_lib.c | 8 ++-- arch/powerpc/sysdev/fsl_rmu.c | 9 ++-- arch/sh/mm/consistent.c | 4 +- drivers/ata/sata_fsl.c | 5 +-- drivers/block/nvme-core.c | 5 +-- drivers/crypto/amcc/crypto4xx_core.c | 8 ++-- drivers/crypto/ixp4xx_crypto.c | 8 ++-- drivers/dma/imx-sdma.c | 5 +-- drivers/dma/mxs-dma.c | 8 ++-- drivers/gpu/drm/drm_pci.c | 6 +-- drivers/infiniband/hw/cxgb3/cxio_hal.c | 7 ++-- drivers/infiniband/hw/mthca/mthca_memfree.c | 5 +-- drivers/infiniband/hw/ocrdma/ocrdma_hw.c | 20 ++++----- drivers/infiniband/hw/ocrdma/ocrdma_stats.c | 6 +-- drivers/infiniband/hw/ocrdma/ocrdma_verbs.c | 8 ++-- drivers/mmc/host/msm_sdcc.c | 8 ++-- drivers/net/ethernet/broadcom/bcm63xx_enet.c | 6 +-- drivers/net/ethernet/hisilicon/hix5hd2_gmac.c | 5 +-- drivers/net/ethernet/intel/ixgb/ixgb_main.c | 6 +-- drivers/net/wireless/ath/ath10k/pci.c | 8 +--- drivers/net/wireless/ath/ath10k/wmi.c | 8 +--- drivers/net/wireless/ath/wcn36xx/dxe.c | 6 +-- drivers/net/wireless/rt2x00/rt2x00mmio.c | 8 ++-- drivers/scsi/bfa/bfad_bsg.c | 6 +-- drivers/scsi/bnx2fc/bnx2fc_hwi.c | 59 ++++++++++++--------------- drivers/scsi/bnx2fc/bnx2fc_tgt.c | 51 ++++++++++------------- drivers/scsi/bnx2i/bnx2i_hwi.c | 14 +++---- drivers/scsi/dpt_i2o.c | 19 ++++----- drivers/scsi/lpfc/lpfc_bsg.c | 5 +-- drivers/scsi/lpfc/lpfc_init.c | 22 ++++------ drivers/scsi/lpfc/lpfc_mbox.c | 6 +-- drivers/scsi/lpfc/lpfc_sli.c | 14 +++---- drivers/scsi/megaraid/megaraid_sas_fusion.c | 9 ++-- drivers/scsi/mvsas/mv_init.c | 26 +++++------- drivers/scsi/qla2xxx/qla_init.c | 10 ++--- drivers/scsi/qla4xxx/ql4_init.c | 5 +-- drivers/scsi/qla4xxx/ql4_mbx.c | 21 ++++------ drivers/scsi/qla4xxx/ql4_nx.c | 5 +-- drivers/scsi/qla4xxx/ql4_os.c | 12 +++--- drivers/usb/dwc2/hcd_ddma.c | 20 ++++----- drivers/usb/host/uhci-hcd.c | 7 ++-- drivers/video/fbdev/da8xx-fb.c | 9 ++-- sound/aoa/soundbus/i2sbus/core.c | 12 ++---- sound/sparc/dbri.c | 6 +-- 44 files changed, 197 insertions(+), 308 deletions(-) -- 1.8.1.2.459.gbcd45b4.dirty

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[alsa-devel] Fake Guru Hamid Arabnia and bogus WORLDCOMP and CSCI
by jermy23＠hushmail.com 14 Jun '14

14 Jun '14

Fake Guru Hamid Arabnia and bogus WORLDCOMP and CSCI If you have any thought of attending the world’s biggest fake or bogus conference in computer science, WORLDCOMP http://www.world-academy-of-science.org you must visit any websites below https://sites.google.com/site/worlddump1 or https://sites.google.com/site/dumpconf https://sites.google.com/site/moneycomp1 https://sites.google.com/site/worlddump4 The organizer of this conference is Hamid Arabnia http://www.cs.uga.edu/~hra a professor from University of Georgia, USA. The primary goal of this conference is to collect registration fee by accepting all submitted papers. Several fake papers were published in the conference proceedings. He already earned millions of dollars from the registration fee and is using part of the money for anti-Christmas greetings campaign. http://worldcomp-fake-bogus.blogspot.com has additional information on this campaign. He recently started another new conference CSCI due to his hunger for money http://www.americancse.org Hamid Arabnia failed to reveal the reviews and reviewers' information for all the papers he received, despite repeated requests and challenges. The reason for his failure is there were no reviews and reviewers and he just cheated the research community for more than a decade by announcing that each draft paper is reviewed by two experts. We challenge him to publish these details at the conference website. Where are these experts? Where are these reviews? He pays tens of thousands of dollars and brings keynote speakers and project his conference as a great one. He lists the names of current and former keynote speakers on the front page of the conference and use those names as baits to attract paper submissions from innocent researchers. He always conducts the conferences in the “sin city” to attract people who prefer “other” activities. DBLP stopped indexing these conferences since 2011 and displayed an explicit message; The DBLP Advisory Board decided to discontinue indexing of this conference series. Visit http://www.informatik.uni-trier.de/~ley/db/conf/biocomp/index.html as a sample. He was forced to remove his name, the university of Georgia name, and university of Georgia email address from the conference’s contact page because the University has banned him from doing that. He is running the conference behind the scenes and there is no coordinator or general chair listed for this conference. Almost every week (until the conference date is so close), Hamid Arabnia updates the conference website with things like this: You are invited to submit a "Late Breaking Paper", "Position Paper", "Abstract/Poster Paper" or “Abstract of Thesis or Dissertation” for consideration, etc. He wants more papers because each paper generates around 500 US dollars in to his pocket. Additional information on this conference bogus conference is available by searching internet using the keywords worldcomp fake or Hamid Arabnia fake Apologies for posting to multiple mailing lists. Spreading the word is the only way to stop this conference from cheating innocent researchers.

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[alsa-devel] [RFC PATCH] alsactl: Do not run udev rule before datadir is mounted
by David Henningsson 14 Jun '14

14 Jun '14

The 90-alsa-restore.rules udev rule is primarily meant to handle hotplugging, but sometimes it is also run before /usr is mounted, and alsactl restore depends on /usr/share/alsa being present. If /usr/share/alsa is not present, we're so early in the boot process that alsa-restore.service (or upstart equivalent) will run later on. BugLink: https://bugs.launchpad.net/bugs/1289730 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- I'm not really sure of all quirks in the boot process, and I haven't tested the patch either, hence the RFC above. Do you think it makes sense? Should we also add @sbindir@ to the things to test for? alsactl/90-alsa-restore.rules.in | 2 +- alsactl/Makefile.am | 1 + configure.ac | 1 + 3 files changed, 3 insertions(+), 1 deletion(-) diff --git a/alsactl/90-alsa-restore.rules.in b/alsactl/90-alsa-restore.rules.in index c68119d..c83c6b5 100644 --- a/alsactl/90-alsa-restore.rules.in +++ b/alsactl/90-alsa-restore.rules.in @@ -1,4 +1,4 @@ -ACTION=="add", SUBSYSTEM=="sound", KERNEL=="controlC*", KERNELS!="card*", GOTO="alsa_restore_go" +ACTION=="add", SUBSYSTEM=="sound", KERNEL=="controlC*", KERNELS!="card*", TEST=="@datadir@", GOTO="alsa_restore_go" GOTO="alsa_restore_end" LABEL="alsa_restore_go" diff --git a/alsactl/Makefile.am b/alsactl/Makefile.am index b862412..afbedb3 100644 --- a/alsactl/Makefile.am +++ b/alsactl/Makefile.am @@ -44,6 +44,7 @@ endif edit = \ $(SED) -r -e 's,@sbindir\@,$(sbindir),g' \ + -e 's,@datadir\@,$(mydatadir),g' \ -e 's,@daemonswitch\@,$(ALSACTL_DAEMONSWITCH),g' \ < $< > $@ || rm $@ diff --git a/configure.ac b/configure.ac index 7e6894a..a4d2db3 100644 --- a/configure.ac +++ b/configure.ac @@ -292,6 +292,7 @@ AC_DEFINE_UNQUOTED(SOUNDSDIR, "$soundsdir", [directory containing sample data]) mydatadir="$dir/alsa" AC_DEFINE_UNQUOTED(DATADIR, "$mydatadir", [directory containing alsa configuration]) +AC_SUBST(mydatadir) AC_ARG_WITH(testsound, AS_HELP_STRING([--with-testsound=file], [give the path of test sound file for alsaconf]), -- 1.9.1

5 13

[alsa-devel] [RFC][PATCH 0/2] Fix aplay and alsa-lib write_areas() freezing with 100% cpu usage on stuck slave pcm
by Sergey 13 Jun '14

13 Jun '14

These patches originate from `aplay -vvv /dev/zero` not responding to Ctrl+C when playing to plugged "null", e.g. pcm.!default { type plug slave.pcm { type null } slave.format S32_LE } First patch fixes snd_pcm_write_areas() stuck in a loop because of transfer func() always returning 0. That caused application calling it to freeze. In the example above it have frozen aplay. If it was called by firefox, it would freeze firefox. Is it a good way to fix it? Should snd_pcm_read_areas() be fixed like that too? Second patch fixes aplay itself, not returning from pcm_write() after SIGINT. Additionally aplay may restore original interrupt handler, to make second Ctrl+C stop it for sure. In case of other bugs in alsa-lib or third-party slave pcms it seems better than SIGkilling it manually. Comments and suggestions are welcome. Regardless of these patches' fate null plugin must be fixed on its own. Patches (2): pcm: break out of snd_pcm_write_areas() if write function returned 0 aplay: Escape from pcm_write() when in_aborting state src/pcm/pcm.c | 2 +- aplay/aplay.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 1.7.10.4

2 5

[alsa-devel] [PATCH] drm/i915, HD-audio: Don't continue probing when nomodeset is given
by Takashi Iwai 13 Jun '14

13 Jun '14

When a machine is booted with nomodeset option, i915 driver skips the whole initialization. Meanwhile, HD-audio tries to bind wth i915 just by request_symbol() without knowing that the initialization was skipped, and eventually it hits WARN_ON() in i915_request_power_well() and i915_release_power_well() wrongly but still continues probing, even though it doesn't work at all. In this patch, both functions are changed to return an error in case of uninitialized state instead of WARN_ON(), so that HD-audio driver can give up HDMI controller initialization at the right time. Cc: <stable(a)vger.kernel.org> [3.15] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- The bug actually exists since 3.11, but this commit will be applicable only to recent kernels due to ALSA and i915 code changes, that's why marked with [3.15] for stable. drivers/gpu/drm/i915/intel_pm.c | 14 ++++++++------ include/drm/i915_powerwell.h | 4 ++-- sound/pci/hda/hda_i915.c | 12 ++++++------ sound/pci/hda/hda_i915.h | 4 ++-- sound/pci/hda/hda_intel.c | 7 ++++++- 5 files changed, 24 insertions(+), 17 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_pm.c b/drivers/gpu/drm/i915/intel_pm.c index d93dcf683e8c..0d6bd247009b 100644 --- a/drivers/gpu/drm/i915/intel_pm.c +++ b/drivers/gpu/drm/i915/intel_pm.c @@ -5706,30 +5706,32 @@ void intel_display_power_put(struct drm_i915_private *dev_priv, static struct i915_power_domains *hsw_pwr; /* Display audio driver power well request */ -void i915_request_power_well(void) +int i915_request_power_well(void) { struct drm_i915_private *dev_priv; - if (WARN_ON(!hsw_pwr)) - return; + if (!hsw_pwr) + return -ENODEV; dev_priv = container_of(hsw_pwr, struct drm_i915_private, power_domains); intel_display_power_get(dev_priv, POWER_DOMAIN_AUDIO); + return 0; } EXPORT_SYMBOL_GPL(i915_request_power_well); /* Display audio driver power well release */ -void i915_release_power_well(void) +int i915_release_power_well(void) { struct drm_i915_private *dev_priv; - if (WARN_ON(!hsw_pwr)) - return; + if (!hsw_pwr) + return -ENODEV; dev_priv = container_of(hsw_pwr, struct drm_i915_private, power_domains); intel_display_power_put(dev_priv, POWER_DOMAIN_AUDIO); + return 0; } EXPORT_SYMBOL_GPL(i915_release_power_well); diff --git a/include/drm/i915_powerwell.h b/include/drm/i915_powerwell.h index cfdc884405b7..2baba9996094 100644 --- a/include/drm/i915_powerwell.h +++ b/include/drm/i915_powerwell.h @@ -30,7 +30,7 @@ #define _I915_POWERWELL_H_ /* For use by hda_i915 driver */ -extern void i915_request_power_well(void); -extern void i915_release_power_well(void); +extern int i915_request_power_well(void); +extern int i915_release_power_well(void); #endif /* _I915_POWERWELL_H_ */ diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c index 9d07e4edacdb..e9e8a4a4a9a1 100644 --- a/sound/pci/hda/hda_i915.c +++ b/sound/pci/hda/hda_i915.c @@ -22,20 +22,20 @@ #include <drm/i915_powerwell.h> #include "hda_i915.h" -static void (*get_power)(void); -static void (*put_power)(void); +static int (*get_power)(void); +static int (*put_power)(void); -void hda_display_power(bool enable) +int hda_display_power(bool enable) { if (!get_power || !put_power) - return; + return -ENODEV; pr_debug("HDA display power %s \n", enable ? "Enable" : "Disable"); if (enable) - get_power(); + return get_power(); else - put_power(); + return put_power(); } int hda_i915_init(void) diff --git a/sound/pci/hda/hda_i915.h b/sound/pci/hda/hda_i915.h index 5a63da2c53e5..bfd835f8f1aa 100644 --- a/sound/pci/hda/hda_i915.h +++ b/sound/pci/hda/hda_i915.h @@ -17,11 +17,11 @@ #define __SOUND_HDA_I915_H #ifdef CONFIG_SND_HDA_I915 -void hda_display_power(bool enable); +int hda_display_power(bool enable); int hda_i915_init(void); int hda_i915_exit(void); #else -static inline void hda_display_power(bool enable) {} +static inline int hda_display_power(bool enable) { return 0; } static inline int hda_i915_init(void) { return -ENODEV; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bb65a124e006..23fd6b9aecca 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1656,8 +1656,13 @@ static int azx_probe_continue(struct azx *chip) "Error request power-well from i915\n"); goto out_free; } + err = hda_display_power(true); + if (err < 0) { + dev_err(chip->card->dev, + "Cannot turn on display power on i915\n"); + goto out_free; + } #endif - hda_display_power(true); } err = azx_first_init(chip); -- 1.9.3

2 2

[alsa-devel] [PATCH v3] ASoC: fsl-ssi: fix do_div build warning in fsl_ssi_set_bclk()
by Timur Tabi 13 Jun '14

13 Jun '14

do_div() requires that the first parameter be a 64-bit integer, but clkrate was defined as an unsigned long. Instead, just use a normal division. This change fixes the following warnings: CC sound/soc/fsl/fsl_ssi.o sound/soc/fsl/fsl_ssi.c: In function 'fsl_ssi_set_bclk': sound/soc/fsl/fsl_ssi.c:593:3: warning: comparison of distinct pointer types lacks a cast sound/soc/fsl/fsl_ssi.c:593:3: warning: right shift count >= width of type sound/soc/fsl/fsl_ssi.c:593:3: warning: passing argument 1 of '__div64_32' from incompatible pointer type include/asm-generic/div64.h:35:17: note: expected 'uint64_t *' but argument is of type 'long unsigned int *' Signed-off-by: Timur Tabi <timur(a)tabi.org> --- v3: fix grammatical errors in commit message v2: use normal division instead of do_div Note: I cannot test this code because I do not have a platform that runs the SSI in master mode. sound/soc/fsl/fsl_ssi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9bfef55..3043d57 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -590,8 +590,8 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, else clkrate = clk_round_rate(ssi_private->baudclk, tmprate); - do_div(clkrate, factor); - afreq = (u32)clkrate / (i + 1); + clkrate /= factor; + afreq = clkrate / (i + 1); if (freq == afreq) sub = 0; -- 1.7.11.7

1 0

[alsa-devel] pcm: Add missing callbacks that cannot be NULL
by Alexander E. Patrakov 13 Jun '14

13 Jun '14

Some plugins don't implement some mandatory fast_ops. Thus, these fast_ops get defaulted to NULL, and programs crash when calling the corresponding snd_pcm_* functions. The following callbacks were missing: file: htimestamp multi: rewindable, forwardable null: rewindable, forwardable rate: rewindable, forwardable After this series, all required callbacks are present, but I cannot guarantee that they return meaningful values. All patches in this series are only compile-tested. -- Alexander E. Patrakov

2 9

[alsa-devel] [PATCH] ALSA: hda - Add quirk for external mic on Lifebook U904
by David Henningsson 13 Jun '14

13 Jun '14

According to the bug reporter (Данило Шеган), the external mic starts to work and has proper jack detection if only pin 0x19 is marked properly as an external headset mic. AlsaInfo at https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1328587/+attachme… Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1328587 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9c095fa..89af62f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4246,6 +4246,7 @@ enum { ALC269_FIXUP_HEADSET_MIC, ALC269_FIXUP_QUANTA_MUTE, ALC269_FIXUP_LIFEBOOK, + ALC269_FIXUP_LIFEBOOK_EXTMIC, ALC269_FIXUP_AMIC, ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, @@ -4383,6 +4384,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_QUANTA_MUTE }, + [ALC269_FIXUP_LIFEBOOK_EXTMIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1903c }, /* headset mic, with jack detect */ + { } + }, + }, [ALC269_FIXUP_AMIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -4844,6 +4852,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9099, "Sony VAIO S13", ALC275_FIXUP_SONY_DISABLE_AAMIX), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), + SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - fix a fixup value for codec alc293 in the pin_quirk table
by Hui Wang 13 Jun '14

13 Jun '14

The fixup value for codec alc293 was set to ALC269_FIXUP_DELL1_MIC_NO_PRESENCE by a mistake, if we don't fix it, the Dock mic will be overwriten by the headset mic, this will make the Dock mic can't work. Cc: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9c095fa..06d6063 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5159,7 +5159,7 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1d, 0x40700001}, {0x1e, 0x411111f0}, }, - .value = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + .value = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, }, {} }; -- 1.8.1.2

2 1

[alsa-devel] [RFC][PATCH v1] ASoC: fsl_ssi: Add DAI master mode support for SSI on i.MX series
by Nicolin Chen 13 Jun '14

13 Jun '14

From: Nicolin Chen <b42378(a)freescale.com> This patch adds three main functions for DAI master mode: set_dai_fmt(), set_dai_sysclk() and set_dai_tdm_slot(), and one essential baud clock accordingly. After appending this patch, the fsl_ssi driver on i.MX series has the ability to derive LRCLK and BCLK from baud clock source so as to support some audio Codecs which can only be used in slave mode. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 280 +++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/fsl/fsl_ssi.h | 2 + 2 files changed, 278 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f9f4569..b2ebaf8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -38,6 +38,7 @@ #include <linux/device.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/spinlock.h> #include <linux/of_address.h> #include <linux/of_irq.h> #include <linux/of_platform.h> @@ -139,7 +140,10 @@ struct fsl_ssi_private { bool ssi_on_imx; bool imx_ac97; bool use_dma; + bool baudclk_locked; u8 i2s_mode; + spinlock_t baudclk_lock; + struct clk *baudclk; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -434,13 +438,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + unsigned long flags; /* First, we only do fsl_ssi_setup() when SSI is going to be active. * Second, fsl_ssi_setup was already called by ac97_init earlier if * the driver is in ac97 mode. */ - if (!dai->active && !ssi_private->imx_ac97) + if (!dai->active && !ssi_private->imx_ac97) { fsl_ssi_setup(ssi_private); + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + ssi_private->baudclk_locked = false; + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + } return 0; } @@ -502,6 +511,243 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } /** + * fsl_ssi_set_dai_fmt - configure Digital Audio Interface Format. + */ +static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 strcr = 0, stcr, srcr, scr, mask; + + scr = read_ssi(&ssi->scr) & ~(CCSR_SSI_SCR_SYN | CCSR_SSI_SCR_I2S_MODE_MASK); + scr |= CCSR_SSI_SCR_NET; + + mask = CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR | + CCSR_SSI_STCR_TSCKP | CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TFSL | + CCSR_SSI_STCR_TEFS; + stcr = read_ssi(&ssi->stcr) & ~mask; + srcr = read_ssi(&ssi->srcr) & ~mask; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFM: + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + break; + default: + return -EINVAL; + } + scr |= ssi_private->i2s_mode; + + /* Data on rising edge of bclk, frame low, 1clk before data */ + strcr |= CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* Data on rising edge of bclk, frame high */ + strcr |= CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Data on rising edge of bclk, frame high, 1clk before data */ + strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; + break; + case SND_SOC_DAIFMT_DSP_B: + /* Data on rising edge of bclk, frame high */ + strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0; + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do for both normal cases */ + break; + case SND_SOC_DAIFMT_IB_NF: + /* Invert bit clock */ + strcr ^= CCSR_SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_NB_IF: + /* Invert frame clock */ + strcr ^= CCSR_SSI_STCR_TFSI; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Invert both clocks */ + strcr ^= CCSR_SSI_STCR_TSCKP; + strcr ^= CCSR_SSI_STCR_TFSI; + break; + default: + return -EINVAL; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + strcr |= CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR; + scr |= CCSR_SSI_SCR_SYS_CLK_EN; + break; + case SND_SOC_DAIFMT_CBM_CFM: + scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; + break; + default: + return -EINVAL; + } + + stcr |= strcr; + srcr |= strcr; + + if (ssi_private->cpu_dai_drv.symmetric_rates) { + /* Need to clear RXDIR when using SYNC mode */ + srcr &= ~CCSR_SSI_SRCR_RXDIR; + scr |= CCSR_SSI_SCR_SYN; + } + + write_ssi(stcr, &ssi->stcr); + write_ssi(srcr, &ssi->srcr); + write_ssi(scr, &ssi->scr); + + return 0; +} + +/** + * fsl_ssi_set_dai_sysclk - configure Digital Audio Interface bit clock + * + * Note: This function can be only called when using SSI as DAI master + * + * Quick instruction for parameters: + * freq: Output BCLK frequency = samplerate * 32 (fixed) * channels + * dir: SND_SOC_CLOCK_OUT -> TxBCLK, SND_SOC_CLOCK_IN -> RxBCLK. + */ +static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret; + u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; + unsigned long flags, clkrate, baudrate, tmprate; + u64 sub, savesub = 100000; + + /* Don't apply it to any non-baudclk circumstance */ + if (IS_ERR(ssi_private->baudclk)) + return -EINVAL; + + /* It should be already enough to divide clock by setting pm alone */ + psr = 0; + div2 = 0; + + factor = (div2 + 1) * (7 * psr + 1) * 2; + + for (i = 0; i < 255; i++) { + /* The bclk rate must be smaller than 1/5 sysclk rate */ + if (factor * (i + 1) < 5) + continue; + + tmprate = freq * factor * (i + 2); + clkrate = clk_round_rate(ssi_private->baudclk, tmprate); + + do_div(clkrate, factor); + afreq = (u32)clkrate / (i + 1); + + if (freq == afreq) + sub = 0; + else if (freq / afreq == 1) + sub = freq - afreq; + else if (afreq / freq == 1) + sub = afreq - freq; + else + continue; + + /* Calculate the fraction */ + sub *= 100000; + do_div(sub, freq); + + if (sub < savesub) { + baudrate = tmprate; + savesub = sub; + pm = i; + } + + /* We are lucky */ + if (savesub == 0) + break; + } + + /* No proper pm found if it is still remaining the initial value */ + if (pm == 999) { + dev_err(cpu_dai->dev, "failed to handle the required sysclk\n"); + return -EINVAL; + } + + stccr = CCSR_SSI_SxCCR_PM(pm + 1) | (div2 ? CCSR_SSI_SxCCR_DIV2 : 0) | + (psr ? CCSR_SSI_SxCCR_PSR : 0); + mask = CCSR_SSI_SxCCR_PM_MASK | CCSR_SSI_SxCCR_DIV2 | CCSR_SSI_SxCCR_PSR; + + if (dir == SND_SOC_CLOCK_OUT || synchronous) + write_ssi_mask(&ssi->stccr, mask, stccr); + else + write_ssi_mask(&ssi->srccr, mask, stccr); + + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + if (!ssi_private->baudclk_locked) { + ret = clk_set_rate(ssi_private->baudclk, baudrate); + if (ret) { + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + dev_err(cpu_dai->dev, "failed to set baudclk rate\n"); + return -EINVAL; + } + ssi_private->baudclk_locked = true; + } + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + + return 0; +} + +/** + * fsl_ssi_set_dai_tdm_slot - set TDM slot number + * + * Note: This function can be only called when using SSI as DAI master + */ +static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, + u32 rx_mask, int slots, int slot_width) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 val; + + /* The slot number should be >= 2 if using Network mode or I2S mode */ + val = read_ssi(&ssi->scr) & (CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_NET); + if (val && slots < 2) { + dev_err(cpu_dai->dev, "slot number should be >= 2 in I2S or NET\n"); + return -EINVAL; + } + + write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(slots)); + write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(slots)); + + /* The register SxMSKs needs SSI to provide essential clock due to + * hardware design. So we here temporarily enable SSI to set them. + */ + val = read_ssi(&ssi->scr) & CCSR_SSI_SCR_SSIEN; + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN); + + write_ssi(tx_mask, &ssi->stmsk); + write_ssi(rx_mask, &ssi->srmsk); + + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, val); + + return 0; +} + +/** * fsl_ssi_trigger: start and stop the DMA transfer. * * This function is called by ALSA to start, stop, pause, and resume the DMA @@ -517,6 +763,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; unsigned int sier_bits; + unsigned long flags; /* * Enable only the interrupts and DMA requests @@ -557,8 +804,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); if (!ssi_private->imx_ac97 && (read_ssi(&ssi->scr) & - (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) + (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) { write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + ssi_private->baudclk_locked = false; + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + } break; default: @@ -585,6 +836,9 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, + .set_fmt = fsl_ssi_set_dai_fmt, + .set_sysclk = fsl_ssi_set_dai_sysclk, + .set_tdm_slot = fsl_ssi_set_dai_tdm_slot, .trigger = fsl_ssi_trigger, }; @@ -897,6 +1151,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; + ssi_private->baudclk_locked = false; + spin_lock_init(&ssi_private->baudclk_lock); + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) { u32 dma_events[2]; ssi_private->ssi_on_imx = true; @@ -914,6 +1171,15 @@ static int fsl_ssi_probe(struct platform_device *pdev) goto error_irqmap; } + /* For those SLAVE implementations, we ingore non-baudclk cases + * and, instead, abandon MASTER mode that needs baud clock. + */ + ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); + if (IS_ERR(ssi_private->baudclk)) + dev_warn(&pdev->dev, "could not get baud clock: %d\n", ret); + else + clk_prepare_enable(ssi_private->baudclk); + /* * We have burstsize be "fifo_depth - 2" to match the SSI * watermark setting in fsl_ssi_startup(). @@ -1059,8 +1325,11 @@ error_dev: device_remove_file(&pdev->dev, dev_attr); error_clk: - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { + if (!IS_ERR(ssi_private->baudclk)) + clk_disable_unprepare(ssi_private->baudclk); clk_disable_unprepare(ssi_private->clk); + } error_irqmap: irq_dispose_mapping(ssi_private->irq); @@ -1076,8 +1345,11 @@ static int fsl_ssi_remove(struct platform_device *pdev) platform_device_unregister(ssi_private->pdev); snd_soc_unregister_component(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { + if (!IS_ERR(ssi_private->baudclk)) + clk_disable_unprepare(ssi_private->baudclk); clk_disable_unprepare(ssi_private->clk); + } irq_dispose_mapping(ssi_private->irq); return 0; diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index e6b9a69..e6b6324 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -125,7 +125,9 @@ struct ccsr_ssi { #define CCSR_SSI_SRCR_REFS 0x00000001 /* STCCR and SRCCR */ +#define CCSR_SSI_SxCCR_DIV2_SHIFT 18 #define CCSR_SSI_SxCCR_DIV2 0x00040000 +#define CCSR_SSI_SxCCR_PSR_SHIFT 17 #define CCSR_SSI_SxCCR_PSR 0x00020000 #define CCSR_SSI_SxCCR_WL_SHIFT 13 #define CCSR_SSI_SxCCR_WL_MASK 0x0001E000 -- 1.8.4

4 9

[alsa-devel] [PATCH] ALSA: au1x00: Use resource_size
by Himangi Saraogi 13 Jun '14

13 Jun '14

Use the function resource_size, which reduces the chance of introducing off-by-one errors in calculating the resource size. The semantic patch that makes this change is as follows: // <smpl> @@ struct resource *res; @@ - (res->end - res->start) + 1 + resource_size(res) // </smpl> Signed-off-by: Himangi Saraogi <himangi774(a)gmail.com> Acked-by: Julia Lawall <julia.lawall(a)lip6.fr> --- Not compile tested. sound/mips/au1x00.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index d10ef76..86583c7 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -649,7 +649,7 @@ static int au1000_ac97_probe(struct platform_device *pdev) err = -EBUSY; au1000->ac97_res_port = request_mem_region(r->start, - r->end - r->start + 1, pdev->name); + resource_size(r), pdev->name); if (!au1000->ac97_res_port) { snd_printk(KERN_ERR "ALSA AC97: can't grab AC97 port\n"); goto out; -- 1.9.1

2 1

[alsa-devel] ALSA userspace release time?
by David Henningsson 13 Jun '14

13 Jun '14

Hi, It was around a year since we last had a release of ALSA's userspace component. Is it perhaps time for 1.0.28, or are we waiting for something that would be nice to get in first? We've currently got e g speaker-test improvements/fixes, 2.1 surround support, "alsactl monitor" command, and random fixes here and there. Sounds good enough for a release to me. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

4 6

[alsa-devel] [PATCH 4/4] Sound: tlv320aic31xx: Fixed Coding Style Issues
by Shahina Shaik 13 Jun '14

13 Jun '14

From: Shahina Shaik <sharab.shaik(a)gmail.com> Fixed coding style issues of "Missing Blank line after declaration" Signed-off-by: Shahina Shaik <sharab.shaik(a)gmail.com> --- sound/soc/codecs/tlv320aic31xx.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 2341910..a9c4a8b 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -329,6 +329,7 @@ static int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, unsigned int bits; int counter = count; int ret = regmap_read(aic31xx->regmap, reg, &bits); + while ((bits & mask) != wbits && counter && !ret) { usleep_range(sleep, sleep * 2); ret = regmap_read(aic31xx->regmap, reg, &bits); @@ -435,6 +436,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + switch (event) { case SND_SOC_DAPM_POST_PMU: /* change mic bias voltage to user defined */ -- 1.7.9.5

1 0

[alsa-devel] [PATCH 1/1] ASoC: mc13783: Add missing of_node_put
by Sachin Kamat 13 Jun '14

13 Jun '14

of_get_child_by_name expects of_node_put be called when done. Signed-off-by: Sachin Kamat <sachin.kamat(a)samsung.com> --- sound/soc/codecs/mc13783.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 9965277b595a..388f90a597fa 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -766,11 +766,11 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port); if (ret) - return ret; + goto out; ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port); if (ret) - return ret; + goto out; } dev_set_drvdata(&pdev->dev, priv); @@ -783,6 +783,8 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); +out: + of_node_put(np); return ret; } -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ASoC: fsl-ssi: fix do_div build warning in fsl_ssi_set_bclk()
by Timur Tabi 13 Jun '14

13 Jun '14

do_div() requires that the first parameter is a 64-bit integer, which but clkrate was defined as an unsigned long. This caused the following warnings: CC sound/soc/fsl/fsl_ssi.o sound/soc/fsl/fsl_ssi.c: In function 'fsl_ssi_set_bclk': sound/soc/fsl/fsl_ssi.c:593:3: warning: comparison of distinct pointer types lacks a cast sound/soc/fsl/fsl_ssi.c:593:3: warning: right shift count >= width of type sound/soc/fsl/fsl_ssi.c:593:3: warning: passing argument 1 of '__div64_32' from incompatible pointer type include/asm-generic/div64.h:35:17: note: expected 'uint64_t *' but argument is of type 'long unsigned int *' Signed-off-by: Timur Tabi <timur(a)tabi.org> --- Note: I cannot test this code because I do not have a platform that runs the SSI in master mode. sound/soc/fsl/fsl_ssi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9bfef55..9d98340 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -555,8 +555,8 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct regmap *regs = ssi_private->regs; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret; u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; - unsigned long clkrate, baudrate, tmprate; - u64 sub, savesub = 100000; + unsigned long baudrate, tmprate; + u64 clkrate, sub, savesub = 100000; unsigned int freq; bool baudclk_is_used; -- 1.7.11.7

1 0

[alsa-devel] [PATCH] aplay: Fix VU-meter stdout/stderr mess
by Sergey 13 Jun '14

13 Jun '14

Commit 6bbeb78d forgot to change some of VU-meter output to stderr. This patch makes the output of `aplay -vvv` consistent again. --- aplay/aplay.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 33a8ed5..77e40a3 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -1774,12 +1774,12 @@ static void compute_max_peak(u_char *data, size_t count) fflush(stderr); } else if(verbose==3) { - printf(_("Max peak (%li samples): 0x%08x "), (long)ocount, max_peak[0]); + fprintf(stderr, _("Max peak (%li samples): 0x%08x "), (long)ocount, max_peak[0]); for (val = 0; val < 20; val++) if (val <= perc[0] / 5) - putchar('#'); + putc('#', stderr); else - putchar(' '); + putc(' ', stderr); fprintf(stderr, " %i%%\n", perc[0]); fflush(stderr); } -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/2] [RFC] ASoC: samsung: move s3c24xx over to dmaengine
by Arnd Bergmann 12 Jun '14

12 Jun '14

The s3c24xx sound support keeps giving randconfig build errors, this is a new attempt to solve it by moving the driver over to use dmaengine exclusively. The removal of the legacy DMA support and the addition of the s3c24xx_dma_filter pointer are fairly obvious here. The part I'm not completely sure about is the removal of the s3c2410_dma_ctrl(..., S3C2410_DMAOP_STARTED) and dma_data->ops->started() calls. My understanding is that these are only required for drivers that do not support cyclic transfers, which the new dma engine driver now does, so we can simply remove them. This would also fix at least one bug in the ac97 driver on newer machines, which currently gives us a NULL pointer dereference from trying to call dma_data->ops->started(). Any insights about this, or testing would be very welcome. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Heiko Stuebner <heiko(a)sntech.de> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: Kukjin Kim <kgene.kim(a)samsung.com> Cc: linux-samsung-soc(a)vger.kernel.org --- sound/soc/samsung/Kconfig | 13 -- sound/soc/samsung/Makefile | 4 +- sound/soc/samsung/ac97.c | 16 -- sound/soc/samsung/dma.c | 454 ------------------------------------ sound/soc/samsung/dmaengine.c | 3 + sound/soc/samsung/s3c24xx-i2s.c | 1 - 6 files changed, 4 insertions(+), 487 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 7745629..32bfcf1 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,25 +1,15 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" depends on PLAT_SAMSUNG - select S3C2410_DMA if ARCH_S3C24XX select S3C64XX_PL080 if ARCH_S3C64XX - select SND_S3C_DMA if !ARCH_S3C24XX - select SND_S3C_DMA_LEGACY if ARCH_S3C24XX select SND_SOC_GENERIC_DMAENGINE_PCM if !ARCH_S3C24XX help Say Y or M if you want to add support for codecs attached to the Samsung SoCs' Audio interfaces. You will also need to select the audio interfaces to support below. -config SND_S3C_DMA - tristate - -config SND_S3C_DMA_LEGACY - tristate - config SND_S3C24XX_I2S tristate - select S3C24XX_DMA config SND_S3C_I2SV2_SOC tristate @@ -27,7 +17,6 @@ config SND_S3C_I2SV2_SOC config SND_S3C2412_SOC_I2S tristate select SND_S3C_I2SV2_SOC - select S3C2410_DMA config SND_SAMSUNG_PCM tristate @@ -83,7 +72,6 @@ config SND_SOC_SAMSUNG_SMDK_WM8994 config SND_SOC_SAMSUNG_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_SOC_SAMSUNG && MACH_SMDK2443 - select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_CODEC select SND_SAMSUNG_AC97 @@ -94,7 +82,6 @@ config SND_SOC_SAMSUNG_SMDK2443_WM9710 config SND_SOC_SAMSUNG_LN2440SBC_ALC650 tristate "SoC AC97 Audio support for LN2440SBC - ALC650" depends on SND_SOC_SAMSUNG && ARCH_S3C24XX - select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_CODEC select SND_SAMSUNG_AC97 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 6d0212b..f69f490 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,6 +1,5 @@ # S3c24XX Platform Support snd-soc-s3c-dma-objs := dmaengine.o -snd-soc-s3c-dma-legacy-objs := dma.o snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o @@ -10,8 +9,7 @@ snd-soc-samsung-spdif-objs := spdif.o snd-soc-pcm-objs := pcm.o snd-soc-i2s-objs := i2s.o -obj-$(CONFIG_SND_S3C_DMA) += snd-soc-s3c-dma.o -obj-$(CONFIG_SND_S3C_DMA_LEGACY) += snd-soc-s3c-dma-legacy.o +obj-$(CONFIG_SND_SOC_SAMSUNG) += snd-soc-s3c-dma.o obj-$(CONFIG_SND_S3C24XX_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_SAMSUNG_AC97) += snd-soc-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 68d9303..a674827 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -225,9 +225,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) @@ -253,11 +250,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - if (!dma_data->ops) - dma_data->ops = samsung_dma_get_ops(); - - dma_data->ops->started(dma_data->channel); - return 0; } @@ -265,9 +257,6 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; @@ -287,11 +276,6 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - if (!dma_data->ops) - dma_data->ops = samsung_dma_get_ops(); - - dma_data->ops->started(dma_data->channel); - return 0; } diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c deleted file mode 100644 index d9dc7bc..0000000 --- a/sound/soc/samsung/dma.c +++ /dev/null @@ -1,454 +0,0 @@ -/* - * dma.c -- ALSA Soc Audio Layer - * - * (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * Copyright 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/module.h> - -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include <asm/dma.h> -#include <mach/hardware.h> -#include <mach/dma.h> - -#include "dma.h" - -#define ST_RUNNING (1<<0) -#define ST_OPENED (1<<1) - -static const struct snd_pcm_hardware dma_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, - .buffer_bytes_max = 128*1024, - .period_bytes_min = PAGE_SIZE, - .period_bytes_max = PAGE_SIZE*2, - .periods_min = 2, - .periods_max = 128, - .fifo_size = 32, -}; - -struct runtime_data { - spinlock_t lock; - int state; - unsigned int dma_loaded; - unsigned int dma_period; - dma_addr_t dma_start; - dma_addr_t dma_pos; - dma_addr_t dma_end; - struct s3c_dma_params *params; -}; - -static void audio_buffdone(void *data); - -/* dma_enqueue - * - * place a dma buffer onto the queue for the dma system - * to handle. - */ -static void dma_enqueue(struct snd_pcm_substream *substream) -{ - struct runtime_data *prtd = substream->runtime->private_data; - dma_addr_t pos = prtd->dma_pos; - unsigned int limit; - struct samsung_dma_prep dma_info; - - pr_debug("Entered %s\n", __func__); - - limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; - - pr_debug("%s: loaded %d, limit %d\n", - __func__, prtd->dma_loaded, limit); - - dma_info.cap = (samsung_dma_has_circular() ? DMA_CYCLIC : DMA_SLAVE); - dma_info.direction = - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); - dma_info.fp = audio_buffdone; - dma_info.fp_param = substream; - dma_info.period = prtd->dma_period; - dma_info.len = prtd->dma_period*limit; - - if (dma_info.cap == DMA_CYCLIC) { - dma_info.buf = pos; - prtd->params->ops->prepare(prtd->params->ch, &dma_info); - prtd->dma_loaded += limit; - return; - } - - while (prtd->dma_loaded < limit) { - pr_debug("dma_loaded: %d\n", prtd->dma_loaded); - - if ((pos + dma_info.period) > prtd->dma_end) { - dma_info.period = prtd->dma_end - pos; - pr_debug("%s: corrected dma len %ld\n", - __func__, dma_info.period); - } - - dma_info.buf = pos; - prtd->params->ops->prepare(prtd->params->ch, &dma_info); - - prtd->dma_loaded++; - pos += prtd->dma_period; - if (pos >= prtd->dma_end) - pos = prtd->dma_start; - } - - prtd->dma_pos = pos; -} - -static void audio_buffdone(void *data) -{ - struct snd_pcm_substream *substream = data; - struct runtime_data *prtd = substream->runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - if (prtd->state & ST_RUNNING) { - prtd->dma_pos += prtd->dma_period; - if (prtd->dma_pos >= prtd->dma_end) - prtd->dma_pos = prtd->dma_start; - - if (substream) - snd_pcm_period_elapsed(substream); - - spin_lock(&prtd->lock); - if (!samsung_dma_has_circular()) { - prtd->dma_loaded--; - dma_enqueue(substream); - } - spin_unlock(&prtd->lock); - } -} - -static int dma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - unsigned long totbytes = params_buffer_bytes(params); - struct s3c_dma_params *dma = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - struct samsung_dma_req req; - struct samsung_dma_config config; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; - - /* this may get called several times by oss emulation - * with different params -HW */ - if (prtd->params == NULL) { - /* prepare DMA */ - prtd->params = dma; - - pr_debug("params %p, client %p, channel %d\n", prtd->params, - prtd->params->client, prtd->params->channel); - - prtd->params->ops = samsung_dma_get_ops(); - - req.cap = (samsung_dma_has_circular() ? - DMA_CYCLIC : DMA_SLAVE); - req.client = prtd->params->client; - config.direction = - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); - config.width = prtd->params->dma_size; - config.fifo = prtd->params->dma_addr; - prtd->params->ch = prtd->params->ops->request( - prtd->params->channel, &req, rtd->cpu_dai->dev, - prtd->params->ch_name); - if (!prtd->params->ch) { - pr_err("Failed to allocate DMA channel\n"); - return -ENXIO; - } - prtd->params->ops->config(prtd->params->ch, &config); - } - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - runtime->dma_bytes = totbytes; - - spin_lock_irq(&prtd->lock); - prtd->dma_loaded = 0; - prtd->dma_period = params_period_bytes(params); - prtd->dma_start = runtime->dma_addr; - prtd->dma_pos = prtd->dma_start; - prtd->dma_end = prtd->dma_start + totbytes; - spin_unlock_irq(&prtd->lock); - - return 0; -} - -static int dma_hw_free(struct snd_pcm_substream *substream) -{ - struct runtime_data *prtd = substream->runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - snd_pcm_set_runtime_buffer(substream, NULL); - - if (prtd->params) { - prtd->params->ops->flush(prtd->params->ch); - prtd->params->ops->release(prtd->params->ch, - prtd->params->client); - prtd->params = NULL; - } - - return 0; -} - -static int dma_prepare(struct snd_pcm_substream *substream) -{ - struct runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!prtd->params) - return 0; - - /* flush the DMA channel */ - prtd->params->ops->flush(prtd->params->ch); - - prtd->dma_loaded = 0; - prtd->dma_pos = prtd->dma_start; - - /* enqueue dma buffers */ - dma_enqueue(substream); - - return ret; -} - -static int dma_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->state |= ST_RUNNING; - prtd->params->ops->trigger(prtd->params->ch); - break; - - case SNDRV_PCM_TRIGGER_STOP: - prtd->state &= ~ST_RUNNING; - prtd->params->ops->stop(prtd->params->ch); - break; - - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static snd_pcm_uframes_t -dma_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd = runtime->private_data; - unsigned long res; - - pr_debug("Entered %s\n", __func__); - - res = prtd->dma_pos - prtd->dma_start; - - pr_debug("Pointer offset: %lu\n", res); - - /* we seem to be getting the odd error from the pcm library due - * to out-of-bounds pointers. this is maybe due to the dma engine - * not having loaded the new values for the channel before being - * called... (todo - fix ) - */ - - if (res >= snd_pcm_lib_buffer_bytes(substream)) { - if (res == snd_pcm_lib_buffer_bytes(substream)) - res = 0; - } - - return bytes_to_frames(substream->runtime, res); -} - -static int dma_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_soc_set_runtime_hwparams(substream, &dma_hardware); - - prtd = kzalloc(sizeof(struct runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - - runtime->private_data = prtd; - return 0; -} - -static int dma_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd = runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - if (!prtd) - pr_debug("dma_close called with prtd == NULL\n"); - - kfree(prtd); - - return 0; -} - -static int dma_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - pr_debug("Entered %s\n", __func__); - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops dma_ops = { - .open = dma_open, - .close = dma_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = dma_hw_params, - .hw_free = dma_hw_free, - .prepare = dma_prepare, - .trigger = dma_trigger, - .pointer = dma_pointer, - .mmap = dma_mmap, -}; - -static int preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = dma_hardware.buffer_bytes_max; - - pr_debug("Entered %s\n", __func__); - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - return 0; -} - -static void dma_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - pr_debug("Entered %s\n", __func__); - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static int dma_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - pr_debug("Entered %s\n", __func__); - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } -out: - return ret; -} - -static struct snd_soc_platform_driver samsung_asoc_platform = { - .ops = &dma_ops, - .pcm_new = dma_new, - .pcm_free = dma_free_dma_buffers, -}; - -void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, - struct s3c_dma_params *playback, - struct s3c_dma_params *capture) -{ - snd_soc_dai_init_dma_data(dai, playback, capture); -} -EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); - -int samsung_asoc_dma_platform_register(struct device *dev) -{ - return devm_snd_soc_register_platform(dev, &samsung_asoc_platform); -} -EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); - -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("Samsung ASoC DMA Driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index a0e4e79..506f5bf 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -17,6 +17,7 @@ #include <linux/module.h> #include <linux/amba/pl08x.h> +#include <linux/platform_data/dma-s3c24xx.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,8 @@ #ifdef CONFIG_ARCH_S3C64XX #define filter_fn pl08x_filter_id +#elif defined(CONFIG_ARCH_S3C24XX) +#define filter_fn s3c24xx_dma_filter #else #define filter_fn NULL #endif diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 9aba9fb..890d241 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -290,7 +290,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, else s3c24xx_snd_txctrl(1); - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND:

3 5

[alsa-devel] [PATCH] ALSA: hda - verify pin:converter connection on unsol event for HSW and VLV
by mengdong.lin＠intel.com 12 Jun '14

12 Jun '14

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch will verify the pin's coverter selection for an active stream when an unsol event reports this pin becomes available again after a display mode change or hot-plug event. For Haswell+ and Valleyview: display mode change or hot-plug can change the transcoder:port connection and make all the involved audio pins share the 1st converter. So the stream using 1st convertor will flow to multiple pins but active streams using other converters will fail. This workaround is to assure the pin selects the right conveter and an assigned converter is not shared by other unused pins. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index be0a9ee..3e4417b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1594,10 +1594,18 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) * Re-setup pin and infoframe. This is needed e.g. when * - sink is first plugged-in (infoframe is not set up if !monitor_present) * - transcoder can change during stream playback on Haswell + * and this can make HW reset converter selection on a pin. */ - if (eld->eld_valid && !old_eld_valid && per_pin->setup) + if (eld->eld_valid && !old_eld_valid && per_pin->setup) { + if (is_haswell_plus(codec) || is_valleyview(codec)) { + intel_verify_pin_cvt_connect(codec, per_pin); + intel_not_share_assigned_cvt(codec, pin_nid, + per_pin->mux_idx); + } + hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); + } } if (eld_changed) -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: wm5102: Convert snd_kcontrol_chip to snd_soc_kcontrol_codec
by Charles Keepax 12 Jun '14

12 Jun '14

Controls for shaping the ultrasonic frequency response were introduced in this commit: commit 720630c002ffc7b0fa2ed5b3f4bfb36fd8f87ca6 ASoC: wm5102: Add controls to allow shaping of ultrasonic response However, they mistakenly used snd_kcontrol_chip instead of snd_soc_kcontrol_codec, which has replaced it now the framework is moving to componentisation. This patch fixes this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 8613504..fa24d55 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -615,7 +615,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); uint16_t data; @@ -630,7 +630,7 @@ static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); mutex_lock(&codec->mutex); @@ -645,7 +645,7 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); mutex_lock(&codec->mutex); @@ -658,7 +658,7 @@ static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); mutex_lock(&codec->mutex); -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: fsl: Fix build problem
by Guenter Roeck 12 Jun '14

12 Jun '14

Commit 432481220 (ASoC: fsl-ssi: Use regmap) removed struct ccsr_ssi. Unfortunately, the structure is still used. This causes mpc85xx_smp_defconfig and mpc85xx_defconfig builds to fail with sound/soc/fsl/fsl_dma.c:926:50: error: invalid use of undefined type 'struct ccsr_ssi' dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0); ound/soc/fsl/fsl_dma.c:927:50: error: invalid use of undefined type 'struct ccsr_ssi' dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0); Fix by using constants, similar to original commit. Cc: Markus Pargmann <mpa(a)pengutronix.de> Signed-off-by: Guenter Roeck <linux(a)roeck-us.net> --- sound/soc/fsl/fsl_dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 6bb0ea5..a609aaf 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -923,8 +923,8 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) dma->dai.pcm_free = fsl_dma_free_dma_buffers; /* Store the SSI-specific information that we need */ - dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0); - dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0); + dma->ssi_stx_phys = res.start + CCSR_SSI_STX0; + dma->ssi_srx_phys = res.start + CCSR_SSI_SRX0; iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL); if (iprop) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: fixup index of src/dst mod when capture
by Kuninori Morimoto 12 Jun '14

12 Jun '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Index of dma name should use -1, not +1 when capture case. Thank you Dan. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 2d0f98f..5a4d7e0 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -289,7 +289,7 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma, dst_mod = mod[index]; } else { src_mod = mod[index]; - dst_mod = mod[index + 1]; + dst_mod = mod[index - 1]; } index = 0; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] OSS/DMASOUND: add __init to dmasound_init
by Fabian Frederick 11 Jun '14

11 Jun '14

dmasound_init is only called by __init functions: __init dmasound_q40_init __init amiga_audio_probe __init dmasound_atari_init Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- sound/oss/dmasound/dmasound.h | 2 +- sound/oss/dmasound/dmasound_core.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/oss/dmasound/dmasound.h b/sound/oss/dmasound/dmasound.h index 01019f0..a181ed2 100644 --- a/sound/oss/dmasound/dmasound.h +++ b/sound/oss/dmasound/dmasound.h @@ -86,7 +86,7 @@ static inline int ioctl_return(int __user *addr, int value) * Initialization */ -extern int dmasound_init(void); +extern int __init dmasound_init(void); #ifdef MODULE extern void dmasound_deinit(void); #else diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index f4ee85a..1f0a42c 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -1394,7 +1394,7 @@ static int state_init(void) * This function is called by _one_ chipset-specific driver */ -int dmasound_init(void) +int __init dmasound_init(void) { int res ; #ifdef MODULE -- 1.8.4.5

1 1

Re: [alsa-devel] ASoC: rsnd: care DMA slave channel name for DT
by Dan Carpenter 11 Jun '14

11 Jun '14

Hello Kuninori Morimoto, The patch 199e7688bdf7: "ASoC: rsnd: care DMA slave channel name for DT" from May 22, 2014, leads to the following static checker warning: sound/soc/sh/rcar/core.c:318 rsnd_dma_of_name() error: buffer overflow 'mod' 4 <= 4 sound/soc/sh/rcar/core.c 278 struct rsnd_mod *mod[MOD_MAX]; 279 struct rsnd_mod *src_mod, *dst_mod; 280 int i, index; 281 282 283 for (i = 0; i < MOD_MAX; i++) 284 mod[i] = NULL; 285 286 /* 287 * in play case... 288 * 289 * src -> dst 290 * 291 * mem -> SSI 292 * mem -> SRC -> SSI 293 * mem -> SRC -> DVC -> SSI I don't understand what this comment means. 294 */ 295 mod[0] = NULL; /* for "mem" */ We memset everything to NULL in the loop above so this isn't needed. 296 index = 1; 297 for (i = 1; i < MOD_MAX; i++) { 298 if (!src) { 299 mod[i] = ssi; 300 break; 301 } else if (!dvc) { 302 mod[i] = src; 303 src = NULL; 304 } else { 305 mod[i] = dvc; 306 dvc = NULL; 307 } 308 309 if (mod[i] == this) 310 index = i; In theory, index can be MOD_MAX - 1. 311 } 312 313 if (is_play) { 314 src_mod = mod[index - 1]; 315 dst_mod = mod[index]; 316 } else { 317 src_mod = mod[index]; 318 dst_mod = mod[index + 1]; So then it complains that mod[] has only MOD_MAX elements so we're one space past the end of the array. Probably the way this is called, there is something to prevent it? 319 } regards, dan carpenter

2 1

[alsa-devel] [PATCH 1/1] SND/PCI/HDA: add __init to snd_hda_add_codec_preset
by Fabian Frederick 10 Jun '14

10 Jun '14

snd_hda_add_codec_preset is only called by __init functions: __init patch_via_init __init patch_sigmatel_init __init patch_si3054_init __init patch_realtek_init __init patch_hdmi_init __init patch_conexant_init __init patch_cmedia_init __init patch_cirrus_init __init patch_ca0132_init __init patch_ca0110_init __init patch_analog_init Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_codec.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4c20277..5770c5c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -77,7 +77,7 @@ static struct hda_vendor_id hda_vendor_ids[] = { static DEFINE_MUTEX(preset_mutex); static LIST_HEAD(hda_preset_tables); -int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset) +int __init snd_hda_add_codec_preset(struct hda_codec_preset_list *preset) { mutex_lock(&preset_mutex); list_add_tail(&preset->list, &hda_preset_tables); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5825aa1..02a9123 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -182,7 +182,7 @@ struct hda_codec_preset_list { }; /* initial hook */ -int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset); +int __init snd_hda_add_codec_preset(struct hda_codec_preset_list *preset); int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset); /* ops set by the preset patch */ -- 1.8.4.5

2 2

[alsa-devel] [PATCH 1/2] misc: atmel-ssc: support to read ssc version
by Bo Shen 10 Jun '14

10 Jun '14

Read SSC IP version to check whether it supports frame sync length extension feature. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- drivers/misc/atmel-ssc.c | 7 +++++++ include/linux/atmel-ssc.h | 3 +++ 2 files changed, 10 insertions(+) diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c index 22de137..9f49bcb 100644 --- a/drivers/misc/atmel-ssc.c +++ b/drivers/misc/atmel-ssc.c @@ -136,6 +136,7 @@ static int ssc_probe(struct platform_device *pdev) struct resource *regs; struct ssc_device *ssc; const struct atmel_ssc_platform_data *plat_dat; + u32 version; ssc = devm_kzalloc(&pdev->dev, sizeof(struct ssc_device), GFP_KERNEL); if (!ssc) { @@ -173,8 +174,14 @@ static int ssc_probe(struct platform_device *pdev) clk_prepare_enable(ssc->clk); ssc_writel(ssc->regs, IDR, -1); ssc_readl(ssc->regs, SR); + version = ssc_readl(ssc->regs, VERSION); clk_disable_unprepare(ssc->clk); + if (version >= 0x300) + ssc->has_fslen_ext = 1; + else + ssc->has_fslen_ext = 0; + ssc->irq = platform_get_irq(pdev, 0); if (!ssc->irq) { dev_dbg(&pdev->dev, "could not get irq\n"); diff --git a/include/linux/atmel-ssc.h b/include/linux/atmel-ssc.h index 571a12e..9ee1b68 100644 --- a/include/linux/atmel-ssc.h +++ b/include/linux/atmel-ssc.h @@ -19,6 +19,7 @@ struct ssc_device { int user; int irq; bool clk_from_rk_pin; + int has_fslen_ext; }; struct ssc_device * __must_check ssc_request(unsigned int ssc_num); @@ -258,6 +259,8 @@ void ssc_free(struct ssc_device *ssc); #define SSC_IMR_TXSYN_SIZE 1 #define SSC_IMR_TXSYN_OFFSET 10 +#define SSC_VERSION 0x000000fc + /* SSC PDC Receive Pointer Register */ #define SSC_PDC_RPR 0x00000100 -- 1.8.5.2

2 3

[alsa-devel] [PATCH] ALSA: hda - Add quirk for ABit AA8XE
by David Henningsson 10 Jun '14

10 Jun '14

Bios does not set up the pin config default correctly (everything is set to zero). Reporter claims that 6stack-dig and 6stack-automute solve the problem. Alsa-info at http://www.alsa-project.org/db/?f=376c0804cbdde90bcd2cb94799407cb1cacf5d05 BugLink: https://bugs.launchpad.net/bugs/1319291 Reported-by: Stefano Statuti <stefano.statuti(a)hotmail.it> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d77745a..9c095fa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1427,6 +1427,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS W5A", ALC880_FIXUP_ASUS_W5A), SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_FIXUP_Z71V), SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_FIXUP_GPIO1), + SND_PCI_QUIRK(0x147b, 0x1045, "ABit AA8XE", ALC880_FIXUP_6ST_AUTOMUTE), SND_PCI_QUIRK(0x1558, 0x5401, "Clevo GPIO2", ALC880_FIXUP_GPIO2), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_FIXUP_UNIWILL_DIG), -- 1.9.1

2 1

[alsa-devel] Fwd: Re: Patch for power on headphone amp on Realtek 883
by asd 10 Jun '14

10 Jun '14

Hi Here is content; #uname -a Linux marian1000.go.ro 3.14-1-amd64 #1 SMP Debian 3.14.4-1 (2014-05-13) x86_64 GNU/Linux #cat /proc/asound/Intel/codec#0 Codec: Realtek ALC883 Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10ec0883 Subsystem Id: 0x1043829f Revision Id: 0x100002 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Device: name="ALC883 Analog", type="Audio", device=0 Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=2 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=4 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=6 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="ALC883 Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC883 Analog", type="Audio", device=0 Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x10 0x10] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC883 Alt Analog", type="Audio", device=2 Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=3, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Center Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Side Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x05 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Front Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=05, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Surround Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=06, enabled=1 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Center Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Switch", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Control: name="Line Out CLFE Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=07, enabled=1 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Side Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Side Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=08, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Rear Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19840: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=03, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Front Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Front Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x02a19c50: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Line Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x0181304f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x4, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=04, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Front Headphone Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x02214c20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4015e601: [N/A] Speaker at Ext N/A Conn = Optical, Color = White DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x00: Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Control: name="SPDIF Phantom Jack", index=0, device=0 Pincap 0x00000010: OUT Pin Default 0x01441130: [Jack] SPDIF Out at Ext Rear Conn = RCA, Color = Black DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=17 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x25 0x0b

2 1

[alsa-devel] [PATCH v2 0/5] As the sama5d3xek and at91sam9n12ek have switch to CCF,
by Bo Shen 10 Jun '14

10 Jun '14

in order to support audio, also switch it to CCF. This patch series based on next-2014-0604 tag. Changes in v2: - wm8904: Remove the mclk checking when try to enable or disable it, as it is checked when required. - dt: List the clock-names, which need by driver. Bo Shen (5): ASoC: atmel_wm8904: switch to CCF ASoC: wm8904: switch to CCF dt: wm8904: add device tree binding document ARM: atmel: sama5d3xek: switch sound to CCF ARM: atmel: at91sam9n12ek: switch sound to CCF Documentation/devicetree/bindings/sound/wm8904.txt | 33 ++++++++++++++ arch/arm/boot/dts/at91sam9n12ek.dts | 2 + arch/arm/boot/dts/sama5d3xmb.dtsi | 2 + sound/soc/atmel/atmel_wm8904.c | 50 ---------------------- sound/soc/codecs/wm8904.c | 11 +++++ 5 files changed, 48 insertions(+), 50 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8904.txt -- 1.8.5.2

3 10

[alsa-devel] [PATCH] sound: soc: pxa2xx-ac97: prepare and unprepare the clocks
by Robert Jarzmik 09 Jun '14

09 Jun '14

Add the clock prepare and unprepare call to the driver initialization phase. This will remove a warning once the PXA architecture is migrated to the clock infrastructure. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- sound/arm/pxa2xx-ac97-lib.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 66de90e..39c3969 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -152,9 +152,9 @@ static inline void pxa_ac97_cold_pxa27x(void) gsr_bits = 0; /* PXA27x Developers Manual section 13.5.2.2.1 */ - clk_enable(ac97conf_clk); + clk_prepare_enable(ac97conf_clk); udelay(5); - clk_disable(ac97conf_clk); + clk_disable_unprepare(ac97conf_clk); GCR = GCR_COLD_RST | GCR_WARM_RST; } #endif @@ -299,14 +299,14 @@ static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id) int pxa2xx_ac97_hw_suspend(void) { GCR |= GCR_ACLINK_OFF; - clk_disable(ac97_clk); + clk_disable_unprepare(ac97_clk); return 0; } EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_suspend); int pxa2xx_ac97_hw_resume(void) { - clk_enable(ac97_clk); + clk_prepare_enable(ac97_clk); return 0; } EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); @@ -368,7 +368,7 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev) goto err_clk; } - ret = clk_enable(ac97_clk); + ret = clk_prepare_enable(ac97_clk); if (ret) goto err_clk2; @@ -403,7 +403,7 @@ void pxa2xx_ac97_hw_remove(struct platform_device *dev) clk_put(ac97conf_clk); ac97conf_clk = NULL; } - clk_disable(ac97_clk); + clk_disable_unprepare(ac97_clk); clk_put(ac97_clk); ac97_clk = NULL; } -- 2.0.0.rc2

2 3

[alsa-devel] [PATCH] ASoC: sigmadsp: Split regmap and I2C support into separate modules
by Lars-Peter Clausen 09 Jun '14

09 Jun '14

When the SigmaDSP module is built-in, but the I2C core is build as a module we'll get a undefined reference: sound/built-in.o: In function `sigma_action_write_i2c': :(.text+0x5d8d4): undefined reference to `i2c_master_send' This can happen if a audio driver that is using the regmap SigmaDSP interface is built into the kernel, but core I2C support is build as a module. To fix this split the SigmaDSP module into three modules, one module providing the core infrastructure and two small modules implementing the regmap and I2C interfaces. This allows e.g. the core infrastructure and regmap support to be built into the kernel while I2C support can still be build as a module. Fixes: dab464b60 ("ASoC: Add ADAU1361/ADAU1761 audio CODEC support") Reported-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/Kconfig | 12 +++++-- sound/soc/codecs/Makefile | 4 +++ sound/soc/codecs/sigmadsp-i2c.c | 35 ++++++++++++++++++++ sound/soc/codecs/sigmadsp-regmap.c | 36 +++++++++++++++++++++ sound/soc/codecs/sigmadsp.c | 65 ++------------------------------------ sound/soc/codecs/sigmadsp.h | 20 ++++++++++++ 6 files changed, 107 insertions(+), 65 deletions(-) create mode 100644 sound/soc/codecs/sigmadsp-i2c.c create mode 100644 sound/soc/codecs/sigmadsp-regmap.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cbfa1e1..0b9571c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -225,11 +225,11 @@ config SND_SOC_ADAU1373 config SND_SOC_ADAU1701 tristate "Analog Devices ADAU1701 CODEC" depends on I2C - select SND_SOC_SIGMADSP + select SND_SOC_SIGMADSP_I2C config SND_SOC_ADAU17X1 tristate - select SND_SOC_SIGMADSP + select SND_SOC_SIGMADSP_REGMAP config SND_SOC_ADAU1761 tristate @@ -476,6 +476,14 @@ config SND_SOC_SIGMADSP tristate select CRC32 +config SND_SOC_SIGMADSP_I2C + tristate + select SND_SOC_SIGMADSP + +config SND_SOC_SIGMADSP_REGMAP + tristate + select SND_SOC_SIGMADSP + config SND_SOC_SIRF_AUDIO_CODEC tristate "SiRF SoC internal audio codec" select REGMAP_MMIO diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index be3377b..1bd6e1c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -77,6 +77,8 @@ snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o +snd-soc-sigmadsp-i2c-objs := sigmadsp-i2c.o +snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o snd-soc-si476x-objs := si476x.o snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o snd-soc-sn95031-objs := sn95031.o @@ -240,6 +242,8 @@ obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o +obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o +obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c new file mode 100644 index 0000000..246081a --- /dev/null +++ b/sound/soc/codecs/sigmadsp-i2c.c @@ -0,0 +1,35 @@ +/* + * Load Analog Devices SigmaStudio firmware files + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/i2c.h> +#include <linux/export.h> +#include <linux/module.h> + +#include "sigmadsp.h" + +static int sigma_action_write_i2c(void *control_data, + const struct sigma_action *sa, size_t len) +{ + return i2c_master_send(control_data, (const unsigned char *)&sa->addr, + len); +} + +int process_sigma_firmware(struct i2c_client *client, const char *name) +{ + struct sigma_firmware ssfw; + + ssfw.control_data = client; + ssfw.write = sigma_action_write_i2c; + + return _process_sigma_firmware(&client->dev, &ssfw, name); +} +EXPORT_SYMBOL(process_sigma_firmware); + +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_DESCRIPTION("SigmaDSP I2C firmware loader"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp-regmap.c b/sound/soc/codecs/sigmadsp-regmap.c new file mode 100644 index 0000000..f78ed8d --- /dev/null +++ b/sound/soc/codecs/sigmadsp-regmap.c @@ -0,0 +1,36 @@ +/* + * Load Analog Devices SigmaStudio firmware files + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/regmap.h> +#include <linux/export.h> +#include <linux/module.h> + +#include "sigmadsp.h" + +static int sigma_action_write_regmap(void *control_data, + const struct sigma_action *sa, size_t len) +{ + return regmap_raw_write(control_data, be16_to_cpu(sa->addr), + sa->payload, len - 2); +} + +int process_sigma_firmware_regmap(struct device *dev, struct regmap *regmap, + const char *name) +{ + struct sigma_firmware ssfw; + + ssfw.control_data = regmap; + ssfw.write = sigma_action_write_regmap; + + return _process_sigma_firmware(dev, &ssfw, name); +} +EXPORT_SYMBOL(process_sigma_firmware_regmap); + +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_DESCRIPTION("SigmaDSP regmap firmware loader"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 4068f24..f2de7e0 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -34,23 +34,6 @@ enum { SIGMA_ACTION_END, }; -struct sigma_action { - u8 instr; - u8 len_hi; - __le16 len; - __be16 addr; - unsigned char payload[]; -} __packed; - -struct sigma_firmware { - const struct firmware *fw; - size_t pos; - - void *control_data; - int (*write)(void *control_data, const struct sigma_action *sa, - size_t len); -}; - static inline u32 sigma_action_len(struct sigma_action *sa) { return (sa->len_hi << 16) | le16_to_cpu(sa->len); @@ -138,7 +121,7 @@ process_sigma_actions(struct sigma_firmware *ssfw) return 0; } -static int _process_sigma_firmware(struct device *dev, +int _process_sigma_firmware(struct device *dev, struct sigma_firmware *ssfw, const char *name) { int ret; @@ -197,50 +180,6 @@ static int _process_sigma_firmware(struct device *dev, return ret; } - -#if IS_ENABLED(CONFIG_I2C) - -static int sigma_action_write_i2c(void *control_data, - const struct sigma_action *sa, size_t len) -{ - return i2c_master_send(control_data, (const unsigned char *)&sa->addr, - len); -} - -int process_sigma_firmware(struct i2c_client *client, const char *name) -{ - struct sigma_firmware ssfw; - - ssfw.control_data = client; - ssfw.write = sigma_action_write_i2c; - - return _process_sigma_firmware(&client->dev, &ssfw, name); -} -EXPORT_SYMBOL(process_sigma_firmware); - -#endif - -#if IS_ENABLED(CONFIG_REGMAP) - -static int sigma_action_write_regmap(void *control_data, - const struct sigma_action *sa, size_t len) -{ - return regmap_raw_write(control_data, be16_to_cpu(sa->addr), - sa->payload, len - 2); -} - -int process_sigma_firmware_regmap(struct device *dev, struct regmap *regmap, - const char *name) -{ - struct sigma_firmware ssfw; - - ssfw.control_data = regmap; - ssfw.write = sigma_action_write_regmap; - - return _process_sigma_firmware(dev, &ssfw, name); -} -EXPORT_SYMBOL(process_sigma_firmware_regmap); - -#endif +EXPORT_SYMBOL_GPL(_process_sigma_firmware); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp.h b/sound/soc/codecs/sigmadsp.h index e439cbd..c47cd23 100644 --- a/sound/soc/codecs/sigmadsp.h +++ b/sound/soc/codecs/sigmadsp.h @@ -12,6 +12,26 @@ #include <linux/device.h> #include <linux/regmap.h> +struct sigma_action { + u8 instr; + u8 len_hi; + __le16 len; + __be16 addr; + unsigned char payload[]; +} __packed; + +struct sigma_firmware { + const struct firmware *fw; + size_t pos; + + void *control_data; + int (*write)(void *control_data, const struct sigma_action *sa, + size_t len); +}; + +int _process_sigma_firmware(struct device *dev, + struct sigma_firmware *ssfw, const char *name); + struct i2c_client; extern int process_sigma_firmware(struct i2c_client *client, const char *name); -- 1.8.0

2 2

[alsa-devel] [PATCH v2] ASoC: wm5102: Add controls to allow shaping of ultrasonic response
by Charles Keepax 09 Jun '14

09 Jun '14

Add controls to allow custom shaping of the ultrasonic response. This custom shaping can be turned on/off at runtime, although, it should be noted that settings will not affect a currently open audio stream, they will be applied when the next audio stream is started. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Now uses the existing BYTES_EXT control - Delayed other patch in the series, this patch should still work fine, just some pointless caching of the DAC_COMP registers will happen include/linux/mfd/arizona/core.h | 3 ++ sound/soc/codecs/arizona.c | 34 +++++++++++++++++++++ sound/soc/codecs/wm5102.c | 62 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 99 insertions(+), 0 deletions(-) diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index 11783b5..5592651 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -113,6 +113,9 @@ struct arizona { int tdm_width[ARIZONA_MAX_AIF]; int tdm_slots[ARIZONA_MAX_AIF]; + + uint16_t dac_comp_coeff; + uint8_t dac_comp_enabled; }; int arizona_clk32k_enable(struct arizona *arizona); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e77f61c..41b56ee 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1127,6 +1127,31 @@ static int arizona_startup(struct snd_pcm_substream *substream, constraint); } +static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec, + unsigned int rate) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + struct reg_default dac_comp[] = { + { 0x80, 0x3 }, + { ARIZONA_DAC_COMP_1, 0 }, + { ARIZONA_DAC_COMP_2, 0 }, + { 0x80, 0x0 }, + }; + + mutex_lock(&codec->mutex); + + dac_comp[1].def = arizona->dac_comp_coeff; + if (rate >= 176400) + dac_comp[2].def = arizona->dac_comp_enabled; + + mutex_unlock(&codec->mutex); + + regmap_multi_reg_write(arizona->regmap, + dac_comp, + ARRAY_SIZE(dac_comp)); +} + static int arizona_hw_params_rate(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1153,6 +1178,15 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream, switch (dai_priv->clk) { case ARIZONA_CLK_SYSCLK: + switch (priv->arizona->type) { + case WM5102: + arizona_wm5102_set_dac_comp(codec, + params_rate(params)); + break; + default: + break; + } + snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, ARIZONA_SAMPLE_RATE_1_MASK, sr_val); if (base) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 289b64d..8613504 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -612,6 +612,62 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, return 0; } +static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + uint16_t data; + + mutex_lock(&codec->mutex); + data = cpu_to_be16(arizona->dac_comp_coeff); + memcpy(ucontrol->value.bytes.data, &data, sizeof(data)); + mutex_unlock(&codec->mutex); + + return 0; +} + +static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + + mutex_lock(&codec->mutex); + memcpy(&arizona->dac_comp_coeff, ucontrol->value.bytes.data, + sizeof(arizona->dac_comp_coeff)); + arizona->dac_comp_coeff = be16_to_cpu(arizona->dac_comp_coeff); + mutex_unlock(&codec->mutex); + + return 0; +} + +static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + + mutex_lock(&codec->mutex); + ucontrol->value.integer.value[0] = arizona->dac_comp_enabled; + mutex_unlock(&codec->mutex); + + return 0; +} + +static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + + mutex_lock(&codec->mutex); + arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + mutex_unlock(&codec->mutex); + + return 0; +} + static const char *wm5102_osr_text[] = { "Low power", "Normal", "High performance", }; @@ -843,6 +899,12 @@ SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv), SOC_ENUM("Noise Gate Hold", arizona_ng_hold), +SND_SOC_BYTES_EXT("Output Compensation Coefficient", 2, + wm5102_out_comp_coeff_get, wm5102_out_comp_coeff_put), + +SOC_SINGLE_EXT("Output Compensation Switch", 0, 0, 1, 0, + wm5102_out_comp_switch_get, wm5102_out_comp_switch_put), + WM5102_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L), WM5102_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R), WM5102_NG_SRC("HPOUT2L", ARIZONA_NOISE_GATE_SELECT_2L), -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: rl6231: Remove unneeded inclusion of header files
by Axel Lin 09 Jun '14

09 Jun '14

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rl6231.c | 19 ------------------- 1 file changed, 19 deletions(-) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 7b82fbe..56650d6 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -11,25 +11,6 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/init.h> -#include <linux/delay.h> -#include <linux/pm.h> -#include <linux/gpio.h> -#include <linux/i2c.h> -#include <linux/regmap.h> -#include <linux/of.h> -#include <linux/of_gpio.h> -#include <linux/platform_device.h> -#include <linux/spi/spi.h> -#include <linux/acpi.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <sound/initval.h> -#include <sound/tlv.h> #include "rl6231.h" -- 1.8.3.2

3 2

[alsa-devel] [PATCH 1/3] ASoC: rt5677: Convert to use module_i2c_driver
by Axel Lin 09 Jun '14

09 Jun '14

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5677.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 833231e..60fff7f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3480,18 +3480,7 @@ static struct i2c_driver rt5677_i2c_driver = { .remove = rt5677_i2c_remove, .id_table = rt5677_i2c_id, }; - -static int __init rt5677_modinit(void) -{ - return i2c_add_driver(&rt5677_i2c_driver); -} -module_init(rt5677_modinit); - -static void __exit rt5677_modexit(void) -{ - i2c_del_driver(&rt5677_i2c_driver); -} -module_exit(rt5677_modexit); +module_i2c_driver(rt5677_i2c_driver); MODULE_DESCRIPTION("ASoC RT5677 driver"); MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); -- 1.8.3.2

3 6

[alsa-devel] [PATCH] ASoC: fsl_spdif: Fix integer overflow when calculating divisors
by Anssi Hannula 09 Jun '14

09 Jun '14

The calculation code does u64 = (u32 - u32) * 100000; The 64 bits are of no help here as the type is casted only after the multiplication, and therefore the result may overflow, possibly causing inoptimal or wrong clock setup in an unfortunate case (the maximum result value of the first substraction is currently 47999). Fix the code to cast before multiplication. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index b912d45a2a4c..0bd9136279a2 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1076,7 +1076,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, goto out; } else if (arate / rate[index] == 1) { /* A little bigger than expect */ - sub = (arate - rate[index]) * 100000; + sub = (u64)(arate - rate[index]) * 100000; do_div(sub, rate[index]); if (sub >= savesub) continue; @@ -1086,7 +1086,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, spdif_priv->txrate[index] = arate; } else if (rate[index] / arate == 1) { /* A little smaller than expect */ - sub = (rate[index] - arate) * 100000; + sub = (u64)(rate[index] - arate) * 100000; do_div(sub, rate[index]); if (sub >= savesub) continue; -- 1.8.4.5

3 2

[alsa-devel] [PATCH] ASoC: dapm: Make sure register value is in sync with DAPM kcontrol state
by Jarkko Nikula 09 Jun '14

09 Jun '14

Commit c9e065c27fe9 ("ASoC: dapm: Make sure to always update the DAPM graph in _put_volsw()") stopped updating register values in those cases where initial after boot state of kcontrol appears to not change but where register value still needs update because it is not in sync with the kcontrol state. Fix this by doing snd_soc_test_bits() unconditionally as it was before but by using separate flags for kcontrol and register state changes. This allow both DAPM graph to be updated when disabling auto-muted control and update register if it is out-of-sync in respect of kcontrol state. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- For 3.16-rc. I noticed this with RT5640 where alsa-utils wasn't able to unmute some mixers controls during boot. For instance "HPO MIX DAC1 Switch" which is inverted in hw thus muted when bit is 1 and reset default is also 1. Reason for this was that dapm_kcontrol_set_value() returns false when unmuting inverted control because data->value == value == 0 after boot. Only way to unmute was to do mute-unmute cycle. I don't see there's reason to update initial data->value states from hw/regmap but just do snd_soc_test_bits() unconditionally in snd_soc_dapm_put_volsw() as it was before. --- sound/soc/soc-dapm.c | 29 ++++++++++++++++------------- 1 file changed, 16 insertions(+), 13 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a74b9bf23d9f..cdc837ed144d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2755,7 +2755,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val; - int connect, change; + int connect, change, reg_change = 0; struct snd_soc_dapm_update update; int ret = 0; @@ -2773,20 +2773,23 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = dapm_kcontrol_set_value(kcontrol, val); - if (change) { - if (reg != SND_SOC_NOPM) { - mask = mask << shift; - val = val << shift; - - if (snd_soc_test_bits(codec, reg, mask, val)) { - update.kcontrol = kcontrol; - update.reg = reg; - update.mask = mask; - update.val = val; - card->update = &update; - } + if (reg != SND_SOC_NOPM) { + mask = mask << shift; + val = val << shift; + + reg_change = snd_soc_test_bits(codec, reg, mask, val); + } + + if (change || reg_change) { + if (reg_change) { + update.kcontrol = kcontrol; + update.reg = reg; + update.mask = mask; + update.val = val; + card->update = &update; } + change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect); -- 2.0.0

3 2

[alsa-devel] [PATCH] ASoC: Intel: byt-rt5640: Enable headset mic bias voltage
by Jarkko Nikula 09 Jun '14

09 Jun '14

Connect "Headset Mic" to "MICBIAS1" supply widget of RT5640 in order to enable bias voltage for headset microphones. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/byt-rt5640.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 53d160d39972..234a58de3c53 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -34,6 +34,7 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { }; static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { + {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, {"IN2N", NULL, "Headset Mic"}, {"DMIC1", NULL, "Internal Mic"}, -- 2.0.0

2 1

[alsa-devel] patch_realtek. issue
by asd 09 Jun '14

09 Jun '14

Hi Sending command to the codec from patch_realtek.c is on big mistake because entire structure (struct hda_codec.c) is bloched and nust be accesed via mutex.

1 0

[alsa-devel] [PATCH 1/2] Revert "ALSA: hda - mask buggy stream DMA0 for Broadwell display controller"
by libin.yang＠intel.com 09 Jun '14

09 Jun '14

From: Libin Yang <libin.yang(a)intel.com> This reverts commit 7189eb9b8f7962474956196c301676470542f253. It will use LPIB to get the DMA position on Broadwell HDMI Audio. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/hda_intel.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index cd77b9b..f93f338 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1367,12 +1367,6 @@ static int azx_first_init(struct azx *chip) /* initialize streams */ azx_init_stream(chip); - /* workaround for Broadwell HDMI: the first stream is broken, - * so mask it by keeping it as if opened - */ - if (pci->vendor == 0x8086 && pci->device == 0x160c) - chip->azx_dev[0].opened = 1; - /* initialize chip */ azx_init_pci(chip); azx_init_chip(chip, (probe_only[dev] & 2) == 0); -- 1.8.3.2

3 3

[alsa-devel] [PATCH] ASoC: rsnd: enable DVC when capture
by Kuninori Morimoto 09 Jun '14

09 Jun '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current DVC can be enabled only when playback, but, this came from misunderstanding. It is not correct. DVC <-> DMA relationship is... Playback: MEM -> DMAC -> SRC -> DVC -> DMACp -> SSI Capture: SSI -> DMACp -> SRC -> DVC -> DMAC -> MEM DVC can be used for both Playback/Capture Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dvc.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index ed00070..deef310 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -191,24 +191,20 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, struct snd_soc_pcm_runtime *rtd) { struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct device *dev = rsnd_priv_to_dev(priv); struct snd_card *card = rtd->card->snd_card; struct snd_kcontrol *kctrl; static struct snd_kcontrol_new knew = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Playback Volume", .info = rsnd_dvc_volume_info, .get = rsnd_dvc_volume_get, .put = rsnd_dvc_volume_put, }; int ret; - if (!rsnd_dai_is_play(rdai, io)) { - dev_err(dev, "DVC%d is connected to Capture DAI\n", - rsnd_mod_id(mod)); - return -EINVAL; - } + if (rsnd_dai_is_play(rdai, io)) + knew.name = "Playback Volume"; + else + knew.name = "Capture Volume"; kctrl = snd_ctl_new1(&knew, mod); if (!kctrl) -- 1.7.9.5

1 0

[alsa-devel] Patch for power on headphone amp on Realtek 883
by asd 09 Jun '14

09 Jun '14

Index: hda/hda_codec.h =================================================================== --- hda.orig/hda_codec.h 2014-06-08 22:54:07.000000000 +0300 +++ hda/hda_codec.h 2014-06-09 09:43:42.359115743 +0300 @@ -259,6 +259,7 @@ HDA_PCM_NTYPES }; + /* for PCM creation */ struct hda_pcm { char *name; @@ -287,6 +288,14 @@ u32 subsystem_id; u32 revision_id; + /* custom pin complex headphone amp */ + bool hp_set; + hda_nid_t hp_front; + hda_nid_t hp_rear; + hda_nid_t hp_clfe; + hda_nid_t hp_side; + hda_nid_t hp_fout; + /* detected preset */ const struct hda_codec_preset *preset; struct module *owner; Index: hda/patch_realtek.c =================================================================== --- hda.orig/patch_realtek.c 2014-06-07 12:38:24.000000000 +0300 +++ hda/patch_realtek.c 2014-06-09 09:22:53.306622242 +0300 @@ -322,6 +322,12 @@ case 0x10ec0880: case 0x10ec0882: case 0x10ec0883: + codec->hp_set = 1; + codec->hp_front = 0x14; + codec->hp_rear = 0x15; + codec->hp_clfe = 0x16; + codec->hp_side = 0x17; + codec->hp_fout = 0x1b; case 0x10ec0885: case 0x10ec0887: /*case 0x10ec0889:*/ /* this causes an SPDIF problem */ Index: hda/hda_codec.c =================================================================== --- hda.orig/hda_codec.c 2014-06-08 22:54:07.000000000 +0300 +++ hda/hda_codec.c 2014-06-09 09:45:32.473468469 +0300 @@ -27,6 +27,7 @@ #include <linux/mutex.h> #include <linux/module.h> #include <linux/async.h> +#include <linux/delay.h> #include <sound/core.h> #include "hda_codec.h" #include <sound/asoundef.h> @@ -120,6 +121,56 @@ #define hda_call_pm_notify(codec, state) {} #endif +static void enable_hp_codec(struct hda_codec *codec) +{ + + mutex_unlock(&codec->control_mutex); + if ((codec->hp_set = 1)) { + snd_hda_codec_write(codec, codec->hp_front, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0); + + snd_hda_codec_write(codec, codec->hp_rear, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0); + + snd_hda_codec_write(codec, codec->hp_clfe, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0); + + snd_hda_codec_write(codec, codec->hp_side, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0); + + snd_hda_codec_write(codec, codec->hp_fout, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0); + } + + mutex_lock(&codec->control_mutex); + +} + +static void mute_all_analog_outs(struct hda_codec *codec) +{ + mutex_unlock(&codec->control_mutex); + if ((codec->hp_set = 1)) { + snd_hda_codec_write(codec,codec->hp_front, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + + snd_hda_codec_write(codec,codec->hp_rear, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + + snd_hda_codec_write(codec,codec->hp_clfe, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + + snd_hda_codec_write(codec,codec->hp_side, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + + snd_hda_codec_write(codec,codec->hp_fout, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + } + mutex_lock(&codec->control_mutex); +} + + + + /** * snd_hda_get_jack_location - Give a location string of the jack * @cfg: pin default config value @@ -5418,8 +5469,9 @@ /* FIXME: need notify? */ } } - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); } + enable_hp_codec(codec); return snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); } @@ -5530,6 +5582,7 @@ mout->dig_out_used = 0; } mutex_unlock(&codec->spdif_mutex); + mute_all_analog_outs(codec); return 0; } EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_cleanup);

2 1

[alsa-devel] [PATCH] ASoC: add RT5677 CODEC driver
by Oder Chiou 09 Jun '14

09 Jun '14

This patch adds the Realtek ALC5677 codec driver. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5677.h | 21 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5677.c | 3498 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5677.h | 1451 +++++++++++++++++++ 5 files changed, 4976 insertions(+) create mode 100644 include/sound/rt5677.h create mode 100644 sound/soc/codecs/rt5677.c create mode 100644 sound/soc/codecs/rt5677.h diff --git a/include/sound/rt5677.h b/include/sound/rt5677.h new file mode 100644 index 0000000..3da1431 --- /dev/null +++ b/include/sound/rt5677.h @@ -0,0 +1,21 @@ +/* + * linux/sound/rt5677.h -- Platform data for RT5677 + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT5677_H +#define __LINUX_SND_RT5677_H + +struct rt5677_platform_data { + /* IN1 IN2 can optionally be differential */ + bool in1_diff; + bool in2_diff; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c59943a..08c5e3a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -74,6 +74,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5640 if I2C select SND_SOC_RT5645 if I2C select SND_SOC_RT5651 if I2C + select SND_SOC_RT5677 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIRF_AUDIO_CODEC @@ -414,6 +415,9 @@ config SND_SOC_RT5645 config SND_SOC_RT5651 tristate +config SND_SOC_RT5677 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate "Freescale SGTL5000 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1ccdaf0..afd8d93 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -64,6 +64,7 @@ snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o +snd-soc-rt5677-objs := rt5677.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -220,6 +221,7 @@ obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o +obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c new file mode 100644 index 0000000..833231e --- /dev/null +++ b/sound/soc/codecs/rt5677.c @@ -0,0 +1,3498 @@ +/* + * rt5677.c -- RT5677 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/fs.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5677.h" + +#define RT5677_DEVICE_ID 0x6327 + +#define RT5677_PR_RANGE_BASE (0xff + 1) +#define RT5677_PR_SPACING 0x100 + +#define RT5677_PR_BASE (RT5677_PR_RANGE_BASE + (0 * RT5677_PR_SPACING)) + +static const struct regmap_range_cfg rt5677_ranges[] = { + { + .name = "PR", + .range_min = RT5677_PR_BASE, + .range_max = RT5677_PR_BASE + 0xfd, + .selector_reg = RT5677_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5677_PRIV_DATA, + .window_len = 0x1, + }, +}; + +static const struct reg_default init_list[] = { + {RT5677_PR_BASE + 0x3d, 0x364d}, + {RT5677_PR_BASE + 0x17, 0x4fc0}, + {RT5677_PR_BASE + 0x13, 0x0312}, + {RT5677_PR_BASE + 0x1e, 0x0000}, + {RT5677_PR_BASE + 0x12, 0x0eaa}, + {RT5677_PR_BASE + 0x14, 0x018a}, +}; +#define RT5677_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt5677_reg[] = { + {RT5677_RESET , 0x0000}, + {RT5677_LOUT1 , 0xa800}, + {RT5677_IN1 , 0x0000}, + {RT5677_MICBIAS , 0x0000}, + {RT5677_SLIMBUS_PARAM , 0x0000}, + {RT5677_SLIMBUS_RX , 0x0000}, + {RT5677_SLIMBUS_CTRL , 0x0000}, + {RT5677_SIDETONE_CTRL , 0x000b}, + {RT5677_ANA_DAC1_2_3_SRC , 0x0000}, + {RT5677_IF_DSP_DAC3_4_MIXER , 0x1111}, + {RT5677_DAC4_DIG_VOL , 0xafaf}, + {RT5677_DAC3_DIG_VOL , 0xafaf}, + {RT5677_DAC1_DIG_VOL , 0xafaf}, + {RT5677_DAC2_DIG_VOL , 0xafaf}, + {RT5677_IF_DSP_DAC2_MIXER , 0x0011}, + {RT5677_STO1_ADC_DIG_VOL , 0x2f2f}, + {RT5677_MONO_ADC_DIG_VOL , 0x2f2f}, + {RT5677_STO1_2_ADC_BST , 0x0000}, + {RT5677_STO2_ADC_DIG_VOL , 0x2f2f}, + {RT5677_ADC_BST_CTRL2 , 0x0000}, + {RT5677_STO3_4_ADC_BST , 0x0000}, + {RT5677_STO3_ADC_DIG_VOL , 0x2f2f}, + {RT5677_STO4_ADC_DIG_VOL , 0x2f2f}, + {RT5677_STO4_ADC_MIXER , 0xd4c0}, + {RT5677_STO3_ADC_MIXER , 0xd4c0}, + {RT5677_STO2_ADC_MIXER , 0xd4c0}, + {RT5677_STO1_ADC_MIXER , 0xd4c0}, + {RT5677_MONO_ADC_MIXER , 0xd4d1}, + {RT5677_ADC_IF_DSP_DAC1_MIXER , 0x8080}, + {RT5677_STO1_DAC_MIXER , 0xaaaa}, + {RT5677_MONO_DAC_MIXER , 0xaaaa}, + {RT5677_DD1_MIXER , 0xaaaa}, + {RT5677_DD2_MIXER , 0xaaaa}, + {RT5677_IF3_DATA , 0x0000}, + {RT5677_IF4_DATA , 0x0000}, + {RT5677_PDM_OUT_CTRL , 0x8888}, + {RT5677_PDM_DATA_CTRL1 , 0x0000}, + {RT5677_PDM_DATA_CTRL2 , 0x0000}, + {RT5677_PDM1_DATA_CTRL2 , 0x0000}, + {RT5677_PDM1_DATA_CTRL3 , 0x0000}, + {RT5677_PDM1_DATA_CTRL4 , 0x0000}, + {RT5677_PDM2_DATA_CTRL2 , 0x0000}, + {RT5677_PDM2_DATA_CTRL3 , 0x0000}, + {RT5677_PDM2_DATA_CTRL4 , 0x0000}, + {RT5677_TDM1_CTRL1 , 0x0300}, + {RT5677_TDM1_CTRL2 , 0x0000}, + {RT5677_TDM1_CTRL3 , 0x4000}, + {RT5677_TDM1_CTRL4 , 0x0123}, + {RT5677_TDM1_CTRL5 , 0x4567}, + {RT5677_TDM2_CTRL1 , 0x0300}, + {RT5677_TDM2_CTRL2 , 0x0000}, + {RT5677_TDM2_CTRL3 , 0x4000}, + {RT5677_TDM2_CTRL4 , 0x0123}, + {RT5677_TDM2_CTRL5 , 0x4567}, + {RT5677_I2C_MASTER_CTRL1 , 0x0001}, + {RT5677_I2C_MASTER_CTRL2 , 0x0000}, + {RT5677_I2C_MASTER_CTRL3 , 0x0000}, + {RT5677_I2C_MASTER_CTRL4 , 0x0000}, + {RT5677_I2C_MASTER_CTRL5 , 0x0000}, + {RT5677_I2C_MASTER_CTRL6 , 0x0000}, + {RT5677_I2C_MASTER_CTRL7 , 0x0000}, + {RT5677_I2C_MASTER_CTRL8 , 0x0000}, + {RT5677_DMIC_CTRL1 , 0x1505}, + {RT5677_DMIC_CTRL2 , 0x0055}, + {RT5677_HAP_GENE_CTRL1 , 0x0111}, + {RT5677_HAP_GENE_CTRL2 , 0x0064}, + {RT5677_HAP_GENE_CTRL3 , 0xef0e}, + {RT5677_HAP_GENE_CTRL4 , 0xf0f0}, + {RT5677_HAP_GENE_CTRL5 , 0xef0e}, + {RT5677_HAP_GENE_CTRL6 , 0xf0f0}, + {RT5677_HAP_GENE_CTRL7 , 0xef0e}, + {RT5677_HAP_GENE_CTRL8 , 0xf0f0}, + {RT5677_HAP_GENE_CTRL9 , 0xf000}, + {RT5677_HAP_GENE_CTRL10 , 0x0000}, + {RT5677_PWR_DIG1 , 0x0000}, + {RT5677_PWR_DIG2 , 0x0000}, + {RT5677_PWR_ANLG1 , 0x0055}, + {RT5677_PWR_ANLG2 , 0x0000}, + {RT5677_PWR_DSP1 , 0x0001}, + {RT5677_PWR_DSP_ST , 0x0000}, + {RT5677_PWR_DSP2 , 0x0000}, + {RT5677_ADC_DAC_HPF_CTRL1 , 0x0e00}, + {RT5677_PRIV_INDEX , 0x0000}, + {RT5677_PRIV_DATA , 0x0000}, + {RT5677_I2S4_SDP , 0x8000}, + {RT5677_I2S1_SDP , 0x8000}, + {RT5677_I2S2_SDP , 0x8000}, + {RT5677_I2S3_SDP , 0x8000}, + {RT5677_CLK_TREE_CTRL1 , 0x1111}, + {RT5677_CLK_TREE_CTRL2 , 0x1111}, + {RT5677_CLK_TREE_CTRL3 , 0x0000}, + {RT5677_PLL1_CTRL1 , 0x0000}, + {RT5677_PLL1_CTRL2 , 0x0000}, + {RT5677_PLL2_CTRL1 , 0x0c60}, + {RT5677_PLL2_CTRL2 , 0x2000}, + {RT5677_GLB_CLK1 , 0x0000}, + {RT5677_GLB_CLK2 , 0x0000}, + {RT5677_ASRC_1 , 0x0000}, + {RT5677_ASRC_2 , 0x0000}, + {RT5677_ASRC_3 , 0x0000}, + {RT5677_ASRC_4 , 0x0000}, + {RT5677_ASRC_5 , 0x0000}, + {RT5677_ASRC_6 , 0x0000}, + {RT5677_ASRC_7 , 0x0000}, + {RT5677_ASRC_8 , 0x0000}, + {RT5677_ASRC_9 , 0x0000}, + {RT5677_ASRC_10 , 0x0000}, + {RT5677_ASRC_11 , 0x0000}, + {RT5677_ASRC_12 , 0x0008}, + {RT5677_ASRC_13 , 0x0000}, + {RT5677_ASRC_14 , 0x0000}, + {RT5677_ASRC_15 , 0x0000}, + {RT5677_ASRC_16 , 0x0000}, + {RT5677_ASRC_17 , 0x0000}, + {RT5677_ASRC_18 , 0x0000}, + {RT5677_ASRC_19 , 0x0000}, + {RT5677_ASRC_20 , 0x0000}, + {RT5677_ASRC_21 , 0x000c}, + {RT5677_ASRC_22 , 0x0000}, + {RT5677_ASRC_23 , 0x0000}, + {RT5677_VAD_CTRL1 , 0x2184}, + {RT5677_VAD_CTRL2 , 0x010a}, + {RT5677_VAD_CTRL3 , 0x0aea}, + {RT5677_VAD_CTRL4 , 0x000c}, + {RT5677_VAD_CTRL5 , 0x0000}, + {RT5677_DSP_INB_CTRL1 , 0x0000}, + {RT5677_DSP_INB_CTRL2 , 0x0000}, + {RT5677_DSP_IN_OUTB_CTRL , 0x0000}, + {RT5677_DSP_OUTB0_1_DIG_VOL , 0x2f2f}, + {RT5677_DSP_OUTB2_3_DIG_VOL , 0x2f2f}, + {RT5677_DSP_OUTB4_5_DIG_VOL , 0x2f2f}, + {RT5677_DSP_OUTB6_7_DIG_VOL , 0x2f2f}, + {RT5677_ADC_EQ_CTRL1 , 0x6000}, + {RT5677_ADC_EQ_CTRL2 , 0x0000}, + {RT5677_EQ_CTRL1 , 0xc000}, + {RT5677_EQ_CTRL2 , 0x0000}, + {RT5677_EQ_CTRL3 , 0x0000}, + {RT5677_SOFT_VOL_ZERO_CROSS1 , 0x0009}, + {RT5677_JD_CTRL1 , 0x0000}, + {RT5677_JD_CTRL2 , 0x0000}, + {RT5677_JD_CTRL3 , 0x0000}, + {RT5677_IRQ_CTRL1 , 0x0000}, + {RT5677_IRQ_CTRL2 , 0x0000}, + {RT5677_GPIO_ST , 0x0000}, + {RT5677_GPIO_CTRL1 , 0x0000}, + {RT5677_GPIO_CTRL2 , 0x0000}, + {RT5677_GPIO_CTRL3 , 0x0000}, + {RT5677_STO1_ADC_HI_FILTER1 , 0xb320}, + {RT5677_STO1_ADC_HI_FILTER2 , 0x0000}, + {RT5677_MONO_ADC_HI_FILTER1 , 0xb300}, + {RT5677_MONO_ADC_HI_FILTER2 , 0x0000}, + {RT5677_STO2_ADC_HI_FILTER1 , 0xb300}, + {RT5677_STO2_ADC_HI_FILTER2 , 0x0000}, + {RT5677_STO3_ADC_HI_FILTER1 , 0xb300}, + {RT5677_STO3_ADC_HI_FILTER2 , 0x0000}, + {RT5677_STO4_ADC_HI_FILTER1 , 0xb300}, + {RT5677_STO4_ADC_HI_FILTER2 , 0x0000}, + {RT5677_MB_DRC_CTRL1 , 0x0f20}, + {RT5677_DRC1_CTRL1 , 0x001f}, + {RT5677_DRC1_CTRL2 , 0x020c}, + {RT5677_DRC1_CTRL3 , 0x1f00}, + {RT5677_DRC1_CTRL4 , 0x0000}, + {RT5677_DRC1_CTRL5 , 0x0000}, + {RT5677_DRC1_CTRL6 , 0x0029}, + {RT5677_DRC2_CTRL1 , 0x001f}, + {RT5677_DRC2_CTRL2 , 0x020c}, + {RT5677_DRC2_CTRL3 , 0x1f00}, + {RT5677_DRC2_CTRL4 , 0x0000}, + {RT5677_DRC2_CTRL5 , 0x0000}, + {RT5677_DRC2_CTRL6 , 0x0029}, + {RT5677_DRC1_HL_CTRL1 , 0x8000}, + {RT5677_DRC1_HL_CTRL2 , 0x0200}, + {RT5677_DRC2_HL_CTRL1 , 0x8000}, + {RT5677_DRC2_HL_CTRL2 , 0x0200}, + {RT5677_DSP_INB1_SRC_CTRL1 , 0x5800}, + {RT5677_DSP_INB1_SRC_CTRL2 , 0x0000}, + {RT5677_DSP_INB1_SRC_CTRL3 , 0x0000}, + {RT5677_DSP_INB1_SRC_CTRL4 , 0x0800}, + {RT5677_DSP_INB2_SRC_CTRL1 , 0x5800}, + {RT5677_DSP_INB2_SRC_CTRL2 , 0x0000}, + {RT5677_DSP_INB2_SRC_CTRL3 , 0x0000}, + {RT5677_DSP_INB2_SRC_CTRL4 , 0x0800}, + {RT5677_DSP_INB3_SRC_CTRL1 , 0x5800}, + {RT5677_DSP_INB3_SRC_CTRL2 , 0x0000}, + {RT5677_DSP_INB3_SRC_CTRL3 , 0x0000}, + {RT5677_DSP_INB3_SRC_CTRL4 , 0x0800}, + {RT5677_DSP_OUTB1_SRC_CTRL1 , 0x5800}, + {RT5677_DSP_OUTB1_SRC_CTRL2 , 0x0000}, + {RT5677_DSP_OUTB1_SRC_CTRL3 , 0x0000}, + {RT5677_DSP_OUTB1_SRC_CTRL4 , 0x0800}, + {RT5677_DSP_OUTB2_SRC_CTRL1 , 0x5800}, + {RT5677_DSP_OUTB2_SRC_CTRL2 , 0x0000}, + {RT5677_DSP_OUTB2_SRC_CTRL3 , 0x0000}, + {RT5677_DSP_OUTB2_SRC_CTRL4 , 0x0800}, + {RT5677_DSP_OUTB_0123_MIXER_CTRL, 0xfefe}, + {RT5677_DSP_OUTB_45_MIXER_CTRL , 0xfefe}, + {RT5677_DSP_OUTB_67_MIXER_CTRL , 0xfefe}, + {RT5677_DIG_MISC , 0x0000}, + {RT5677_GEN_CTRL1 , 0x0000}, + {RT5677_GEN_CTRL2 , 0x0000}, + {RT5677_VENDOR_ID , 0x0000}, + {RT5677_VENDOR_ID1 , 0x10ec}, + {RT5677_VENDOR_ID2 , 0x6327}, +}; + +static bool rt5677_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5677_ranges); i++) { + if (reg >= rt5677_ranges[i].range_min && + reg <= rt5677_ranges[i].range_max) { + return true; + } + } + + switch (reg) { + case RT5677_RESET: + case RT5677_SLIMBUS_PARAM: + case RT5677_PDM_DATA_CTRL1: + case RT5677_PDM_DATA_CTRL2: + case RT5677_PDM1_DATA_CTRL4: + case RT5677_PDM2_DATA_CTRL4: + case RT5677_I2C_MASTER_CTRL1: + case RT5677_I2C_MASTER_CTRL7: + case RT5677_I2C_MASTER_CTRL8: + case RT5677_HAP_GENE_CTRL2: + case RT5677_PWR_DSP_ST: + case RT5677_PRIV_DATA: + case RT5677_PLL1_CTRL2: + case RT5677_PLL2_CTRL2: + case RT5677_ASRC_22: + case RT5677_ASRC_23: + case RT5677_VAD_CTRL5: + case RT5677_ADC_EQ_CTRL1: + case RT5677_EQ_CTRL1: + case RT5677_IRQ_CTRL1: + case RT5677_IRQ_CTRL2: + case RT5677_GPIO_ST: + case RT5677_DSP_INB1_SRC_CTRL4: + case RT5677_DSP_INB2_SRC_CTRL4: + case RT5677_DSP_INB3_SRC_CTRL4: + case RT5677_DSP_OUTB1_SRC_CTRL4: + case RT5677_DSP_OUTB2_SRC_CTRL4: + case RT5677_VENDOR_ID: + case RT5677_VENDOR_ID1: + case RT5677_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool rt5677_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5677_ranges); i++) { + if (reg >= rt5677_ranges[i].range_min && + reg <= rt5677_ranges[i].range_max) { + return true; + } + } + + switch (reg) { + case RT5677_RESET: + case RT5677_LOUT1: + case RT5677_IN1: + case RT5677_MICBIAS: + case RT5677_SLIMBUS_PARAM: + case RT5677_SLIMBUS_RX: + case RT5677_SLIMBUS_CTRL: + case RT5677_SIDETONE_CTRL: + case RT5677_ANA_DAC1_2_3_SRC: + case RT5677_IF_DSP_DAC3_4_MIXER: + case RT5677_DAC4_DIG_VOL: + case RT5677_DAC3_DIG_VOL: + case RT5677_DAC1_DIG_VOL: + case RT5677_DAC2_DIG_VOL: + case RT5677_IF_DSP_DAC2_MIXER: + case RT5677_STO1_ADC_DIG_VOL: + case RT5677_MONO_ADC_DIG_VOL: + case RT5677_STO1_2_ADC_BST: + case RT5677_STO2_ADC_DIG_VOL: + case RT5677_ADC_BST_CTRL2: + case RT5677_STO3_4_ADC_BST: + case RT5677_STO3_ADC_DIG_VOL: + case RT5677_STO4_ADC_DIG_VOL: + case RT5677_STO4_ADC_MIXER: + case RT5677_STO3_ADC_MIXER: + case RT5677_STO2_ADC_MIXER: + case RT5677_STO1_ADC_MIXER: + case RT5677_MONO_ADC_MIXER: + case RT5677_ADC_IF_DSP_DAC1_MIXER: + case RT5677_STO1_DAC_MIXER: + case RT5677_MONO_DAC_MIXER: + case RT5677_DD1_MIXER: + case RT5677_DD2_MIXER: + case RT5677_IF3_DATA: + case RT5677_IF4_DATA: + case RT5677_PDM_OUT_CTRL: + case RT5677_PDM_DATA_CTRL1: + case RT5677_PDM_DATA_CTRL2: + case RT5677_PDM1_DATA_CTRL2: + case RT5677_PDM1_DATA_CTRL3: + case RT5677_PDM1_DATA_CTRL4: + case RT5677_PDM2_DATA_CTRL2: + case RT5677_PDM2_DATA_CTRL3: + case RT5677_PDM2_DATA_CTRL4: + case RT5677_TDM1_CTRL1: + case RT5677_TDM1_CTRL2: + case RT5677_TDM1_CTRL3: + case RT5677_TDM1_CTRL4: + case RT5677_TDM1_CTRL5: + case RT5677_TDM2_CTRL1: + case RT5677_TDM2_CTRL2: + case RT5677_TDM2_CTRL3: + case RT5677_TDM2_CTRL4: + case RT5677_TDM2_CTRL5: + case RT5677_I2C_MASTER_CTRL1: + case RT5677_I2C_MASTER_CTRL2: + case RT5677_I2C_MASTER_CTRL3: + case RT5677_I2C_MASTER_CTRL4: + case RT5677_I2C_MASTER_CTRL5: + case RT5677_I2C_MASTER_CTRL6: + case RT5677_I2C_MASTER_CTRL7: + case RT5677_I2C_MASTER_CTRL8: + case RT5677_DMIC_CTRL1: + case RT5677_DMIC_CTRL2: + case RT5677_HAP_GENE_CTRL1: + case RT5677_HAP_GENE_CTRL2: + case RT5677_HAP_GENE_CTRL3: + case RT5677_HAP_GENE_CTRL4: + case RT5677_HAP_GENE_CTRL5: + case RT5677_HAP_GENE_CTRL6: + case RT5677_HAP_GENE_CTRL7: + case RT5677_HAP_GENE_CTRL8: + case RT5677_HAP_GENE_CTRL9: + case RT5677_HAP_GENE_CTRL10: + case RT5677_PWR_DIG1: + case RT5677_PWR_DIG2: + case RT5677_PWR_ANLG1: + case RT5677_PWR_ANLG2: + case RT5677_PWR_DSP1: + case RT5677_PWR_DSP_ST: + case RT5677_PWR_DSP2: + case RT5677_ADC_DAC_HPF_CTRL1: + case RT5677_PRIV_INDEX: + case RT5677_PRIV_DATA: + case RT5677_I2S4_SDP: + case RT5677_I2S1_SDP: + case RT5677_I2S2_SDP: + case RT5677_I2S3_SDP: + case RT5677_CLK_TREE_CTRL1: + case RT5677_CLK_TREE_CTRL2: + case RT5677_CLK_TREE_CTRL3: + case RT5677_PLL1_CTRL1: + case RT5677_PLL1_CTRL2: + case RT5677_PLL2_CTRL1: + case RT5677_PLL2_CTRL2: + case RT5677_GLB_CLK1: + case RT5677_GLB_CLK2: + case RT5677_ASRC_1: + case RT5677_ASRC_2: + case RT5677_ASRC_3: + case RT5677_ASRC_4: + case RT5677_ASRC_5: + case RT5677_ASRC_6: + case RT5677_ASRC_7: + case RT5677_ASRC_8: + case RT5677_ASRC_9: + case RT5677_ASRC_10: + case RT5677_ASRC_11: + case RT5677_ASRC_12: + case RT5677_ASRC_13: + case RT5677_ASRC_14: + case RT5677_ASRC_15: + case RT5677_ASRC_16: + case RT5677_ASRC_17: + case RT5677_ASRC_18: + case RT5677_ASRC_19: + case RT5677_ASRC_20: + case RT5677_ASRC_21: + case RT5677_ASRC_22: + case RT5677_ASRC_23: + case RT5677_VAD_CTRL1: + case RT5677_VAD_CTRL2: + case RT5677_VAD_CTRL3: + case RT5677_VAD_CTRL4: + case RT5677_VAD_CTRL5: + case RT5677_DSP_INB_CTRL1: + case RT5677_DSP_INB_CTRL2: + case RT5677_DSP_IN_OUTB_CTRL: + case RT5677_DSP_OUTB0_1_DIG_VOL: + case RT5677_DSP_OUTB2_3_DIG_VOL: + case RT5677_DSP_OUTB4_5_DIG_VOL: + case RT5677_DSP_OUTB6_7_DIG_VOL: + case RT5677_ADC_EQ_CTRL1: + case RT5677_ADC_EQ_CTRL2: + case RT5677_EQ_CTRL1: + case RT5677_EQ_CTRL2: + case RT5677_EQ_CTRL3: + case RT5677_SOFT_VOL_ZERO_CROSS1: + case RT5677_JD_CTRL1: + case RT5677_JD_CTRL2: + case RT5677_JD_CTRL3: + case RT5677_IRQ_CTRL1: + case RT5677_IRQ_CTRL2: + case RT5677_GPIO_ST: + case RT5677_GPIO_CTRL1: + case RT5677_GPIO_CTRL2: + case RT5677_GPIO_CTRL3: + case RT5677_STO1_ADC_HI_FILTER1: + case RT5677_STO1_ADC_HI_FILTER2: + case RT5677_MONO_ADC_HI_FILTER1: + case RT5677_MONO_ADC_HI_FILTER2: + case RT5677_STO2_ADC_HI_FILTER1: + case RT5677_STO2_ADC_HI_FILTER2: + case RT5677_STO3_ADC_HI_FILTER1: + case RT5677_STO3_ADC_HI_FILTER2: + case RT5677_STO4_ADC_HI_FILTER1: + case RT5677_STO4_ADC_HI_FILTER2: + case RT5677_MB_DRC_CTRL1: + case RT5677_DRC1_CTRL1: + case RT5677_DRC1_CTRL2: + case RT5677_DRC1_CTRL3: + case RT5677_DRC1_CTRL4: + case RT5677_DRC1_CTRL5: + case RT5677_DRC1_CTRL6: + case RT5677_DRC2_CTRL1: + case RT5677_DRC2_CTRL2: + case RT5677_DRC2_CTRL3: + case RT5677_DRC2_CTRL4: + case RT5677_DRC2_CTRL5: + case RT5677_DRC2_CTRL6: + case RT5677_DRC1_HL_CTRL1: + case RT5677_DRC1_HL_CTRL2: + case RT5677_DRC2_HL_CTRL1: + case RT5677_DRC2_HL_CTRL2: + case RT5677_DSP_INB1_SRC_CTRL1: + case RT5677_DSP_INB1_SRC_CTRL2: + case RT5677_DSP_INB1_SRC_CTRL3: + case RT5677_DSP_INB1_SRC_CTRL4: + case RT5677_DSP_INB2_SRC_CTRL1: + case RT5677_DSP_INB2_SRC_CTRL2: + case RT5677_DSP_INB2_SRC_CTRL3: + case RT5677_DSP_INB2_SRC_CTRL4: + case RT5677_DSP_INB3_SRC_CTRL1: + case RT5677_DSP_INB3_SRC_CTRL2: + case RT5677_DSP_INB3_SRC_CTRL3: + case RT5677_DSP_INB3_SRC_CTRL4: + case RT5677_DSP_OUTB1_SRC_CTRL1: + case RT5677_DSP_OUTB1_SRC_CTRL2: + case RT5677_DSP_OUTB1_SRC_CTRL3: + case RT5677_DSP_OUTB1_SRC_CTRL4: + case RT5677_DSP_OUTB2_SRC_CTRL1: + case RT5677_DSP_OUTB2_SRC_CTRL2: + case RT5677_DSP_OUTB2_SRC_CTRL3: + case RT5677_DSP_OUTB2_SRC_CTRL4: + case RT5677_DSP_OUTB_0123_MIXER_CTRL: + case RT5677_DSP_OUTB_45_MIXER_CTRL: + case RT5677_DSP_OUTB_67_MIXER_CTRL: + case RT5677_DIG_MISC: + case RT5677_GEN_CTRL1: + case RT5677_GEN_CTRL2: + case RT5677_VENDOR_ID: + case RT5677_VENDOR_ID1: + case RT5677_VENDOR_ID2: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static unsigned int bst_tlv[] = { + TLV_DB_RANGE_HEAD(7), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), +}; + +static const struct snd_kcontrol_new rt5677_snd_controls[] = { + /* OUTPUT Control */ + SOC_SINGLE("OUT1 Playback Switch", RT5677_LOUT1, + RT5677_LOUT1_L_MUTE_SFT, 1, 1), + SOC_SINGLE("OUT2 Playback Switch", RT5677_LOUT1, + RT5677_LOUT2_L_MUTE_SFT, 1, 1), + SOC_SINGLE("OUT3 Playback Switch", RT5677_LOUT1, + RT5677_LOUT3_L_MUTE_SFT, 1, 1), + + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5677_DAC1_DIG_VOL, + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + SOC_DOUBLE_TLV("DAC2 Playback Volume", RT5677_DAC2_DIG_VOL, + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + SOC_DOUBLE_TLV("DAC3 Playback Volume", RT5677_DAC3_DIG_VOL, + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + SOC_DOUBLE_TLV("DAC4 Playback Volume", RT5677_DAC4_DIG_VOL, + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + + /* IN1/IN2 Control */ + SOC_SINGLE_TLV("IN1 Boost", RT5677_IN1, RT5677_BST_SFT1, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN2 Boost", RT5677_IN1, RT5677_BST_SFT2, 8, 0, bst_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC1 Capture Switch", RT5677_STO1_ADC_DIG_VOL, + RT5677_L_MUTE_SFT, RT5677_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("ADC2 Capture Switch", RT5677_STO2_ADC_DIG_VOL, + RT5677_L_MUTE_SFT, RT5677_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("ADC3 Capture Switch", RT5677_STO3_ADC_DIG_VOL, + RT5677_L_MUTE_SFT, RT5677_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("ADC4 Capture Switch", RT5677_STO4_ADC_DIG_VOL, + RT5677_L_MUTE_SFT, RT5677_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("Mono ADC Capture Switch", RT5677_MONO_ADC_DIG_VOL, + RT5677_L_MUTE_SFT, RT5677_R_MUTE_SFT, 1, 1), + + SOC_DOUBLE_TLV("ADC1 Capture Volume", RT5677_STO1_ADC_DIG_VOL, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("ADC2 Capture Volume", RT5677_STO2_ADC_DIG_VOL, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("ADC3 Capture Volume", RT5677_STO3_ADC_DIG_VOL, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("ADC4 Capture Volume", RT5677_STO4_ADC_DIG_VOL, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5677_MONO_ADC_DIG_VOL, + RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 127, 0, + adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5677_STO1_2_ADC_BST, + RT5677_STO1_ADC_L_BST_SFT, RT5677_STO1_ADC_R_BST_SFT, 3, 0, + adc_bst_tlv), + SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5677_STO1_2_ADC_BST, + RT5677_STO2_ADC_L_BST_SFT, RT5677_STO2_ADC_R_BST_SFT, 3, 0, + adc_bst_tlv), + SOC_DOUBLE_TLV("STO3 ADC Boost Gain", RT5677_STO3_4_ADC_BST, + RT5677_STO3_ADC_L_BST_SFT, RT5677_STO3_ADC_R_BST_SFT, 3, 0, + adc_bst_tlv), + SOC_DOUBLE_TLV("STO4 ADC Boost Gain", RT5677_STO3_4_ADC_BST, + RT5677_STO4_ADC_L_BST_SFT, RT5677_STO4_ADC_R_BST_SFT, 3, 0, + adc_bst_tlv), + SOC_DOUBLE_TLV("Mono ADC Boost Gain", RT5677_ADC_BST_CTRL2, + RT5677_MONO_ADC_L_BST_SFT, RT5677_MONO_ADC_R_BST_SFT, 3, 0, + adc_bst_tlv), +}; + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + int div[] = {2, 3, 4, 6, 8, 12}, idx = -EINVAL, i; + int rate, red, bound, temp; + + rate = rt5677->sysclk; + red = 3000000 * 12; + for (i = 0; i < ARRAY_SIZE(div); i++) { + bound = div[i] * 3000000; + if (rate > bound) + continue; + temp = bound - rate; + if (temp < red) { + red = temp; + idx = i; + } + } + + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + regmap_update_bits(rt5677->regmap, RT5677_DMIC_CTRL1, + RT5677_DMIC_CLK_MASK, idx << RT5677_DMIC_CLK_SFT); + return idx; +} + +static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(source->codec); + unsigned int val; + + regmap_read(rt5677->regmap, RT5677_GLB_CLK1, &val); + val &= RT5677_SCLK_SRC_MASK; + if (val == RT5677_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5677_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO1_ADC_MIXER, + RT5677_M_STO1_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5677_STO1_ADC_MIXER, + RT5677_M_STO1_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO1_ADC_MIXER, + RT5677_M_STO1_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5677_STO1_ADC_MIXER, + RT5677_M_STO1_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_sto2_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO2_ADC_MIXER, + RT5677_M_STO2_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5677_STO2_ADC_MIXER, + RT5677_M_STO2_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_sto2_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO2_ADC_MIXER, + RT5677_M_STO2_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5677_STO2_ADC_MIXER, + RT5677_M_STO2_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_sto3_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO3_ADC_MIXER, + RT5677_M_STO3_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5677_STO3_ADC_MIXER, + RT5677_M_STO3_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_sto3_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO3_ADC_MIXER, + RT5677_M_STO3_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5677_STO3_ADC_MIXER, + RT5677_M_STO3_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_sto4_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO4_ADC_MIXER, + RT5677_M_STO4_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5677_STO4_ADC_MIXER, + RT5677_M_STO4_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_sto4_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO4_ADC_MIXER, + RT5677_M_STO4_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5677_STO4_ADC_MIXER, + RT5677_M_STO4_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_mono_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5677_MONO_ADC_MIXER, + RT5677_M_MONO_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5677_MONO_ADC_MIXER, + RT5677_M_MONO_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_mono_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5677_MONO_ADC_MIXER, + RT5677_M_MONO_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5677_MONO_ADC_MIXER, + RT5677_M_MONO_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5677_ADC_IF_DSP_DAC1_MIXER, + RT5677_M_ADDA_MIXER1_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5677_ADC_IF_DSP_DAC1_MIXER, + RT5677_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5677_ADC_IF_DSP_DAC1_MIXER, + RT5677_M_ADDA_MIXER1_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5677_ADC_IF_DSP_DAC1_MIXER, + RT5677_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_sto1_dac_l_mix[] = { + SOC_DAPM_SINGLE("ST L Switch", RT5677_STO1_DAC_MIXER, + RT5677_M_ST_DAC1_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER, + RT5677_M_DAC1_L_STO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_STO1_DAC_MIXER, + RT5677_M_DAC2_L_STO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER, + RT5677_M_DAC1_R_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_sto1_dac_r_mix[] = { + SOC_DAPM_SINGLE("ST R Switch", RT5677_STO1_DAC_MIXER, + RT5677_M_ST_DAC1_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER, + RT5677_M_DAC1_R_STO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_STO1_DAC_MIXER, + RT5677_M_DAC2_R_STO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER, + RT5677_M_DAC1_L_STO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_mono_dac_l_mix[] = { + SOC_DAPM_SINGLE("ST L Switch", RT5677_MONO_DAC_MIXER, + RT5677_M_ST_DAC2_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_MONO_DAC_MIXER, + RT5677_M_DAC1_L_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER, + RT5677_M_DAC2_L_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER, + RT5677_M_DAC2_R_MONO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_mono_dac_r_mix[] = { + SOC_DAPM_SINGLE("ST R Switch", RT5677_MONO_DAC_MIXER, + RT5677_M_ST_DAC2_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_MONO_DAC_MIXER, + RT5677_M_DAC1_R_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER, + RT5677_M_DAC2_R_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER, + RT5677_M_DAC2_L_MONO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_dd1_l_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5677_DD1_MIXER, + RT5677_M_STO_L_DD1_L_SFT, 1, 1), + SOC_DAPM_SINGLE("Mono DAC Mix L Switch", RT5677_DD1_MIXER, + RT5677_M_MONO_L_DD1_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC3 L Switch", RT5677_DD1_MIXER, + RT5677_M_DAC3_L_DD1_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC3 R Switch", RT5677_DD1_MIXER, + RT5677_M_DAC3_R_DD1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_dd1_r_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5677_DD1_MIXER, + RT5677_M_STO_R_DD1_R_SFT, 1, 1), + SOC_DAPM_SINGLE("Mono DAC Mix R Switch", RT5677_DD1_MIXER, + RT5677_M_MONO_R_DD1_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC3 R Switch", RT5677_DD1_MIXER, + RT5677_M_DAC3_R_DD1_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC3 L Switch", RT5677_DD1_MIXER, + RT5677_M_DAC3_L_DD1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_dd2_l_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5677_DD2_MIXER, + RT5677_M_STO_L_DD2_L_SFT, 1, 1), + SOC_DAPM_SINGLE("Mono DAC Mix L Switch", RT5677_DD2_MIXER, + RT5677_M_MONO_L_DD2_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC4 L Switch", RT5677_DD2_MIXER, + RT5677_M_DAC4_L_DD2_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC4 R Switch", RT5677_DD2_MIXER, + RT5677_M_DAC4_R_DD2_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_dd2_r_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5677_DD2_MIXER, + RT5677_M_STO_R_DD2_R_SFT, 1, 1), + SOC_DAPM_SINGLE("Mono DAC Mix R Switch", RT5677_DD2_MIXER, + RT5677_M_MONO_R_DD2_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC4 R Switch", RT5677_DD2_MIXER, + RT5677_M_DAC4_R_DD2_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC4 L Switch", RT5677_DD2_MIXER, + RT5677_M_DAC4_L_DD2_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_ob_01_mix[] = { + SOC_DAPM_SINGLE("IB01 Switch", RT5677_DSP_OUTB_0123_MIXER_CTRL, + RT5677_DSP_IB_01_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB23 Switch", RT5677_DSP_OUTB_0123_MIXER_CTRL, + RT5677_DSP_IB_23_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB45 Switch", RT5677_DSP_OUTB_0123_MIXER_CTRL, + RT5677_DSP_IB_45_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB6 Switch", RT5677_DSP_OUTB_0123_MIXER_CTRL, + RT5677_DSP_IB_6_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB7 Switch", RT5677_DSP_OUTB_0123_MIXER_CTRL, + RT5677_DSP_IB_7_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB8 Switch", RT5677_DSP_OUTB_0123_MIXER_CTRL, + RT5677_DSP_IB_8_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB9 Switch", RT5677_DSP_OUTB_0123_MIXER_CTRL, + RT5677_DSP_IB_9_H_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_ob_23_mix[] = { + SOC_DAPM_SINGLE("IB01 Switch", RT5677_DSP_OUTB_0123_MIXER_CTRL, + RT5677_DSP_IB_01_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB23 Switch", RT5677_DSP_OUTB_0123_MIXER_CTRL, + RT5677_DSP_IB_23_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB45 Switch", RT5677_DSP_OUTB_0123_MIXER_CTRL, + RT5677_DSP_IB_45_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB6 Switch", RT5677_DSP_OUTB_0123_MIXER_CTRL, + RT5677_DSP_IB_6_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB7 Switch", RT5677_DSP_OUTB_0123_MIXER_CTRL, + RT5677_DSP_IB_7_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB8 Switch", RT5677_DSP_OUTB_0123_MIXER_CTRL, + RT5677_DSP_IB_8_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB9 Switch", RT5677_DSP_OUTB_0123_MIXER_CTRL, + RT5677_DSP_IB_9_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_ob_4_mix[] = { + SOC_DAPM_SINGLE("IB01 Switch", RT5677_DSP_OUTB_45_MIXER_CTRL, + RT5677_DSP_IB_01_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB23 Switch", RT5677_DSP_OUTB_45_MIXER_CTRL, + RT5677_DSP_IB_23_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB45 Switch", RT5677_DSP_OUTB_45_MIXER_CTRL, + RT5677_DSP_IB_45_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB6 Switch", RT5677_DSP_OUTB_45_MIXER_CTRL, + RT5677_DSP_IB_6_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB7 Switch", RT5677_DSP_OUTB_45_MIXER_CTRL, + RT5677_DSP_IB_7_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB8 Switch", RT5677_DSP_OUTB_45_MIXER_CTRL, + RT5677_DSP_IB_8_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB9 Switch", RT5677_DSP_OUTB_45_MIXER_CTRL, + RT5677_DSP_IB_9_H_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_ob_5_mix[] = { + SOC_DAPM_SINGLE("IB01 Switch", RT5677_DSP_OUTB_45_MIXER_CTRL, + RT5677_DSP_IB_01_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB23 Switch", RT5677_DSP_OUTB_45_MIXER_CTRL, + RT5677_DSP_IB_23_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB45 Switch", RT5677_DSP_OUTB_45_MIXER_CTRL, + RT5677_DSP_IB_45_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB6 Switch", RT5677_DSP_OUTB_45_MIXER_CTRL, + RT5677_DSP_IB_6_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB7 Switch", RT5677_DSP_OUTB_45_MIXER_CTRL, + RT5677_DSP_IB_7_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB8 Switch", RT5677_DSP_OUTB_45_MIXER_CTRL, + RT5677_DSP_IB_8_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB9 Switch", RT5677_DSP_OUTB_45_MIXER_CTRL, + RT5677_DSP_IB_9_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_ob_6_mix[] = { + SOC_DAPM_SINGLE("IB01 Switch", RT5677_DSP_OUTB_67_MIXER_CTRL, + RT5677_DSP_IB_01_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB23 Switch", RT5677_DSP_OUTB_67_MIXER_CTRL, + RT5677_DSP_IB_23_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB45 Switch", RT5677_DSP_OUTB_67_MIXER_CTRL, + RT5677_DSP_IB_45_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB6 Switch", RT5677_DSP_OUTB_67_MIXER_CTRL, + RT5677_DSP_IB_6_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB7 Switch", RT5677_DSP_OUTB_67_MIXER_CTRL, + RT5677_DSP_IB_7_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB8 Switch", RT5677_DSP_OUTB_67_MIXER_CTRL, + RT5677_DSP_IB_8_H_SFT, 1, 1), + SOC_DAPM_SINGLE("IB9 Switch", RT5677_DSP_OUTB_67_MIXER_CTRL, + RT5677_DSP_IB_9_H_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5677_ob_7_mix[] = { + SOC_DAPM_SINGLE("IB01 Switch", RT5677_DSP_OUTB_67_MIXER_CTRL, + RT5677_DSP_IB_01_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB23 Switch", RT5677_DSP_OUTB_67_MIXER_CTRL, + RT5677_DSP_IB_23_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB45 Switch", RT5677_DSP_OUTB_67_MIXER_CTRL, + RT5677_DSP_IB_45_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB6 Switch", RT5677_DSP_OUTB_67_MIXER_CTRL, + RT5677_DSP_IB_6_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB7 Switch", RT5677_DSP_OUTB_67_MIXER_CTRL, + RT5677_DSP_IB_7_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB8 Switch", RT5677_DSP_OUTB_67_MIXER_CTRL, + RT5677_DSP_IB_8_L_SFT, 1, 1), + SOC_DAPM_SINGLE("IB9 Switch", RT5677_DSP_OUTB_67_MIXER_CTRL, + RT5677_DSP_IB_9_L_SFT, 1, 1), +}; + + +/* Mux */ +/* DAC1 L/R source */ /* MX-29 [10:8] */ +static const char * const rt5677_dac1_src[] = { + "IF1 DAC 01", "IF2 DAC 01", "IF3 DAC LR", "IF4 DAC LR", "SLB DAC 01", + "OB 01" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_dac1_enum, RT5677_ADC_IF_DSP_DAC1_MIXER, + RT5677_DAC1_L_SEL_SFT, rt5677_dac1_src); + +static const struct snd_kcontrol_new rt5677_dac1_mux = + SOC_DAPM_ENUM("DAC1 source", rt5677_dac1_enum); + +/* ADDA1 L/R source */ /* MX-29 [1:0] */ +static const char * const rt5677_adda1_src[] = { + "STO1 ADC MIX", "STO2 ADC MIX", "OB 67", +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_adda1_enum, RT5677_ADC_IF_DSP_DAC1_MIXER, + RT5677_ADDA1_SEL_SFT, rt5677_adda1_src); + +static const struct snd_kcontrol_new rt5677_adda1_mux = + SOC_DAPM_ENUM("ADDA1 source", rt5677_adda1_enum); + + +/*DAC2 L/R source*/ /* MX-1B [6:4] [2:0] */ +static const char * const rt5677_dac2l_src[] = { + "IF1 DAC 2", "IF2 DAC 2", "IF3 DAC L", "IF4 DAC L", "SLB DAC 2", + "OB 2", +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_dac2l_enum, RT5677_IF_DSP_DAC2_MIXER, + RT5677_SEL_DAC2_L_SRC_SFT, rt5677_dac2l_src); + +static const struct snd_kcontrol_new rt5677_dac2_l_mux = + SOC_DAPM_ENUM("DAC2 L source", rt5677_dac2l_enum); + +static const char * const rt5677_dac2r_src[] = { + "IF1 DAC 3", "IF2 DAC 3", "IF3 DAC R", "IF4 DAC R", "SLB DAC 3", + "OB 3", "Haptic Generator", "VAD ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_dac2r_enum, RT5677_IF_DSP_DAC2_MIXER, + RT5677_SEL_DAC2_R_SRC_SFT, rt5677_dac2r_src); + +static const struct snd_kcontrol_new rt5677_dac2_r_mux = + SOC_DAPM_ENUM("DAC2 R source", rt5677_dac2r_enum); + +/*DAC3 L/R source*/ /* MX-16 [6:4] [2:0] */ +static const char * const rt5677_dac3l_src[] = { + "IF1 DAC 4", "IF2 DAC 4", "IF3 DAC L", "IF4 DAC L", + "SLB DAC 4", "OB 4" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_dac3l_enum, RT5677_IF_DSP_DAC3_4_MIXER, + RT5677_SEL_DAC3_L_SRC_SFT, rt5677_dac3l_src); + +static const struct snd_kcontrol_new rt5677_dac3_l_mux = + SOC_DAPM_ENUM("DAC3 L source", rt5677_dac3l_enum); + +static const char * const rt5677_dac3r_src[] = { + "IF1 DAC 5", "IF2 DAC 5", "IF3 DAC R", "IF4 DAC R", + "SLB DAC 5", "OB 5" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_dac3r_enum, RT5677_IF_DSP_DAC3_4_MIXER, + RT5677_SEL_DAC3_R_SRC_SFT, rt5677_dac3r_src); + +static const struct snd_kcontrol_new rt5677_dac3_r_mux = + SOC_DAPM_ENUM("DAC3 R source", rt5677_dac3r_enum); + +/*DAC4 L/R source*/ /* MX-16 [14:12] [10:8] */ +static const char * const rt5677_dac4l_src[] = { + "IF1 DAC 6", "IF2 DAC 6", "IF3 DAC L", "IF4 DAC L", + "SLB DAC 6", "OB 6" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_dac4l_enum, RT5677_IF_DSP_DAC3_4_MIXER, + RT5677_SEL_DAC4_L_SRC_SFT, rt5677_dac4l_src); + +static const struct snd_kcontrol_new rt5677_dac4_l_mux = + SOC_DAPM_ENUM("DAC4 L source", rt5677_dac4l_enum); + +static const char * const rt5677_dac4r_src[] = { + "IF1 DAC 7", "IF2 DAC 7", "IF3 DAC R", "IF4 DAC R", + "SLB DAC 7", "OB 7" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_dac4r_enum, RT5677_IF_DSP_DAC3_4_MIXER, + RT5677_SEL_DAC4_R_SRC_SFT, rt5677_dac4r_src); + +static const struct snd_kcontrol_new rt5677_dac4_r_mux = + SOC_DAPM_ENUM("DAC4 R source", rt5677_dac4r_enum); + +/* In/OutBound Source Pass SRC */ /* MX-A5 [3] [4] [0] [1] [2] */ +static const char * const rt5677_iob_bypass_src[] = { + "Bypass", "Pass SRC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_ob01_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL, + RT5677_SEL_SRC_OB01_SFT, rt5677_iob_bypass_src); + +static const struct snd_kcontrol_new rt5677_ob01_bypass_src_mux = + SOC_DAPM_ENUM("OB01 Bypass source", rt5677_ob01_bypass_src_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_ob23_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL, + RT5677_SEL_SRC_OB23_SFT, rt5677_iob_bypass_src); + +static const struct snd_kcontrol_new rt5677_ob23_bypass_src_mux = + SOC_DAPM_ENUM("OB23 Bypass source", rt5677_ob23_bypass_src_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_ib01_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL, + RT5677_SEL_SRC_IB01_SFT, rt5677_iob_bypass_src); + +static const struct snd_kcontrol_new rt5677_ib01_bypass_src_mux = + SOC_DAPM_ENUM("IB01 Bypass source", rt5677_ib01_bypass_src_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_ib23_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL, + RT5677_SEL_SRC_IB23_SFT, rt5677_iob_bypass_src); + +static const struct snd_kcontrol_new rt5677_ib23_bypass_src_mux = + SOC_DAPM_ENUM("IB23 Bypass source", rt5677_ib23_bypass_src_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_ib45_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL, + RT5677_SEL_SRC_IB45_SFT, rt5677_iob_bypass_src); + +static const struct snd_kcontrol_new rt5677_ib45_bypass_src_mux = + SOC_DAPM_ENUM("IB45 Bypass source", rt5677_ib45_bypass_src_enum); + +/* Stereo ADC Source 2 */ /* MX-27 MX26 MX25 [11:10] */ +static const char * const rt5677_stereo_adc2_src[] = { + "DD MIX1", "DMIC", "Stereo DAC MIX" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_stereo1_adc2_enum, RT5677_STO1_ADC_MIXER, + RT5677_SEL_STO1_ADC2_SFT, rt5677_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5677_sto1_adc2_mux = + SOC_DAPM_ENUM("Stereo1 ADC2 source", rt5677_stereo1_adc2_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_stereo2_adc2_enum, RT5677_STO2_ADC_MIXER, + RT5677_SEL_STO2_ADC2_SFT, rt5677_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5677_sto2_adc2_mux = + SOC_DAPM_ENUM("Stereo2 ADC2 source", rt5677_stereo2_adc2_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_stereo3_adc2_enum, RT5677_STO3_ADC_MIXER, + RT5677_SEL_STO3_ADC2_SFT, rt5677_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5677_sto3_adc2_mux = + SOC_DAPM_ENUM("Stereo3 ADC2 source", rt5677_stereo3_adc2_enum); + +/* DMIC Source */ /* MX-28 [9:8][1:0] MX-27 MX-26 MX-25 MX-24 [9:8] */ +static const char * const rt5677_dmic_src[] = { + "DMIC1", "DMIC2", "DMIC3", "DMIC4" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_mono_dmic_l_enum, RT5677_MONO_ADC_MIXER, + RT5677_SEL_MONO_DMIC_L_SFT, rt5677_dmic_src); + +static const struct snd_kcontrol_new rt5677_mono_dmic_l_mux = + SOC_DAPM_ENUM("Mono DMIC L source", rt5677_mono_dmic_l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_mono_dmic_r_enum, RT5677_MONO_ADC_MIXER, + RT5677_SEL_MONO_DMIC_R_SFT, rt5677_dmic_src); + +static const struct snd_kcontrol_new rt5677_mono_dmic_r_mux = + SOC_DAPM_ENUM("Mono DMIC R source", rt5677_mono_dmic_r_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_stereo1_dmic_enum, RT5677_STO1_ADC_MIXER, + RT5677_SEL_STO1_DMIC_SFT, rt5677_dmic_src); + +static const struct snd_kcontrol_new rt5677_sto1_dmic_mux = + SOC_DAPM_ENUM("Stereo1 DMIC source", rt5677_stereo1_dmic_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_stereo2_dmic_enum, RT5677_STO2_ADC_MIXER, + RT5677_SEL_STO2_DMIC_SFT, rt5677_dmic_src); + +static const struct snd_kcontrol_new rt5677_sto2_dmic_mux = + SOC_DAPM_ENUM("Stereo2 DMIC source", rt5677_stereo2_dmic_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_stereo3_dmic_enum, RT5677_STO3_ADC_MIXER, + RT5677_SEL_STO3_DMIC_SFT, rt5677_dmic_src); + +static const struct snd_kcontrol_new rt5677_sto3_dmic_mux = + SOC_DAPM_ENUM("Stereo3 DMIC source", rt5677_stereo3_dmic_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_stereo4_dmic_enum, RT5677_STO4_ADC_MIXER, + RT5677_SEL_STO4_DMIC_SFT, rt5677_dmic_src); + +static const struct snd_kcontrol_new rt5677_sto4_dmic_mux = + SOC_DAPM_ENUM("Stereo4 DMIC source", rt5677_stereo4_dmic_enum); + +/* Stereo2 ADC source */ /* MX-26 [0] */ +static const char * const rt5677_stereo2_adc_lr_src[] = { + "L", "LR" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_stereo2_adc_lr_enum, RT5677_STO2_ADC_MIXER, + RT5677_SEL_STO2_LR_MIX_SFT, rt5677_stereo2_adc_lr_src); + +static const struct snd_kcontrol_new rt5677_sto2_adc_lr_mux = + SOC_DAPM_ENUM("Stereo2 ADC LR source", rt5677_stereo2_adc_lr_enum); + +/* Stereo1 ADC Source 1 */ /* MX-27 MX26 MX25 [13:12] */ +static const char * const rt5677_stereo_adc1_src[] = { + "DD MIX1", "ADC1/2", "Stereo DAC MIX" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_stereo1_adc1_enum, RT5677_STO1_ADC_MIXER, + RT5677_SEL_STO1_ADC1_SFT, rt5677_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5677_sto1_adc1_mux = + SOC_DAPM_ENUM("Stereo1 ADC1 source", rt5677_stereo1_adc1_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_stereo2_adc1_enum, RT5677_STO2_ADC_MIXER, + RT5677_SEL_STO2_ADC1_SFT, rt5677_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5677_sto2_adc1_mux = + SOC_DAPM_ENUM("Stereo2 ADC1 source", rt5677_stereo2_adc1_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_stereo3_adc1_enum, RT5677_STO3_ADC_MIXER, + RT5677_SEL_STO3_ADC1_SFT, rt5677_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5677_sto3_adc1_mux = + SOC_DAPM_ENUM("Stereo3 ADC1 source", rt5677_stereo3_adc1_enum); + +/* Mono ADC Left source 2 */ /* MX-28 [11:10] */ +static const char * const rt5677_mono_adc2_l_src[] = { + "DD MIX1L", "DMIC", "MONO DAC MIXL" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_mono_adc2_l_enum, RT5677_MONO_ADC_MIXER, + RT5677_SEL_MONO_ADC_L2_SFT, rt5677_mono_adc2_l_src); + +static const struct snd_kcontrol_new rt5677_mono_adc2_l_mux = + SOC_DAPM_ENUM("Mono ADC2 L source", rt5677_mono_adc2_l_enum); + +/* Mono ADC Left source 1 */ /* MX-28 [13:12] */ +static const char * const rt5677_mono_adc1_l_src[] = { + "DD MIX1L", "ADC1", "MONO DAC MIXL" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_mono_adc1_l_enum, RT5677_MONO_ADC_MIXER, + RT5677_SEL_MONO_ADC_L1_SFT, rt5677_mono_adc1_l_src); + +static const struct snd_kcontrol_new rt5677_mono_adc1_l_mux = + SOC_DAPM_ENUM("Mono ADC1 L source", rt5677_mono_adc1_l_enum); + +/* Mono ADC Right source 2 */ /* MX-28 [3:2] */ +static const char * const rt5677_mono_adc2_r_src[] = { + "DD MIX1R", "DMIC", "MONO DAC MIXR" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_mono_adc2_r_enum, RT5677_MONO_ADC_MIXER, + RT5677_SEL_MONO_ADC_R2_SFT, rt5677_mono_adc2_r_src); + +static const struct snd_kcontrol_new rt5677_mono_adc2_r_mux = + SOC_DAPM_ENUM("Mono ADC2 R source", rt5677_mono_adc2_r_enum); + +/* Mono ADC Right source 1 */ /* MX-28 [5:4] */ +static const char * const rt5677_mono_adc1_r_src[] = { + "DD MIX1R", "ADC2", "MONO DAC MIXR" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_mono_adc1_r_enum, RT5677_MONO_ADC_MIXER, + RT5677_SEL_MONO_ADC_R1_SFT, rt5677_mono_adc1_r_src); + +static const struct snd_kcontrol_new rt5677_mono_adc1_r_mux = + SOC_DAPM_ENUM("Mono ADC1 R source", rt5677_mono_adc1_r_enum); + +/* Stereo4 ADC Source 2 */ /* MX-24 [11:10] */ +static const char * const rt5677_stereo4_adc2_src[] = { + "DD MIX1", "DMIC", "DD MIX2" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_stereo4_adc2_enum, RT5677_STO4_ADC_MIXER, + RT5677_SEL_STO4_ADC2_SFT, rt5677_stereo4_adc2_src); + +static const struct snd_kcontrol_new rt5677_sto4_adc2_mux = + SOC_DAPM_ENUM("Stereo4 ADC2 source", rt5677_stereo4_adc2_enum); + + +/* Stereo4 ADC Source 1 */ /* MX-24 [13:12] */ +static const char * const rt5677_stereo4_adc1_src[] = { + "DD MIX1", "ADC1/2", "DD MIX2" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_stereo4_adc1_enum, RT5677_STO4_ADC_MIXER, + RT5677_SEL_STO4_ADC1_SFT, rt5677_stereo4_adc1_src); + +static const struct snd_kcontrol_new rt5677_sto4_adc1_mux = + SOC_DAPM_ENUM("Stereo4 ADC1 source", rt5677_stereo4_adc1_enum); + +/* InBound0/1 Source */ /* MX-A3 [14:12] */ +static const char * const rt5677_inbound01_src[] = { + "IF1 DAC 01", "IF2 DAC 01", "SLB DAC 01", "STO1 ADC MIX", + "VAD ADC/DAC1 FS" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_inbound01_enum, RT5677_DSP_INB_CTRL1, + RT5677_IB01_SRC_SFT, rt5677_inbound01_src); + +static const struct snd_kcontrol_new rt5677_ib01_src_mux = + SOC_DAPM_ENUM("InBound0/1 Source", rt5677_inbound01_enum); + +/* InBound2/3 Source */ /* MX-A3 [10:8] */ +static const char * const rt5677_inbound23_src[] = { + "IF1 DAC 23", "IF2 DAC 23", "SLB DAC 23", "STO2 ADC MIX", + "DAC1 FS", "IF4 DAC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_inbound23_enum, RT5677_DSP_INB_CTRL1, + RT5677_IB23_SRC_SFT, rt5677_inbound23_src); + +static const struct snd_kcontrol_new rt5677_ib23_src_mux = + SOC_DAPM_ENUM("InBound2/3 Source", rt5677_inbound23_enum); + +/* InBound4/5 Source */ /* MX-A3 [6:4] */ +static const char * const rt5677_inbound45_src[] = { + "IF1 DAC 45", "IF2 DAC 45", "SLB DAC 45", "STO3 ADC MIX", + "IF3 DAC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_inbound45_enum, RT5677_DSP_INB_CTRL1, + RT5677_IB45_SRC_SFT, rt5677_inbound45_src); + +static const struct snd_kcontrol_new rt5677_ib45_src_mux = + SOC_DAPM_ENUM("InBound4/5 Source", rt5677_inbound45_enum); + +/* InBound6 Source */ /* MX-A3 [2:0] */ +static const char * const rt5677_inbound6_src[] = { + "IF1 DAC 6", "IF2 DAC 6", "SLB DAC 6", "STO4 ADC MIX L", + "IF4 DAC L", "STO1 ADC MIX L", "STO2 ADC MIX L", "STO3 ADC MIX L" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_inbound6_enum, RT5677_DSP_INB_CTRL1, + RT5677_IB6_SRC_SFT, rt5677_inbound6_src); + +static const struct snd_kcontrol_new rt5677_ib6_src_mux = + SOC_DAPM_ENUM("InBound6 Source", rt5677_inbound6_enum); + +/* InBound7 Source */ /* MX-A4 [14:12] */ +static const char * const rt5677_inbound7_src[] = { + "IF1 DAC 7", "IF2 DAC 7", "SLB DAC 7", "STO4 ADC MIX R", + "IF4 DAC R", "STO1 ADC MIX R", "STO2 ADC MIX R", "STO3 ADC MIX R" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_inbound7_enum, RT5677_DSP_INB_CTRL2, + RT5677_IB7_SRC_SFT, rt5677_inbound7_src); + +static const struct snd_kcontrol_new rt5677_ib7_src_mux = + SOC_DAPM_ENUM("InBound7 Source", rt5677_inbound7_enum); + +/* InBound8 Source */ /* MX-A4 [10:8] */ +static const char * const rt5677_inbound8_src[] = { + "STO1 ADC MIX L", "STO2 ADC MIX L", "STO3 ADC MIX L", "STO4 ADC MIX L", + "MONO ADC MIX L", "DACL1 FS" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_inbound8_enum, RT5677_DSP_INB_CTRL2, + RT5677_IB8_SRC_SFT, rt5677_inbound8_src); + +static const struct snd_kcontrol_new rt5677_ib8_src_mux = + SOC_DAPM_ENUM("InBound8 Source", rt5677_inbound8_enum); + +/* InBound9 Source */ /* MX-A4 [6:4] */ +static const char * const rt5677_inbound9_src[] = { + "STO1 ADC MIX R", "STO2 ADC MIX R", "STO3 ADC MIX R", "STO4 ADC MIX R", + "MONO ADC MIX R", "DACR1 FS", "DAC1 FS" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_inbound9_enum, RT5677_DSP_INB_CTRL2, + RT5677_IB9_SRC_SFT, rt5677_inbound9_src); + +static const struct snd_kcontrol_new rt5677_ib9_src_mux = + SOC_DAPM_ENUM("InBound9 Source", rt5677_inbound9_enum); + +/* VAD Source */ /* MX-9F [6:4] */ +static const char * const rt5677_vad_src[] = { + "STO1 ADC MIX L", "MONO ADC MIX L", "MONO ADC MIX R", "STO2 ADC MIX L", + "STO3 ADC MIX L" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_vad_enum, RT5677_VAD_CTRL4, + RT5677_VAD_SRC_SFT, rt5677_vad_src); + +static const struct snd_kcontrol_new rt5677_vad_src_mux = + SOC_DAPM_ENUM("VAD Source", rt5677_vad_enum); + +/* Sidetone Source */ /* MX-13 [11:9] */ +static const char * const rt5677_sidetone_src[] = { + "DMIC1 L", "DMIC2 L", "DMIC3 L", "DMIC4 L", "ADC1", "ADC2" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_sidetone_enum, RT5677_SIDETONE_CTRL, + RT5677_ST_SEL_SFT, rt5677_sidetone_src); + +static const struct snd_kcontrol_new rt5677_sidetone_mux = + SOC_DAPM_ENUM("Sidetone Source", rt5677_sidetone_enum); + +/* DAC1/2 Source */ /* MX-15 [1:0] */ +static const char * const rt5677_dac12_src[] = { + "STO1 DAC MIX", "MONO DAC MIX", "DD MIX1", "DD MIX2" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_dac12_enum, RT5677_ANA_DAC1_2_3_SRC, + RT5677_ANA_DAC1_2_SRC_SEL_SFT, rt5677_dac12_src); + +static const struct snd_kcontrol_new rt5677_dac12_mux = + SOC_DAPM_ENUM("Analog DAC1/2 Source", rt5677_dac12_enum); + +/* DAC3 Source */ /* MX-15 [5:4] */ +static const char * const rt5677_dac3_src[] = { + "MONO DAC MIXL", "MONO DAC MIXR", "DD MIX1L", "DD MIX2L" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_dac3_enum, RT5677_ANA_DAC1_2_3_SRC, + RT5677_ANA_DAC3_SRC_SEL_SFT, rt5677_dac3_src); + +static const struct snd_kcontrol_new rt5677_dac3_mux = + SOC_DAPM_ENUM("Analog DAC3 Source", rt5677_dac3_enum); + +/* PDM channel source */ /* MX-31 [13:12][9:8][5:4][1:0] */ +static const char * const rt5677_pdm_src[] = { + "STO1 DAC MIX", "MONO DAC MIX", "DD MIX1", "DD MIX2" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_pdm1_l_enum, RT5677_PDM_OUT_CTRL, + RT5677_SEL_PDM1_L_SFT, rt5677_pdm_src); + +static const struct snd_kcontrol_new rt5677_pdm1_l_mux = + SOC_DAPM_ENUM("PDM1 source", rt5677_pdm1_l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_pdm2_l_enum, RT5677_PDM_OUT_CTRL, + RT5677_SEL_PDM2_L_SFT, rt5677_pdm_src); + +static const struct snd_kcontrol_new rt5677_pdm2_l_mux = + SOC_DAPM_ENUM("PDM2 source", rt5677_pdm2_l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_pdm1_r_enum, RT5677_PDM_OUT_CTRL, + RT5677_SEL_PDM1_R_SFT, rt5677_pdm_src); + +static const struct snd_kcontrol_new rt5677_pdm1_r_mux = + SOC_DAPM_ENUM("PDM1 source", rt5677_pdm1_r_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_pdm2_r_enum, RT5677_PDM_OUT_CTRL, + RT5677_SEL_PDM2_R_SFT, rt5677_pdm_src); + +static const struct snd_kcontrol_new rt5677_pdm2_r_mux = + SOC_DAPM_ENUM("PDM2 source", rt5677_pdm2_r_enum); + +/* TDM IF1/2 SLB ADC1 Data Selection */ /* MX-3C MX-41 [5:4] MX-08 [1:0]*/ +static const char * const rt5677_if12_adc1_src[] = { + "STO1 ADC MIX", "OB01", "VAD ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc1_enum, RT5677_TDM1_CTRL2, + RT5677_IF1_ADC1_SFT, rt5677_if12_adc1_src); + +static const struct snd_kcontrol_new rt5677_if1_adc1_mux = + SOC_DAPM_ENUM("IF1 ADC1 source", rt5677_if1_adc1_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc1_enum, RT5677_TDM2_CTRL2, + RT5677_IF2_ADC1_SFT, rt5677_if12_adc1_src); + +static const struct snd_kcontrol_new rt5677_if2_adc1_mux = + SOC_DAPM_ENUM("IF2 ADC1 source", rt5677_if2_adc1_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_slb_adc1_enum, RT5677_SLIMBUS_RX, + RT5677_SLB_ADC1_SFT, rt5677_if12_adc1_src); + +static const struct snd_kcontrol_new rt5677_slb_adc1_mux = + SOC_DAPM_ENUM("SLB ADC1 source", rt5677_slb_adc1_enum); + +/* TDM IF1/2 SLB ADC2 Data Selection */ /* MX-3C MX-41 [7:6] MX-08 [3:2] */ +static const char * const rt5677_if12_adc2_src[] = { + "STO2 ADC MIX", "OB23" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc2_enum, RT5677_TDM1_CTRL2, + RT5677_IF1_ADC2_SFT, rt5677_if12_adc2_src); + +static const struct snd_kcontrol_new rt5677_if1_adc2_mux = + SOC_DAPM_ENUM("IF1 ADC2 source", rt5677_if1_adc2_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc2_enum, RT5677_TDM2_CTRL2, + RT5677_IF2_ADC2_SFT, rt5677_if12_adc2_src); + +static const struct snd_kcontrol_new rt5677_if2_adc2_mux = + SOC_DAPM_ENUM("IF2 ADC2 source", rt5677_if2_adc2_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_slb_adc2_enum, RT5677_SLIMBUS_RX, + RT5677_SLB_ADC2_SFT, rt5677_if12_adc2_src); + +static const struct snd_kcontrol_new rt5677_slb_adc2_mux = + SOC_DAPM_ENUM("SLB ADC2 source", rt5677_slb_adc2_enum); + +/* TDM IF1/2 SLB ADC3 Data Selection */ /* MX-3C MX-41 [9:8] MX-08 [5:4] */ +static const char * const rt5677_if12_adc3_src[] = { + "STO3 ADC MIX", "MONO ADC MIX", "OB45" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc3_enum, RT5677_TDM1_CTRL2, + RT5677_IF1_ADC3_SFT, rt5677_if12_adc3_src); + +static const struct snd_kcontrol_new rt5677_if1_adc3_mux = + SOC_DAPM_ENUM("IF1 ADC3 source", rt5677_if1_adc3_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc3_enum, RT5677_TDM2_CTRL2, + RT5677_IF2_ADC3_SFT, rt5677_if12_adc3_src); + +static const struct snd_kcontrol_new rt5677_if2_adc3_mux = + SOC_DAPM_ENUM("IF2 ADC3 source", rt5677_if2_adc3_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_slb_adc3_enum, RT5677_SLIMBUS_RX, + RT5677_SLB_ADC3_SFT, rt5677_if12_adc3_src); + +static const struct snd_kcontrol_new rt5677_slb_adc3_mux = + SOC_DAPM_ENUM("SLB ADC3 source", rt5677_slb_adc3_enum); + +/* TDM IF1/2 SLB ADC4 Data Selection */ /* MX-3C MX-41 [11:10] MX-08 [7:6] */ +static const char * const rt5677_if12_adc4_src[] = { + "STO4 ADC MIX", "OB67", "OB01" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc4_enum, RT5677_TDM1_CTRL2, + RT5677_IF1_ADC4_SFT, rt5677_if12_adc4_src); + +static const struct snd_kcontrol_new rt5677_if1_adc4_mux = + SOC_DAPM_ENUM("IF1 ADC4 source", rt5677_if1_adc4_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc4_enum, RT5677_TDM2_CTRL2, + RT5677_IF2_ADC4_SFT, rt5677_if12_adc4_src); + +static const struct snd_kcontrol_new rt5677_if2_adc4_mux = + SOC_DAPM_ENUM("IF2 ADC4 source", rt5677_if2_adc4_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_slb_adc4_enum, RT5677_SLIMBUS_RX, + RT5677_SLB_ADC4_SFT, rt5677_if12_adc4_src); + +static const struct snd_kcontrol_new rt5677_slb_adc4_mux = + SOC_DAPM_ENUM("SLB ADC4 source", rt5677_slb_adc4_enum); + +/* Interface3/4 ADC Data Input */ /* MX-2F [3:0] MX-30 [7:4]*/ +static const char * const rt5677_if34_adc_src[] = { + "STO1 ADC MIX", "STO2 ADC MIX", "STO3 ADC MIX", "STO4 ADC MIX", + "MONO ADC MIX", "OB01", "OB23", "VAD ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if3_adc_enum, RT5677_IF3_DATA, + RT5677_IF3_ADC_IN_SFT, rt5677_if34_adc_src); + +static const struct snd_kcontrol_new rt5677_if3_adc_mux = + SOC_DAPM_ENUM("IF3 ADC source", rt5677_if3_adc_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if4_adc_enum, RT5677_IF4_DATA, + RT5677_IF4_ADC_IN_SFT, rt5677_if34_adc_src); + +static const struct snd_kcontrol_new rt5677_if4_adc_mux = + SOC_DAPM_ENUM("IF4 ADC source", rt5677_if4_adc_enum); + +static int rt5677_bst1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, + RT5677_PWR_BST1_P, RT5677_PWR_BST1_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, + RT5677_PWR_BST1_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5677_bst2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, + RT5677_PWR_BST2_P, RT5677_PWR_BST2_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, + RT5677_PWR_BST2_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5677_set_pll1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x2); + regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x0); + break; + default: + return 0; + } + + return 0; +} + +static int rt5677_set_pll2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x2); + regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x0); + break; + default: + return 0; + } + + return 0; +} + +static int rt5677_set_micbias1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, + RT5677_PWR_CLK_MB1 | RT5677_PWR_PP_MB1 | + RT5677_PWR_CLK_MB, RT5677_PWR_CLK_MB1 | + RT5677_PWR_PP_MB1 | RT5677_PWR_CLK_MB); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT, + 0, rt5677_set_pll1_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("PLL2", RT5677_PWR_ANLG2, RT5677_PWR_PLL2_BIT, + 0, rt5677_set_pll2_event, SND_SOC_DAPM_POST_PMU), + + /* Input Side */ + /* micbias */ + SND_SOC_DAPM_SUPPLY("micbias1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT, + 0, rt5677_set_micbias1_event, SND_SOC_DAPM_POST_PMU), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + SND_SOC_DAPM_INPUT("DMIC L2"), + SND_SOC_DAPM_INPUT("DMIC R2"), + SND_SOC_DAPM_INPUT("DMIC L3"), + SND_SOC_DAPM_INPUT("DMIC R3"), + SND_SOC_DAPM_INPUT("DMIC L4"), + SND_SOC_DAPM_INPUT("DMIC R4"), + + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + SND_SOC_DAPM_INPUT("Haptic Generator"), + + SND_SOC_DAPM_PGA("DMIC1", RT5677_DMIC_CTRL1, RT5677_DMIC_1_EN_SFT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("DMIC2", RT5677_DMIC_CTRL1, RT5677_DMIC_2_EN_SFT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("DMIC3", RT5677_DMIC_CTRL1, RT5677_DMIC_3_EN_SFT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("DMIC4", RT5677_DMIC_CTRL2, RT5677_DMIC_4_EN_SFT, 0, + NULL, 0), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + + /* Boost */ + SND_SOC_DAPM_PGA_E("BST1", RT5677_PWR_ANLG2, + RT5677_PWR_BST1_BIT, 0, NULL, 0, rt5677_bst1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("BST2", RT5677_PWR_ANLG2, + RT5677_PWR_BST2_BIT, 0, NULL, 0, rt5677_bst2_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, + 0, 0), + SND_SOC_DAPM_ADC("ADC 2", NULL, SND_SOC_NOPM, + 0, 0), + SND_SOC_DAPM_PGA("ADC 1_2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("ADC 1 power", RT5677_PWR_DIG1, + RT5677_PWR_ADC_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC 2 power", RT5677_PWR_DIG1, + RT5677_PWR_ADC_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 clock", RT5677_PWR_DIG1, + RT5677_PWR_ADCFED1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC2 clock", RT5677_PWR_DIG1, + RT5677_PWR_ADCFED2_BIT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5677_sto1_dmic_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC1 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_sto1_adc1_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC2 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_sto1_adc2_mux), + SND_SOC_DAPM_MUX("Stereo2 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5677_sto2_dmic_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC1 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_sto2_adc1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC2 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_sto2_adc2_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC LR Mux", SND_SOC_NOPM, 0, 0, + &rt5677_sto2_adc_lr_mux), + SND_SOC_DAPM_MUX("Stereo3 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5677_sto3_dmic_mux), + SND_SOC_DAPM_MUX("Stereo3 ADC1 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_sto3_adc1_mux), + SND_SOC_DAPM_MUX("Stereo3 ADC2 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_sto3_adc2_mux), + SND_SOC_DAPM_MUX("Stereo4 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5677_sto4_dmic_mux), + SND_SOC_DAPM_MUX("Stereo4 ADC1 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_sto4_adc1_mux), + SND_SOC_DAPM_MUX("Stereo4 ADC2 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_sto4_adc2_mux), + SND_SOC_DAPM_MUX("Mono DMIC L Mux", SND_SOC_NOPM, 0, 0, + &rt5677_mono_dmic_l_mux), + SND_SOC_DAPM_MUX("Mono DMIC R Mux", SND_SOC_NOPM, 0, 0, + &rt5677_mono_dmic_r_mux), + SND_SOC_DAPM_MUX("Mono ADC2 L Mux", SND_SOC_NOPM, 0, 0, + &rt5677_mono_adc2_l_mux), + SND_SOC_DAPM_MUX("Mono ADC1 L Mux", SND_SOC_NOPM, 0, 0, + &rt5677_mono_adc1_l_mux), + SND_SOC_DAPM_MUX("Mono ADC1 R Mux", SND_SOC_NOPM, 0, 0, + &rt5677_mono_adc1_r_mux), + SND_SOC_DAPM_MUX("Mono ADC2 R Mux", SND_SOC_NOPM, 0, 0, + &rt5677_mono_adc2_r_mux), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("adc stereo1 filter", RT5677_PWR_DIG2, + RT5677_PWR_ADC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("adc stereo2 filter", RT5677_PWR_DIG2, + RT5677_PWR_ADC_S2F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("adc stereo3 filter", RT5677_PWR_DIG2, + RT5677_PWR_ADC_S3F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("adc stereo4 filter", RT5677_PWR_DIG2, + RT5677_PWR_ADC_S4F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5677_sto1_adc_l_mix, ARRAY_SIZE(rt5677_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5677_sto1_adc_r_mix, ARRAY_SIZE(rt5677_sto1_adc_r_mix)), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5677_sto2_adc_l_mix, ARRAY_SIZE(rt5677_sto2_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5677_sto2_adc_r_mix, ARRAY_SIZE(rt5677_sto2_adc_r_mix)), + SND_SOC_DAPM_MIXER("Sto3 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5677_sto3_adc_l_mix, ARRAY_SIZE(rt5677_sto3_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto3 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5677_sto3_adc_r_mix, ARRAY_SIZE(rt5677_sto3_adc_r_mix)), + SND_SOC_DAPM_MIXER("Sto4 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5677_sto4_adc_l_mix, ARRAY_SIZE(rt5677_sto4_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto4 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5677_sto4_adc_r_mix, ARRAY_SIZE(rt5677_sto4_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("adc mono left filter", RT5677_PWR_DIG2, + RT5677_PWR_ADC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5677_mono_adc_l_mix, ARRAY_SIZE(rt5677_mono_adc_l_mix)), + SND_SOC_DAPM_SUPPLY("adc mono right filter", RT5677_PWR_DIG2, + RT5677_PWR_ADC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5677_mono_adc_r_mix, ARRAY_SIZE(rt5677_mono_adc_r_mix)), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo1 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo3 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo3 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo3 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo4 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo4 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo4 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Sto2 ADC LR MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DSP */ + SND_SOC_DAPM_MUX("IB9 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_ib9_src_mux), + SND_SOC_DAPM_MUX("IB8 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_ib8_src_mux), + SND_SOC_DAPM_MUX("IB7 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_ib7_src_mux), + SND_SOC_DAPM_MUX("IB6 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_ib6_src_mux), + SND_SOC_DAPM_MUX("IB45 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_ib45_src_mux), + SND_SOC_DAPM_MUX("IB23 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_ib23_src_mux), + SND_SOC_DAPM_MUX("IB01 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_ib01_src_mux), + SND_SOC_DAPM_MUX("IB45 Bypass Mux", SND_SOC_NOPM, 0, 0, + &rt5677_ib45_bypass_src_mux), + SND_SOC_DAPM_MUX("IB23 Bypass Mux", SND_SOC_NOPM, 0, 0, + &rt5677_ib23_bypass_src_mux), + SND_SOC_DAPM_MUX("IB01 Bypass Mux", SND_SOC_NOPM, 0, 0, + &rt5677_ib01_bypass_src_mux), + SND_SOC_DAPM_MUX("OB23 Bypass Mux", SND_SOC_NOPM, 0, 0, + &rt5677_ob23_bypass_src_mux), + SND_SOC_DAPM_MUX("OB01 Bypass Mux", SND_SOC_NOPM, 0, 0, + &rt5677_ob01_bypass_src_mux), + + SND_SOC_DAPM_PGA("OB45", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OB67", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_PGA("OutBound2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OutBound3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OutBound4", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OutBound5", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OutBound6", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OutBound7", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5677_PWR_DIG1, + RT5677_PWR_I2S1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC0", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC4", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC5", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC6", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC7", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC01", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC23", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC45", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC67", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("I2S2", RT5677_PWR_DIG1, + RT5677_PWR_I2S2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC0", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC4", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC5", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC6", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC7", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC01", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC23", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC45", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC67", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("I2S3", RT5677_PWR_DIG1, + RT5677_PWR_I2S3_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF3 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF3 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF3 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF3 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF3 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF3 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("I2S4", RT5677_PWR_DIG1, + RT5677_PWR_I2S4_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF4 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF4 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF4 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF4 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF4 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF4 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("SLB", RT5677_PWR_DIG1, + RT5677_PWR_SLB_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB DAC0", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB DAC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB DAC4", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB DAC5", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB DAC6", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB DAC7", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB DAC01", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB DAC23", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB DAC45", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB DAC67", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SLB ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 ADC1 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc1_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc2_mux), + SND_SOC_DAPM_MUX("IF1 ADC3 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc3_mux), + SND_SOC_DAPM_MUX("IF1 ADC4 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc4_mux), + SND_SOC_DAPM_MUX("IF2 ADC1 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc1_mux), + SND_SOC_DAPM_MUX("IF2 ADC2 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc2_mux), + SND_SOC_DAPM_MUX("IF2 ADC3 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc3_mux), + SND_SOC_DAPM_MUX("IF2 ADC4 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc4_mux), + SND_SOC_DAPM_MUX("IF3 ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if3_adc_mux), + SND_SOC_DAPM_MUX("IF4 ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if4_adc_mux), + SND_SOC_DAPM_MUX("SLB ADC1 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_slb_adc1_mux), + SND_SOC_DAPM_MUX("SLB ADC2 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_slb_adc2_mux), + SND_SOC_DAPM_MUX("SLB ADC3 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_slb_adc3_mux), + SND_SOC_DAPM_MUX("SLB ADC4 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_slb_adc4_mux), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3RX", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3TX", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF4RX", "AIF4 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF4TX", "AIF4 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SLBRX", "SLIMBus Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLBTX", "SLIMBus Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Sidetone Mux */ + SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0, + &rt5677_sidetone_mux), + /* VAD Mux*/ + SND_SOC_DAPM_MUX("VAD ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5677_vad_src_mux), + + /* Tensilica DSP */ + SND_SOC_DAPM_PGA("Tensilica DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("OB01 MIX", SND_SOC_NOPM, 0, 0, + rt5677_ob_01_mix, ARRAY_SIZE(rt5677_ob_01_mix)), + SND_SOC_DAPM_MIXER("OB23 MIX", SND_SOC_NOPM, 0, 0, + rt5677_ob_23_mix, ARRAY_SIZE(rt5677_ob_23_mix)), + SND_SOC_DAPM_MIXER("OB4 MIX", SND_SOC_NOPM, 0, 0, + rt5677_ob_4_mix, ARRAY_SIZE(rt5677_ob_4_mix)), + SND_SOC_DAPM_MIXER("OB5 MIX", SND_SOC_NOPM, 0, 0, + rt5677_ob_5_mix, ARRAY_SIZE(rt5677_ob_5_mix)), + SND_SOC_DAPM_MIXER("OB6 MIX", SND_SOC_NOPM, 0, 0, + rt5677_ob_6_mix, ARRAY_SIZE(rt5677_ob_6_mix)), + SND_SOC_DAPM_MIXER("OB7 MIX", SND_SOC_NOPM, 0, 0, + rt5677_ob_7_mix, ARRAY_SIZE(rt5677_ob_7_mix)), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5677_dac_l_mix, ARRAY_SIZE(rt5677_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5677_dac_r_mix, ARRAY_SIZE(rt5677_dac_r_mix)), + SND_SOC_DAPM_PGA("DAC1 FS", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC Mux */ + SND_SOC_DAPM_MUX("DAC1 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_dac1_mux), + SND_SOC_DAPM_MUX("ADDA1 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_adda1_mux), + SND_SOC_DAPM_MUX("DAC12 SRC Mux", SND_SOC_NOPM, 0, 0, + &rt5677_dac12_mux), + SND_SOC_DAPM_MUX("DAC3 SRC Mux", SND_SOC_NOPM, 0, 0, + &rt5677_dac3_mux), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC2 L Mux", SND_SOC_NOPM, 0, 0, + &rt5677_dac2_l_mux), + SND_SOC_DAPM_MUX("DAC2 R Mux", SND_SOC_NOPM, 0, 0, + &rt5677_dac2_r_mux), + + /* DAC3 channel Mux */ + SND_SOC_DAPM_MUX("DAC3 L Mux", SND_SOC_NOPM, 0, 0, + &rt5677_dac3_l_mux), + SND_SOC_DAPM_MUX("DAC3 R Mux", SND_SOC_NOPM, 0, 0, + &rt5677_dac3_r_mux), + + /* DAC4 channel Mux */ + SND_SOC_DAPM_MUX("DAC4 L Mux", SND_SOC_NOPM, 0, 0, + &rt5677_dac4_l_mux), + SND_SOC_DAPM_MUX("DAC4 R Mux", SND_SOC_NOPM, 0, 0, + &rt5677_dac4_r_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono left filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M2F_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono right filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M2F_R_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5677_sto1_dac_l_mix, ARRAY_SIZE(rt5677_sto1_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5677_sto1_dac_r_mix, ARRAY_SIZE(rt5677_sto1_dac_r_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5677_mono_dac_l_mix, ARRAY_SIZE(rt5677_mono_dac_l_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5677_mono_dac_r_mix, ARRAY_SIZE(rt5677_mono_dac_r_mix)), + SND_SOC_DAPM_MIXER("DD1 MIXL", SND_SOC_NOPM, 0, 0, + rt5677_dd1_l_mix, ARRAY_SIZE(rt5677_dd1_l_mix)), + SND_SOC_DAPM_MIXER("DD1 MIXR", SND_SOC_NOPM, 0, 0, + rt5677_dd1_r_mix, ARRAY_SIZE(rt5677_dd1_r_mix)), + SND_SOC_DAPM_MIXER("DD2 MIXL", SND_SOC_NOPM, 0, 0, + rt5677_dd2_l_mix, ARRAY_SIZE(rt5677_dd2_l_mix)), + SND_SOC_DAPM_MIXER("DD2 MIXR", SND_SOC_NOPM, 0, 0, + rt5677_dd2_r_mix, ARRAY_SIZE(rt5677_dd2_r_mix)), + SND_SOC_DAPM_PGA("Stereo DAC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono DAC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DD1 MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DD2 MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC 1", NULL, RT5677_PWR_DIG1, + RT5677_PWR_DAC1_BIT, 0), + SND_SOC_DAPM_DAC("DAC 2", NULL, RT5677_PWR_DIG1, + RT5677_PWR_DAC2_BIT, 0), + SND_SOC_DAPM_DAC("DAC 3", NULL, RT5677_PWR_DIG1, + RT5677_PWR_DAC3_BIT, 0), + + /* PDM */ + SND_SOC_DAPM_SUPPLY("PDM1 Power", RT5677_PWR_DIG2, + RT5677_PWR_PDM1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5677_PWR_DIG2, + RT5677_PWR_PDM2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MUX("PDM1 L Mux", RT5677_PDM_OUT_CTRL, RT5677_M_PDM1_L_SFT, + 1, &rt5677_pdm1_l_mux), + SND_SOC_DAPM_MUX("PDM1 R Mux", RT5677_PDM_OUT_CTRL, RT5677_M_PDM1_R_SFT, + 1, &rt5677_pdm1_r_mux), + SND_SOC_DAPM_MUX("PDM2 L Mux", RT5677_PDM_OUT_CTRL, RT5677_M_PDM2_L_SFT, + 1, &rt5677_pdm2_l_mux), + SND_SOC_DAPM_MUX("PDM2 R Mux", RT5677_PDM_OUT_CTRL, RT5677_M_PDM2_R_SFT, + 1, &rt5677_pdm2_r_mux), + + SND_SOC_DAPM_PGA_S("LOUT1 amp", 1, RT5677_PWR_ANLG1, RT5677_PWR_LO1_BIT, + 0, NULL, 0), + SND_SOC_DAPM_PGA_S("LOUT2 amp", 1, RT5677_PWR_ANLG1, RT5677_PWR_LO2_BIT, + 0, NULL, 0), + SND_SOC_DAPM_PGA_S("LOUT3 amp", 1, RT5677_PWR_ANLG1, RT5677_PWR_LO3_BIT, + 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("LOUT3"), + SND_SOC_DAPM_OUTPUT("PDM1L"), + SND_SOC_DAPM_OUTPUT("PDM1R"), + SND_SOC_DAPM_OUTPUT("PDM2L"), + SND_SOC_DAPM_OUTPUT("PDM2R"), +}; + +static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { + { "DMIC1", NULL, "DMIC L1" }, + { "DMIC1", NULL, "DMIC R1" }, + { "DMIC2", NULL, "DMIC L2" }, + { "DMIC2", NULL, "DMIC R2" }, + { "DMIC3", NULL, "DMIC L3" }, + { "DMIC3", NULL, "DMIC R3" }, + { "DMIC4", NULL, "DMIC L4" }, + { "DMIC4", NULL, "DMIC R4" }, + + { "DMIC L1", NULL, "DMIC CLK" }, + { "DMIC R1", NULL, "DMIC CLK" }, + { "DMIC L2", NULL, "DMIC CLK" }, + { "DMIC R2", NULL, "DMIC CLK" }, + { "DMIC L3", NULL, "DMIC CLK" }, + { "DMIC R3", NULL, "DMIC CLK" }, + { "DMIC L4", NULL, "DMIC CLK" }, + { "DMIC R4", NULL, "DMIC CLK" }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST2", NULL, "IN2P" }, + { "BST2", NULL, "IN2N" }, + + { "IN1P", NULL, "micbias1" }, + { "IN1N", NULL, "micbias1" }, + { "IN2P", NULL, "micbias1" }, + { "IN2N", NULL, "micbias1" }, + + { "ADC 1", NULL, "BST1" }, + { "ADC 1", NULL, "ADC 1 power" }, + { "ADC 1", NULL, "ADC1 clock" }, + { "ADC 2", NULL, "BST2" }, + { "ADC 2", NULL, "ADC 2 power" }, + { "ADC 2", NULL, "ADC2 clock" }, + + { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo1 DMIC Mux", "DMIC3", "DMIC3" }, + { "Stereo1 DMIC Mux", "DMIC4", "DMIC4" }, + + { "Stereo2 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo2 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo2 DMIC Mux", "DMIC3", "DMIC3" }, + { "Stereo2 DMIC Mux", "DMIC4", "DMIC4" }, + + { "Stereo3 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo3 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo3 DMIC Mux", "DMIC3", "DMIC3" }, + { "Stereo3 DMIC Mux", "DMIC4", "DMIC4" }, + + { "Stereo4 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo4 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo4 DMIC Mux", "DMIC3", "DMIC3" }, + { "Stereo4 DMIC Mux", "DMIC4", "DMIC4" }, + + { "Mono DMIC L Mux", "DMIC1", "DMIC1" }, + { "Mono DMIC L Mux", "DMIC2", "DMIC2" }, + { "Mono DMIC L Mux", "DMIC3", "DMIC3" }, + { "Mono DMIC L Mux", "DMIC4", "DMIC4" }, + + { "Mono DMIC R Mux", "DMIC1", "DMIC1" }, + { "Mono DMIC R Mux", "DMIC2", "DMIC2" }, + { "Mono DMIC R Mux", "DMIC3", "DMIC3" }, + { "Mono DMIC R Mux", "DMIC4", "DMIC4" }, + + { "ADC 1_2", NULL, "ADC 1" }, + { "ADC 1_2", NULL, "ADC 2" }, + + { "Stereo1 ADC1 Mux", "DD MIX1", "DD1 MIX" }, + { "Stereo1 ADC1 Mux", "ADC1/2", "ADC 1_2" }, + { "Stereo1 ADC1 Mux", "Stereo DAC MIX", "Stereo DAC MIX" }, + + { "Stereo1 ADC2 Mux", "DD MIX1", "DD1 MIX" }, + { "Stereo1 ADC2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC2 Mux", "Stereo DAC MIX", "Stereo DAC MIX" }, + + { "Stereo2 ADC1 Mux", "DD MIX1", "DD1 MIX" }, + { "Stereo2 ADC1 Mux", "ADC1/2", "ADC 1_2" }, + { "Stereo2 ADC1 Mux", "Stereo DAC MIX", "Stereo DAC MIX" }, + + { "Stereo2 ADC2 Mux", "DD MIX1", "DD1 MIX" }, + { "Stereo2 ADC2 Mux", "DMIC", "Stereo2 DMIC Mux" }, + { "Stereo2 ADC2 Mux", "Stereo DAC MIX", "Stereo DAC MIX" }, + + { "Stereo3 ADC1 Mux", "DD MIX1", "DD1 MIX" }, + { "Stereo3 ADC1 Mux", "ADC1/2", "ADC 1_2" }, + { "Stereo3 ADC1 Mux", "Stereo DAC MIX", "Stereo DAC MIX" }, + + { "Stereo3 ADC2 Mux", "DD MIX1", "DD1 MIX" }, + { "Stereo3 ADC2 Mux", "DMIC", "Stereo3 DMIC Mux" }, + { "Stereo3 ADC2 Mux", "Stereo DAC MIX", "Stereo DAC MIX" }, + + { "Stereo4 ADC1 Mux", "DD MIX1", "DD1 MIX" }, + { "Stereo4 ADC1 Mux", "ADC1/2", "ADC 1_2" }, + { "Stereo4 ADC1 Mux", "DD MIX2", "DD2 MIX" }, + + { "Stereo4 ADC2 Mux", "DD MIX1", "DD1 MIX" }, + { "Stereo4 ADC2 Mux", "DMIC", "Stereo3 DMIC Mux" }, + { "Stereo4 ADC2 Mux", "DD MIX2", "DD2 MIX" }, + + { "Mono ADC2 L Mux", "DD MIX1L", "DD1 MIXL" }, + { "Mono ADC2 L Mux", "DMIC", "Mono DMIC L Mux" }, + { "Mono ADC2 L Mux", "MONO DAC MIXL", "Mono DAC MIXL" }, + + { "Mono ADC1 L Mux", "DD MIX1L", "DD1 MIXL" }, + { "Mono ADC1 L Mux", "ADC1", "ADC 1" }, + { "Mono ADC1 L Mux", "MONO DAC MIXL", "Mono DAC MIXL" }, + + { "Mono ADC1 R Mux", "DD MIX1R", "DD1 MIXR" }, + { "Mono ADC1 R Mux", "ADC2", "ADC 2" }, + { "Mono ADC1 R Mux", "MONO DAC MIXR", "Mono DAC MIXR" }, + + { "Mono ADC2 R Mux", "DD MIX1R", "DD1 MIXR" }, + { "Mono ADC2 R Mux", "DMIC", "Mono DMIC R Mux" }, + { "Mono ADC2 R Mux", "MONO DAC MIXR", "Mono DAC MIXR" }, + + { "Sto1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC1 Mux" }, + { "Sto1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC2 Mux" }, + { "Sto1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC1 Mux" }, + { "Sto1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC2 Mux" }, + + { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, + { "Stereo1 ADC MIXL", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, + { "Stereo1 ADC MIXR", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL" }, + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR" }, + + { "Sto2 ADC MIXL", "ADC1 Switch", "Stereo2 ADC1 Mux" }, + { "Sto2 ADC MIXL", "ADC2 Switch", "Stereo2 ADC2 Mux" }, + { "Sto2 ADC MIXR", "ADC1 Switch", "Stereo2 ADC1 Mux" }, + { "Sto2 ADC MIXR", "ADC2 Switch", "Stereo2 ADC2 Mux" }, + + { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXL" }, + { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXR" }, + + { "Stereo2 ADC LR Mux", "L", "Sto2 ADC MIXL" }, + { "Stereo2 ADC LR Mux", "LR", "Sto2 ADC LR MIX" }, + + { "Stereo2 ADC MIXL", NULL, "Stereo2 ADC LR Mux" }, + { "Stereo2 ADC MIXL", NULL, "adc stereo2 filter" }, + { "adc stereo2 filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo2 ADC MIXR", NULL, "Sto2 ADC MIXR" }, + { "Stereo2 ADC MIXR", NULL, "adc stereo2 filter" }, + { "adc stereo2 filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo2 ADC MIX", NULL, "Stereo2 ADC MIXL" }, + { "Stereo2 ADC MIX", NULL, "Stereo2 ADC MIXR" }, + + { "Sto3 ADC MIXL", "ADC1 Switch", "Stereo3 ADC1 Mux" }, + { "Sto3 ADC MIXL", "ADC2 Switch", "Stereo3 ADC2 Mux" }, + { "Sto3 ADC MIXR", "ADC1 Switch", "Stereo3 ADC1 Mux" }, + { "Sto3 ADC MIXR", "ADC2 Switch", "Stereo3 ADC2 Mux" }, + + { "Stereo3 ADC MIXL", NULL, "Sto3 ADC MIXL" }, + { "Stereo3 ADC MIXL", NULL, "adc stereo3 filter" }, + { "adc stereo3 filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo3 ADC MIXR", NULL, "Sto3 ADC MIXR" }, + { "Stereo3 ADC MIXR", NULL, "adc stereo3 filter" }, + { "adc stereo3 filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo3 ADC MIX", NULL, "Stereo3 ADC MIXL" }, + { "Stereo3 ADC MIX", NULL, "Stereo3 ADC MIXR" }, + + { "Sto4 ADC MIXL", "ADC1 Switch", "Stereo4 ADC1 Mux" }, + { "Sto4 ADC MIXL", "ADC2 Switch", "Stereo4 ADC2 Mux" }, + { "Sto4 ADC MIXR", "ADC1 Switch", "Stereo4 ADC1 Mux" }, + { "Sto4 ADC MIXR", "ADC2 Switch", "Stereo4 ADC2 Mux" }, + + { "Stereo4 ADC MIXL", NULL, "Sto4 ADC MIXL" }, + { "Stereo4 ADC MIXL", NULL, "adc stereo4 filter" }, + { "adc stereo4 filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo4 ADC MIXR", NULL, "Sto4 ADC MIXR" }, + { "Stereo4 ADC MIXR", NULL, "adc stereo4 filter" }, + { "adc stereo4 filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo4 ADC MIX", NULL, "Stereo4 ADC MIXL" }, + { "Stereo4 ADC MIX", NULL, "Stereo4 ADC MIXR" }, + + { "Mono ADC MIXL", "ADC1 Switch", "Mono ADC1 L Mux" }, + { "Mono ADC MIXL", "ADC2 Switch", "Mono ADC2 L Mux" }, + { "Mono ADC MIXL", NULL, "adc mono left filter" }, + { "adc mono left filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Mono ADC MIXR", "ADC1 Switch", "Mono ADC1 R Mux" }, + { "Mono ADC MIXR", "ADC2 Switch", "Mono ADC2 R Mux" }, + { "Mono ADC MIXR", NULL, "adc mono right filter" }, + { "adc mono right filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Mono ADC MIX", NULL, "Mono ADC MIXL" }, + { "Mono ADC MIX", NULL, "Mono ADC MIXR" }, + + { "VAD ADC Mux", "STO1 ADC MIX L", "Stereo1 ADC MIXL" }, + { "VAD ADC Mux", "MONO ADC MIX L", "Mono ADC MIXL" }, + { "VAD ADC Mux", "MONO ADC MIX R", "Mono ADC MIXR" }, + { "VAD ADC Mux", "STO2 ADC MIX L", "Stereo2 ADC MIXL" }, + { "VAD ADC Mux", "STO3 ADC MIX L", "Stereo3 ADC MIXL" }, + + { "IF1 ADC1 Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, + { "IF1 ADC1 Mux", "OB01", "OB01 Bypass Mux" }, + { "IF1 ADC1 Mux", "VAD ADC", "VAD ADC Mux" }, + + { "IF1 ADC2 Mux", "STO2 ADC MIX", "Stereo2 ADC MIX" }, + { "IF1 ADC2 Mux", "OB23", "OB23 Bypass Mux" }, + + { "IF1 ADC3 Mux", "STO3 ADC MIX", "Stereo3 ADC MIX" }, + { "IF1 ADC3 Mux", "MONO ADC MIX", "Mono ADC MIX" }, + { "IF1 ADC3 Mux", "OB45", "OB45" }, + + { "IF1 ADC4 Mux", "STO4 ADC MIX", "Stereo4 ADC MIX" }, + { "IF1 ADC4 Mux", "OB67", "OB67" }, + { "IF1 ADC4 Mux", "OB01", "OB01 Bypass Mux" }, + + { "AIF1TX", NULL, "I2S1" }, + { "AIF1TX", NULL, "IF1 ADC1 Mux" }, + { "AIF1TX", NULL, "IF1 ADC2 Mux" }, + { "AIF1TX", NULL, "IF1 ADC3 Mux" }, + { "AIF1TX", NULL, "IF1 ADC4 Mux" }, + + { "IF2 ADC1 Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, + { "IF2 ADC1 Mux", "OB01", "OB01 Bypass Mux" }, + { "IF2 ADC1 Mux", "VAD ADC", "VAD ADC Mux" }, + + { "IF2 ADC2 Mux", "STO2 ADC MIX", "Stereo2 ADC MIX" }, + { "IF2 ADC2 Mux", "OB23", "OB23 Bypass Mux" }, + + { "IF2 ADC3 Mux", "STO3 ADC MIX", "Stereo3 ADC MIX" }, + { "IF2 ADC3 Mux", "MONO ADC MIX", "Mono ADC MIX" }, + { "IF2 ADC3 Mux", "OB45", "OB45" }, + + { "IF2 ADC4 Mux", "STO4 ADC MIX", "Stereo4 ADC MIX" }, + { "IF2 ADC4 Mux", "OB67", "OB67" }, + { "IF2 ADC4 Mux", "OB01", "OB01 Bypass Mux" }, + + { "AIF2TX", NULL, "I2S2" }, + { "AIF2TX", NULL, "IF2 ADC1 Mux" }, + { "AIF2TX", NULL, "IF2 ADC2 Mux" }, + { "AIF2TX", NULL, "IF2 ADC3 Mux" }, + { "AIF2TX", NULL, "IF2 ADC4 Mux" }, + + { "IF3 ADC Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, + { "IF3 ADC Mux", "STO2 ADC MIX", "Stereo2 ADC MIX" }, + { "IF3 ADC Mux", "STO3 ADC MIX", "Stereo3 ADC MIX" }, + { "IF3 ADC Mux", "STO4 ADC MIX", "Stereo4 ADC MIX" }, + { "IF3 ADC Mux", "MONO ADC MIX", "Mono ADC MIX" }, + { "IF3 ADC Mux", "OB01", "OB01 Bypass Mux" }, + { "IF3 ADC Mux", "OB23", "OB23 Bypass Mux" }, + { "IF3 ADC Mux", "VAD ADC", "VAD ADC Mux" }, + + { "AIF3TX", NULL, "I2S3" }, + { "AIF3TX", NULL, "IF3 ADC Mux" }, + + { "IF4 ADC Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, + { "IF4 ADC Mux", "STO2 ADC MIX", "Stereo2 ADC MIX" }, + { "IF4 ADC Mux", "STO3 ADC MIX", "Stereo3 ADC MIX" }, + { "IF4 ADC Mux", "STO4 ADC MIX", "Stereo4 ADC MIX" }, + { "IF4 ADC Mux", "MONO ADC MIX", "Mono ADC MIX" }, + { "IF4 ADC Mux", "OB01", "OB01 Bypass Mux" }, + { "IF4 ADC Mux", "OB23", "OB23 Bypass Mux" }, + { "IF4 ADC Mux", "VAD ADC", "VAD ADC Mux" }, + + { "AIF4TX", NULL, "I2S4" }, + { "AIF4TX", NULL, "IF4 ADC Mux" }, + + { "SLB ADC1 Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, + { "SLB ADC1 Mux", "OB01", "OB01 Bypass Mux" }, + { "SLB ADC1 Mux", "VAD ADC", "VAD ADC Mux" }, + + { "SLB ADC2 Mux", "STO2 ADC MIX", "Stereo2 ADC MIX" }, + { "SLB ADC2 Mux", "OB23", "OB23 Bypass Mux" }, + + { "SLB ADC3 Mux", "STO3 ADC MIX", "Stereo3 ADC MIX" }, + { "SLB ADC3 Mux", "MONO ADC MIX", "Mono ADC MIX" }, + { "SLB ADC3 Mux", "OB45", "OB45" }, + + { "SLB ADC4 Mux", "STO4 ADC MIX", "Stereo4 ADC MIX" }, + { "SLB ADC4 Mux", "OB67", "OB67" }, + { "SLB ADC4 Mux", "OB01", "OB01 Bypass Mux" }, + + { "SLBTX", NULL, "SLB" }, + { "SLBTX", NULL, "SLB ADC1 Mux" }, + { "SLBTX", NULL, "SLB ADC2 Mux" }, + { "SLBTX", NULL, "SLB ADC3 Mux" }, + { "SLBTX", NULL, "SLB ADC4 Mux" }, + + { "IB01 Mux", "IF1 DAC 01", "IF1 DAC01" }, + { "IB01 Mux", "IF2 DAC 01", "IF2 DAC01" }, + { "IB01 Mux", "SLB DAC 01", "SLB DAC01" }, + { "IB01 Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, + { "IB01 Mux", "VAD ADC/DAC1 FS", "DAC1 FS" }, + + { "IB01 Bypass Mux", "Bypass", "IB01 Mux" }, + { "IB01 Bypass Mux", "Pass SRC", "IB01 Mux" }, + + { "IB23 Mux", "IF1 DAC 23", "IF1 DAC23" }, + { "IB23 Mux", "IF2 DAC 23", "IF2 DAC23" }, + { "IB23 Mux", "SLB DAC 23", "SLB DAC23" }, + { "IB23 Mux", "STO2 ADC MIX", "Stereo2 ADC MIX" }, + { "IB23 Mux", "DAC1 FS", "DAC1 FS" }, + { "IB23 Mux", "IF4 DAC", "IF4 DAC" }, + + { "IB23 Bypass Mux", "Bypass", "IB23 Mux" }, + { "IB23 Bypass Mux", "Pass SRC", "IB23 Mux" }, + + { "IB45 Mux", "IF1 DAC 45", "IF1 DAC45" }, + { "IB45 Mux", "IF2 DAC 45", "IF2 DAC45" }, + { "IB45 Mux", "SLB DAC 45", "SLB DAC45" }, + { "IB45 Mux", "STO3 ADC MIX", "Stereo3 ADC MIX" }, + { "IB45 Mux", "IF3 DAC", "IF3 DAC" }, + + { "IB45 Bypass Mux", "Bypass", "IB45 Mux" }, + { "IB45 Bypass Mux", "Pass SRC", "IB45 Mux" }, + + { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6" }, + { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6" }, + { "IB6 Mux", "SLB DAC 6", "SLB DAC6" }, + { "IB6 Mux", "STO4 ADC MIX L", "Stereo4 ADC MIXL" }, + { "IB6 Mux", "IF4 DAC L", "IF4 DAC L" }, + { "IB6 Mux", "STO1 ADC MIX L", "Stereo1 ADC MIXL" }, + { "IB6 Mux", "STO2 ADC MIX L", "Stereo2 ADC MIXL" }, + { "IB6 Mux", "STO3 ADC MIX L", "Stereo3 ADC MIXL" }, + + { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7" }, + { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7" }, + { "IB7 Mux", "SLB DAC 7", "SLB DAC7" }, + { "IB7 Mux", "STO4 ADC MIX R", "Stereo4 ADC MIXR" }, + { "IB7 Mux", "IF4 DAC R", "IF4 DAC R" }, + { "IB7 Mux", "STO1 ADC MIX R", "Stereo1 ADC MIXR" }, + { "IB7 Mux", "STO2 ADC MIX R", "Stereo2 ADC MIXR" }, + { "IB7 Mux", "STO3 ADC MIX R", "Stereo3 ADC MIXR" }, + + { "IB8 Mux", "STO1 ADC MIX L", "Stereo1 ADC MIXL" }, + { "IB8 Mux", "STO2 ADC MIX L", "Stereo2 ADC MIXL" }, + { "IB8 Mux", "STO3 ADC MIX L", "Stereo3 ADC MIXL" }, + { "IB8 Mux", "STO4 ADC MIX L", "Stereo4 ADC MIXL" }, + { "IB8 Mux", "MONO ADC MIX L", "Mono ADC MIXL" }, + { "IB8 Mux", "DACL1 FS", "DAC1 MIXL" }, + + { "IB9 Mux", "STO1 ADC MIX R", "Stereo1 ADC MIXR" }, + { "IB9 Mux", "STO2 ADC MIX R", "Stereo2 ADC MIXR" }, + { "IB9 Mux", "STO3 ADC MIX R", "Stereo3 ADC MIXR" }, + { "IB9 Mux", "STO4 ADC MIX R", "Stereo4 ADC MIXR" }, + { "IB9 Mux", "MONO ADC MIX R", "Mono ADC MIXR" }, + { "IB9 Mux", "DACR1 FS", "DAC1 MIXR" }, + { "IB9 Mux", "DAC1 FS", "DAC1 FS" }, + + { "OB01 MIX", "IB01 Switch", "IB01 Bypass Mux" }, + { "OB01 MIX", "IB23 Switch", "IB23 Bypass Mux" }, + { "OB01 MIX", "IB45 Switch", "IB45 Bypass Mux" }, + { "OB01 MIX", "IB6 Switch", "IB6 Mux" }, + { "OB01 MIX", "IB7 Switch", "IB7 Mux" }, + { "OB01 MIX", "IB8 Switch", "IB8 Mux" }, + { "OB01 MIX", "IB9 Switch", "IB9 Mux" }, + + { "OB23 MIX", "IB01 Switch", "IB01 Bypass Mux" }, + { "OB23 MIX", "IB23 Switch", "IB23 Bypass Mux" }, + { "OB23 MIX", "IB45 Switch", "IB45 Bypass Mux" }, + { "OB23 MIX", "IB6 Switch", "IB6 Mux" }, + { "OB23 MIX", "IB7 Switch", "IB7 Mux" }, + { "OB23 MIX", "IB8 Switch", "IB8 Mux" }, + { "OB23 MIX", "IB9 Switch", "IB9 Mux" }, + + { "OB4 MIX", "IB01 Switch", "IB01 Bypass Mux" }, + { "OB4 MIX", "IB23 Switch", "IB23 Bypass Mux" }, + { "OB4 MIX", "IB45 Switch", "IB45 Bypass Mux" }, + { "OB4 MIX", "IB6 Switch", "IB6 Mux" }, + { "OB4 MIX", "IB7 Switch", "IB7 Mux" }, + { "OB4 MIX", "IB8 Switch", "IB8 Mux" }, + { "OB4 MIX", "IB9 Switch", "IB9 Mux" }, + + { "OB5 MIX", "IB01 Switch", "IB01 Bypass Mux" }, + { "OB5 MIX", "IB23 Switch", "IB23 Bypass Mux" }, + { "OB5 MIX", "IB45 Switch", "IB45 Bypass Mux" }, + { "OB5 MIX", "IB6 Switch", "IB6 Mux" }, + { "OB5 MIX", "IB7 Switch", "IB7 Mux" }, + { "OB5 MIX", "IB8 Switch", "IB8 Mux" }, + { "OB5 MIX", "IB9 Switch", "IB9 Mux" }, + + { "OB6 MIX", "IB01 Switch", "IB01 Bypass Mux" }, + { "OB6 MIX", "IB23 Switch", "IB23 Bypass Mux" }, + { "OB6 MIX", "IB45 Switch", "IB45 Bypass Mux" }, + { "OB6 MIX", "IB6 Switch", "IB6 Mux" }, + { "OB6 MIX", "IB7 Switch", "IB7 Mux" }, + { "OB6 MIX", "IB8 Switch", "IB8 Mux" }, + { "OB6 MIX", "IB9 Switch", "IB9 Mux" }, + + { "OB7 MIX", "IB01 Switch", "IB01 Bypass Mux" }, + { "OB7 MIX", "IB23 Switch", "IB23 Bypass Mux" }, + { "OB7 MIX", "IB45 Switch", "IB45 Bypass Mux" }, + { "OB7 MIX", "IB6 Switch", "IB6 Mux" }, + { "OB7 MIX", "IB7 Switch", "IB7 Mux" }, + { "OB7 MIX", "IB8 Switch", "IB8 Mux" }, + { "OB7 MIX", "IB9 Switch", "IB9 Mux" }, + + { "OB01 Bypass Mux", "Bypass", "OB01 MIX" }, + { "OB01 Bypass Mux", "Pass SRC", "OB01 MIX" }, + { "OB23 Bypass Mux", "Bypass", "OB23 MIX" }, + { "OB23 Bypass Mux", "Pass SRC", "OB23 MIX" }, + + { "OutBound2", NULL, "OB23 Bypass Mux" }, + { "OutBound3", NULL, "OB23 Bypass Mux" }, + { "OutBound4", NULL, "OB4 MIX" }, + { "OutBound5", NULL, "OB5 MIX" }, + { "OutBound6", NULL, "OB6 MIX" }, + { "OutBound7", NULL, "OB7 MIX" }, + + { "OB45", NULL, "OutBound4" }, + { "OB45", NULL, "OutBound5" }, + { "OB67", NULL, "OutBound6" }, + { "OB67", NULL, "OutBound7" }, + + { "IF1 DAC0", NULL, "AIF1RX" }, + { "IF1 DAC1", NULL, "AIF1RX" }, + { "IF1 DAC2", NULL, "AIF1RX" }, + { "IF1 DAC3", NULL, "AIF1RX" }, + { "IF1 DAC4", NULL, "AIF1RX" }, + { "IF1 DAC5", NULL, "AIF1RX" }, + { "IF1 DAC6", NULL, "AIF1RX" }, + { "IF1 DAC7", NULL, "AIF1RX" }, + { "IF1 DAC0", NULL, "I2S1" }, + { "IF1 DAC1", NULL, "I2S1" }, + { "IF1 DAC2", NULL, "I2S1" }, + { "IF1 DAC3", NULL, "I2S1" }, + { "IF1 DAC4", NULL, "I2S1" }, + { "IF1 DAC5", NULL, "I2S1" }, + { "IF1 DAC6", NULL, "I2S1" }, + { "IF1 DAC7", NULL, "I2S1" }, + + { "IF1 DAC01", NULL, "IF1 DAC0" }, + { "IF1 DAC01", NULL, "IF1 DAC1" }, + { "IF1 DAC23", NULL, "IF1 DAC2" }, + { "IF1 DAC23", NULL, "IF1 DAC3" }, + { "IF1 DAC45", NULL, "IF1 DAC4" }, + { "IF1 DAC45", NULL, "IF1 DAC5" }, + { "IF1 DAC67", NULL, "IF1 DAC6" }, + { "IF1 DAC67", NULL, "IF1 DAC7" }, + + { "IF2 DAC0", NULL, "AIF2RX" }, + { "IF2 DAC1", NULL, "AIF2RX" }, + { "IF2 DAC2", NULL, "AIF2RX" }, + { "IF2 DAC3", NULL, "AIF2RX" }, + { "IF2 DAC4", NULL, "AIF2RX" }, + { "IF2 DAC5", NULL, "AIF2RX" }, + { "IF2 DAC6", NULL, "AIF2RX" }, + { "IF2 DAC7", NULL, "AIF2RX" }, + { "IF2 DAC0", NULL, "I2S2" }, + { "IF2 DAC1", NULL, "I2S2" }, + { "IF2 DAC2", NULL, "I2S2" }, + { "IF2 DAC3", NULL, "I2S2" }, + { "IF2 DAC4", NULL, "I2S2" }, + { "IF2 DAC5", NULL, "I2S2" }, + { "IF2 DAC6", NULL, "I2S2" }, + { "IF2 DAC7", NULL, "I2S2" }, + + { "IF2 DAC01", NULL, "IF2 DAC0" }, + { "IF2 DAC01", NULL, "IF2 DAC1" }, + { "IF2 DAC23", NULL, "IF2 DAC2" }, + { "IF2 DAC23", NULL, "IF2 DAC3" }, + { "IF2 DAC45", NULL, "IF2 DAC4" }, + { "IF2 DAC45", NULL, "IF2 DAC5" }, + { "IF2 DAC67", NULL, "IF2 DAC6" }, + { "IF2 DAC67", NULL, "IF2 DAC7" }, + + { "IF3 DAC", NULL, "AIF3RX" }, + { "IF3 DAC", NULL, "I2S3" }, + + { "IF4 DAC", NULL, "AIF4RX" }, + { "IF4 DAC", NULL, "I2S4" }, + + { "IF3 DAC L", NULL, "IF3 DAC" }, + { "IF3 DAC R", NULL, "IF3 DAC" }, + + { "IF4 DAC L", NULL, "IF4 DAC" }, + { "IF4 DAC R", NULL, "IF4 DAC" }, + + { "SLB DAC0", NULL, "SLBRX" }, + { "SLB DAC1", NULL, "SLBRX" }, + { "SLB DAC2", NULL, "SLBRX" }, + { "SLB DAC3", NULL, "SLBRX" }, + { "SLB DAC4", NULL, "SLBRX" }, + { "SLB DAC5", NULL, "SLBRX" }, + { "SLB DAC6", NULL, "SLBRX" }, + { "SLB DAC7", NULL, "SLBRX" }, + { "SLB DAC0", NULL, "SLB" }, + { "SLB DAC1", NULL, "SLB" }, + { "SLB DAC2", NULL, "SLB" }, + { "SLB DAC3", NULL, "SLB" }, + { "SLB DAC4", NULL, "SLB" }, + { "SLB DAC5", NULL, "SLB" }, + { "SLB DAC6", NULL, "SLB" }, + { "SLB DAC7", NULL, "SLB" }, + + { "SLB DAC01", NULL, "SLB DAC0" }, + { "SLB DAC01", NULL, "SLB DAC1" }, + { "SLB DAC23", NULL, "SLB DAC2" }, + { "SLB DAC23", NULL, "SLB DAC3" }, + { "SLB DAC45", NULL, "SLB DAC4" }, + { "SLB DAC45", NULL, "SLB DAC5" }, + { "SLB DAC67", NULL, "SLB DAC6" }, + { "SLB DAC67", NULL, "SLB DAC7" }, + + { "ADDA1 Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, + { "ADDA1 Mux", "STO2 ADC MIX", "Stereo2 ADC MIX" }, + { "ADDA1 Mux", "OB 67", "OB67" }, + + { "DAC1 Mux", "IF1 DAC 01", "IF1 DAC01" }, + { "DAC1 Mux", "IF2 DAC 01", "IF2 DAC01" }, + { "DAC1 Mux", "IF3 DAC LR", "IF3 DAC" }, + { "DAC1 Mux", "IF4 DAC LR", "IF4 DAC" }, + { "DAC1 Mux", "SLB DAC 01", "SLB DAC01" }, + { "DAC1 Mux", "OB 01", "OB01 Bypass Mux" }, + + { "DAC1 MIXL", "Stereo ADC Switch", "ADDA1 Mux" }, + { "DAC1 MIXL", "DAC1 Switch", "DAC1 Mux" }, + { "DAC1 MIXL", NULL, "dac stereo1 filter" }, + { "DAC1 MIXR", "Stereo ADC Switch", "ADDA1 Mux" }, + { "DAC1 MIXR", "DAC1 Switch", "DAC1 Mux" }, + { "DAC1 MIXR", NULL, "dac stereo1 filter" }, + + { "DAC1 FS", NULL, "DAC1 MIXL" }, + { "DAC1 FS", NULL, "DAC1 MIXR" }, + + { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2" }, + { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2" }, + { "DAC2 L Mux", "IF3 DAC L", "IF3 DAC L" }, + { "DAC2 L Mux", "IF4 DAC L", "IF4 DAC L" }, + { "DAC2 L Mux", "SLB DAC 2", "SLB DAC2" }, + { "DAC2 L Mux", "OB 2", "OutBound2" }, + + { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3" }, + { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3" }, + { "DAC2 R Mux", "IF3 DAC R", "IF3 DAC R" }, + { "DAC2 R Mux", "IF4 DAC R", "IF4 DAC R" }, + { "DAC2 R Mux", "SLB DAC 3", "SLB DAC3" }, + { "DAC2 R Mux", "OB 3", "OutBound3" }, + { "DAC2 R Mux", "Haptic Generator", "Haptic Generator" }, + { "DAC2 R Mux", "VAD ADC", "VAD ADC Mux" }, + + { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4" }, + { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4" }, + { "DAC3 L Mux", "IF3 DAC L", "IF3 DAC L" }, + { "DAC3 L Mux", "IF4 DAC L", "IF4 DAC L" }, + { "DAC3 L Mux", "SLB DAC 4", "SLB DAC4" }, + { "DAC3 L Mux", "OB 4", "OutBound4" }, + + { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC4" }, + { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC4" }, + { "DAC3 R Mux", "IF3 DAC R", "IF3 DAC R" }, + { "DAC3 R Mux", "IF4 DAC R", "IF4 DAC R" }, + { "DAC3 R Mux", "SLB DAC 5", "SLB DAC5" }, + { "DAC3 R Mux", "OB 5", "OutBound5" }, + + { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6" }, + { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6" }, + { "DAC4 L Mux", "IF3 DAC L", "IF3 DAC L" }, + { "DAC4 L Mux", "IF4 DAC L", "IF4 DAC L" }, + { "DAC4 L Mux", "SLB DAC 6", "SLB DAC6" }, + { "DAC4 L Mux", "OB 6", "OutBound6" }, + + { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7" }, + { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7" }, + { "DAC4 R Mux", "IF3 DAC R", "IF3 DAC R" }, + { "DAC4 R Mux", "IF4 DAC R", "IF4 DAC R" }, + { "DAC4 R Mux", "SLB DAC 7", "SLB DAC7" }, + { "DAC4 R Mux", "OB 7", "OutBound7" }, + + { "Sidetone Mux", "DMIC1 L", "DMIC L1" }, + { "Sidetone Mux", "DMIC2 L", "DMIC L2" }, + { "Sidetone Mux", "DMIC3 L", "DMIC L3" }, + { "Sidetone Mux", "DMIC4 L", "DMIC L4" }, + { "Sidetone Mux", "ADC1", "ADC 1" }, + { "Sidetone Mux", "ADC2", "ADC 2" }, + + { "Stereo DAC MIXL", "ST L Switch", "Sidetone Mux" }, + { "Stereo DAC MIXL", "DAC1 L Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXL", "DAC2 L Switch", "DAC2 L Mux" }, + { "Stereo DAC MIXL", "DAC1 R Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXL", NULL, "dac stereo1 filter" }, + { "Stereo DAC MIXR", "ST R Switch", "Sidetone Mux" }, + { "Stereo DAC MIXR", "DAC1 R Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXR", "DAC2 R Switch", "DAC2 R Mux" }, + { "Stereo DAC MIXR", "DAC1 L Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXR", NULL, "dac stereo1 filter" }, + + { "Mono DAC MIXL", "ST L Switch", "Sidetone Mux" }, + { "Mono DAC MIXL", "DAC1 L Switch", "DAC1 MIXL" }, + { "Mono DAC MIXL", "DAC2 L Switch", "DAC2 L Mux" }, + { "Mono DAC MIXL", "DAC2 R Switch", "DAC2 R Mux" }, + { "Mono DAC MIXL", NULL, "dac mono left filter" }, + { "Mono DAC MIXR", "ST R Switch", "Sidetone Mux" }, + { "Mono DAC MIXR", "DAC1 R Switch", "DAC1 MIXR" }, + { "Mono DAC MIXR", "DAC2 R Switch", "DAC2 R Mux" }, + { "Mono DAC MIXR", "DAC2 L Switch", "DAC2 L Mux" }, + { "Mono DAC MIXR", NULL, "dac mono right filter" }, + + { "DD1 MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, + { "DD1 MIXL", "Mono DAC Mix L Switch", "Mono DAC MIXL" }, + { "DD1 MIXL", "DAC3 L Switch", "DAC3 L Mux" }, + { "DD1 MIXL", "DAC3 R Switch", "DAC3 R Mux" }, + { "DD1 MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, + { "DD1 MIXR", "Mono DAC Mix R Switch", "Mono DAC MIXR" }, + { "DD1 MIXR", "DAC3 L Switch", "DAC3 L Mux" }, + { "DD1 MIXR", "DAC3 R Switch", "DAC3 R Mux" }, + + { "DD2 MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, + { "DD2 MIXL", "Mono DAC Mix L Switch", "Mono DAC MIXL" }, + { "DD2 MIXL", "DAC4 L Switch", "DAC4 L Mux" }, + { "DD2 MIXL", "DAC4 R Switch", "DAC4 R Mux" }, + { "DD2 MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, + { "DD2 MIXR", "Mono DAC Mix R Switch", "Mono DAC MIXR" }, + { "DD2 MIXR", "DAC4 L Switch", "DAC4 L Mux" }, + { "DD2 MIXR", "DAC4 R Switch", "DAC4 R Mux" }, + + { "Stereo DAC MIX", NULL, "Stereo DAC MIXL" }, + { "Stereo DAC MIX", NULL, "Stereo DAC MIXR" }, + { "Mono DAC MIX", NULL, "Mono DAC MIXL" }, + { "Mono DAC MIX", NULL, "Mono DAC MIXR" }, + { "DD1 MIX", NULL, "DD1 MIXL" }, + { "DD1 MIX", NULL, "DD1 MIXR" }, + { "DD2 MIX", NULL, "DD2 MIXL" }, + { "DD2 MIX", NULL, "DD2 MIXR" }, + + { "DAC12 SRC Mux", "STO1 DAC MIX", "Stereo DAC MIX" }, + { "DAC12 SRC Mux", "MONO DAC MIX", "Mono DAC MIX" }, + { "DAC12 SRC Mux", "DD MIX1", "DD1 MIX" }, + { "DAC12 SRC Mux", "DD MIX2", "DD2 MIX" }, + + { "DAC3 SRC Mux", "MONO DAC MIXL", "Mono DAC MIXL" }, + { "DAC3 SRC Mux", "MONO DAC MIXR", "Mono DAC MIXR" }, + { "DAC3 SRC Mux", "DD MIX1L", "DD1 MIXL" }, + { "DAC3 SRC Mux", "DD MIX2L", "DD2 MIXL" }, + + { "DAC 1", NULL, "DAC12 SRC Mux" }, + { "DAC 1", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC 2", NULL, "DAC12 SRC Mux" }, + { "DAC 2", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC 3", NULL, "DAC3 SRC Mux" }, + { "DAC 3", NULL, "PLL1", is_sys_clk_from_pll }, + + { "PDM1 L Mux", "STO1 DAC MIX", "Stereo DAC MIXL" }, + { "PDM1 L Mux", "MONO DAC MIX", "Mono DAC MIXL" }, + { "PDM1 L Mux", "DD MIX1", "DD1 MIXL" }, + { "PDM1 L Mux", "DD MIX2", "DD2 MIXL" }, + { "PDM1 L Mux", NULL, "PDM1 Power" }, + { "PDM1 R Mux", "STO1 DAC MIX", "Stereo DAC MIXR" }, + { "PDM1 R Mux", "MONO DAC MIX", "Mono DAC MIXR" }, + { "PDM1 R Mux", "DD MIX1", "DD1 MIXR" }, + { "PDM1 R Mux", "DD MIX2", "DD2 MIXR" }, + { "PDM1 R Mux", NULL, "PDM1 Power" }, + { "PDM2 L Mux", "STO1 DAC MIX", "Stereo DAC MIXL" }, + { "PDM2 L Mux", "MONO DAC MIX", "Mono DAC MIXL" }, + { "PDM2 L Mux", "DD MIX1", "DD1 MIXL" }, + { "PDM2 L Mux", "DD MIX2", "DD2 MIXL" }, + { "PDM2 L Mux", NULL, "PDM2 Power" }, + { "PDM2 R Mux", "STO1 DAC MIX", "Stereo DAC MIXR" }, + { "PDM2 R Mux", "MONO DAC MIX", "Mono DAC MIXR" }, + { "PDM2 R Mux", "DD MIX1", "DD1 MIXR" }, + { "PDM2 R Mux", "DD MIX1", "DD2 MIXR" }, + { "PDM2 R Mux", NULL, "PDM2 Power" }, + + { "LOUT1 amp", NULL, "DAC 1" }, + { "LOUT2 amp", NULL, "DAC 2" }, + { "LOUT3 amp", NULL, "DAC 3" }, + + { "LOUT1", NULL, "LOUT1 amp" }, + { "LOUT2", NULL, "LOUT2 amp" }, + { "LOUT3", NULL, "LOUT3 amp" }, + + { "PDM1L", NULL, "PDM1 L Mux" }, + { "PDM1R", NULL, "PDM1 R Mux" }, + { "PDM2L", NULL, "PDM2 L Mux" }, + { "PDM2R", NULL, "PDM2 R Mux" }, +}; + +static int get_clk_info(int sclk, int rate) +{ + int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +static int rt5677_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + rt5677->lrck[dai->id] = params_rate(params); + pre_div = get_clk_info(rt5677->sysclk, rt5677->lrck[dai->id]); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting\n"); + return -EINVAL; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + bclk_ms = frame_size > 32; + rt5677->bclk[dai->id] = rt5677->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5677->bclk[dai->id], rt5677->lrck[dai->id]); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= RT5677_I2S_DL_20; + break; + case 24: + val_len |= RT5677_I2S_DL_24; + break; + case 8: + val_len |= RT5677_I2S_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5677_AIF1: + mask_clk = RT5677_I2S_PD1_MASK; + val_clk = pre_div << RT5677_I2S_PD1_SFT; + regmap_update_bits(rt5677->regmap, RT5677_I2S1_SDP, + RT5677_I2S_DL_MASK, val_len); + regmap_update_bits(rt5677->regmap, RT5677_CLK_TREE_CTRL1, + mask_clk, val_clk); + break; + case RT5677_AIF2: + mask_clk = RT5677_I2S_PD2_MASK; + val_clk = pre_div << RT5677_I2S_PD2_SFT; + regmap_update_bits(rt5677->regmap, RT5677_I2S2_SDP, + RT5677_I2S_DL_MASK, val_len); + regmap_update_bits(rt5677->regmap, RT5677_CLK_TREE_CTRL1, + mask_clk, val_clk); + break; + case RT5677_AIF3: + mask_clk = RT5677_I2S_BCLK_MS3_MASK | RT5677_I2S_PD3_MASK; + val_clk = bclk_ms << RT5677_I2S_BCLK_MS3_SFT | + pre_div << RT5677_I2S_PD3_SFT; + regmap_update_bits(rt5677->regmap, RT5677_I2S3_SDP, + RT5677_I2S_DL_MASK, val_len); + regmap_update_bits(rt5677->regmap, RT5677_CLK_TREE_CTRL1, + mask_clk, val_clk); + break; + case RT5677_AIF4: + mask_clk = RT5677_I2S_BCLK_MS4_MASK | RT5677_I2S_PD4_MASK; + val_clk = bclk_ms << RT5677_I2S_BCLK_MS4_SFT | + pre_div << RT5677_I2S_PD4_SFT; + regmap_update_bits(rt5677->regmap, RT5677_I2S4_SDP, + RT5677_I2S_DL_MASK, val_len); + regmap_update_bits(rt5677->regmap, RT5677_CLK_TREE_CTRL1, + mask_clk, val_clk); + break; + default: + break; + } + + return 0; +} + +static int rt5677_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5677->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5677_I2S_MS_S; + rt5677->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5677_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5677_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5677_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5677_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5677_AIF1: + regmap_update_bits(rt5677->regmap, RT5677_I2S1_SDP, + RT5677_I2S_MS_MASK | RT5677_I2S_BP_MASK | + RT5677_I2S_DF_MASK, reg_val); + break; + case RT5677_AIF2: + regmap_update_bits(rt5677->regmap, RT5677_I2S2_SDP, + RT5677_I2S_MS_MASK | RT5677_I2S_BP_MASK | + RT5677_I2S_DF_MASK, reg_val); + break; + case RT5677_AIF3: + regmap_update_bits(rt5677->regmap, RT5677_I2S3_SDP, + RT5677_I2S_MS_MASK | RT5677_I2S_BP_MASK | + RT5677_I2S_DF_MASK, reg_val); + break; + case RT5677_AIF4: + regmap_update_bits(rt5677->regmap, RT5677_I2S4_SDP, + RT5677_I2S_MS_MASK | RT5677_I2S_BP_MASK | + RT5677_I2S_DF_MASK, reg_val); + break; + default: + break; + } + + + return 0; +} + +static int rt5677_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5677->sysclk && clk_id == rt5677->sysclk_src) + return 0; + + switch (clk_id) { + case RT5677_SCLK_S_MCLK: + reg_val |= RT5677_SCLK_SRC_MCLK; + break; + case RT5677_SCLK_S_PLL1: + reg_val |= RT5677_SCLK_SRC_PLL1; + break; + case RT5677_SCLK_S_RCCLK: + reg_val |= RT5677_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_SCLK_SRC_MASK, reg_val); + rt5677->sysclk = freq; + rt5677->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +/** + * rt5677_pll_calc - Calcualte PLL M/N/K code. + * @freq_in: external clock provided to codec. + * @freq_out: target clock which codec works on. + * @pll_code: Pointer to structure with M, N, K, bypass K and bypass M flag. + * + * Calcualte M/N/K code and bypass K/M flag to configure PLL for codec. + * + * Returns 0 for success or negative error code. + */ +static int rt5677_pll_calc(const unsigned int freq_in, + const unsigned int freq_out, struct rt5677_pll_code *pll_code) +{ + int max_n = RT5677_PLL_N_MAX, max_m = RT5677_PLL_M_MAX; + int k, red, n_t, pll_out, in_t; + int n = 0, m = 0, m_t = 0; + int out_t, red_t = abs(freq_out - freq_in); + bool m_bp = false, k_bp = false; + + if (RT5677_PLL_INP_MAX < freq_in || RT5677_PLL_INP_MIN > freq_in) + return -EINVAL; + + k = 100000000 / freq_out - 2; + if (k > RT5677_PLL_K_MAX) + k = RT5677_PLL_K_MAX; + for (n_t = 0; n_t <= max_n; n_t++) { + in_t = freq_in / (k + 2); + pll_out = freq_out / (n_t + 2); + if (in_t < 0) + continue; + if (in_t == pll_out) { + m_bp = true; + n = n_t; + goto code_find; + } + red = abs(in_t - pll_out); + if (red < red_t) { + m_bp = true; + n = n_t; + m = m_t; + if (red == 0) + goto code_find; + red_t = red; + } + for (m_t = 0; m_t <= max_m; m_t++) { + out_t = in_t / (m_t + 2); + red = abs(out_t - pll_out); + if (red < red_t) { + m_bp = false; + n = n_t; + m = m_t; + if (red == 0) + goto code_find; + red_t = red; + } + } + } + pr_debug("Only get approximation about PLL\n"); + +code_find: + + pll_code->m_bp = m_bp; + pll_code->k_bp = k_bp; + pll_code->m_code = m; + pll_code->n_code = n; + pll_code->k_code = k; + return 0; +} + +static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + struct rt5677_pll_code pll_code; + int ret; + + if (source == rt5677->pll_src && freq_in == rt5677->pll_in && + freq_out == rt5677->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5677->pll_in = 0; + rt5677->pll_out = 0; + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_SCLK_SRC_MASK, RT5677_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5677_PLL1_S_MCLK: + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_PLL1_SRC_MASK, RT5677_PLL1_SRC_MCLK); + break; + case RT5677_PLL1_S_BCLK1: + case RT5677_PLL1_S_BCLK2: + case RT5677_PLL1_S_BCLK3: + case RT5677_PLL1_S_BCLK4: + switch (dai->id) { + case RT5677_AIF1: + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_PLL1_SRC_MASK, RT5677_PLL1_SRC_BCLK1); + break; + case RT5677_AIF2: + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_PLL1_SRC_MASK, RT5677_PLL1_SRC_BCLK2); + break; + case RT5677_AIF3: + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_PLL1_SRC_MASK, RT5677_PLL1_SRC_BCLK3); + break; + case RT5677_AIF4: + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_PLL1_SRC_MASK, RT5677_PLL1_SRC_BCLK4); + break; + default: + break; + } + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rt5677_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "m_bypass=%d k_bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, pll_code.k_bp, + (pll_code.m_bp ? 0 : pll_code.m_code), pll_code.n_code, + (pll_code.k_bp ? 0 : pll_code.k_code)); + + regmap_write(rt5677->regmap, RT5677_PLL1_CTRL1, + pll_code.n_code << RT5677_PLL_N_SFT | + pll_code.k_bp << RT5677_PLL_K_BP_SFT | + (pll_code.k_bp ? 0 : pll_code.k_code)); + regmap_write(rt5677->regmap, RT5677_PLL1_CTRL2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5677_PLL_M_SFT | + pll_code.m_bp << RT5677_PLL_M_BP_SFT); + + rt5677->pll_in = freq_in; + rt5677->pll_out = freq_out; + rt5677->pll_src = source; + + return 0; +} + +static int rt5677_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1, + RT5677_LDO1_SEL_MASK | RT5677_LDO2_SEL_MASK, + 0x0055); + regmap_update_bits(rt5677->regmap, + RT5677_PR_BASE + RT5677_BIAS_CUR4, + 0x0f00, 0x0f00); + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1, + RT5677_PWR_VREF1 | RT5677_PWR_MB | + RT5677_PWR_BG | RT5677_PWR_VREF2, + RT5677_PWR_VREF1 | RT5677_PWR_MB | + RT5677_PWR_BG | RT5677_PWR_VREF2); + mdelay(20); + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1, + RT5677_PWR_FV1 | RT5677_PWR_FV2, + RT5677_PWR_FV1 | RT5677_PWR_FV2); + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, + RT5677_PWR_CORE, RT5677_PWR_CORE); + regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, + 0x1, 0x1); + } + break; + + case SND_SOC_BIAS_STANDBY: + break; + + case SND_SOC_BIAS_OFF: + regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x1, 0x0); + regmap_write(rt5677->regmap, RT5677_PWR_DIG1, 0x0000); + regmap_write(rt5677->regmap, RT5677_PWR_DIG2, 0x0000); + regmap_write(rt5677->regmap, RT5677_PWR_ANLG1, 0x0000); + regmap_write(rt5677->regmap, RT5677_PWR_ANLG2, 0x0000); + regmap_update_bits(rt5677->regmap, + RT5677_PR_BASE + RT5677_BIAS_CUR4, 0x0f00, 0x0000); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5677_probe(struct snd_soc_codec *codec) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + rt5677->codec = codec; + + rt5677_set_bias_level(codec, SND_SOC_BIAS_OFF); + + regmap_write(rt5677->regmap, RT5677_DIG_MISC, 0x0020); + regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x0c00); + + return 0; +} + +static int rt5677_remove(struct snd_soc_codec *codec) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); + + return 0; +} + +#ifdef CONFIG_PM +static int rt5677_suspend(struct snd_soc_codec *codec) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5677->regmap, true); + regcache_mark_dirty(rt5677->regmap); + + return 0; +} + +static int rt5677_resume(struct snd_soc_codec *codec) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5677->regmap, false); + regcache_sync(rt5677->regmap); + + return 0; +} +#else +#define rt5677_suspend NULL +#define rt5677_resume NULL +#endif + +#define RT5677_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5677_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static struct snd_soc_dai_ops rt5677_aif_dai_ops = { + .hw_params = rt5677_hw_params, + .set_fmt = rt5677_set_dai_fmt, + .set_sysclk = rt5677_set_dai_sysclk, + .set_pll = rt5677_set_dai_pll, +}; + +static struct snd_soc_dai_driver rt5677_dai[] = { + { + .name = "rt5677-aif1", + .id = RT5677_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5677_STEREO_RATES, + .formats = RT5677_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5677_STEREO_RATES, + .formats = RT5677_FORMATS, + }, + .ops = &rt5677_aif_dai_ops, + }, + { + .name = "rt5677-aif2", + .id = RT5677_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5677_STEREO_RATES, + .formats = RT5677_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5677_STEREO_RATES, + .formats = RT5677_FORMATS, + }, + .ops = &rt5677_aif_dai_ops, + }, + { + .name = "rt5677-aif3", + .id = RT5677_AIF3, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5677_STEREO_RATES, + .formats = RT5677_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5677_STEREO_RATES, + .formats = RT5677_FORMATS, + }, + .ops = &rt5677_aif_dai_ops, + }, + { + .name = "rt5677-aif4", + .id = RT5677_AIF4, + .playback = { + .stream_name = "AIF4 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5677_STEREO_RATES, + .formats = RT5677_FORMATS, + }, + .capture = { + .stream_name = "AIF4 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5677_STEREO_RATES, + .formats = RT5677_FORMATS, + }, + .ops = &rt5677_aif_dai_ops, + }, + { + .name = "rt5677-slimbus", + .id = RT5677_AIF5, + .playback = { + .stream_name = "SLIMBus Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5677_STEREO_RATES, + .formats = RT5677_FORMATS, + }, + .capture = { + .stream_name = "SLIMBus Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5677_STEREO_RATES, + .formats = RT5677_FORMATS, + }, + .ops = &rt5677_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5677 = { + .probe = rt5677_probe, + .remove = rt5677_remove, + .suspend = rt5677_suspend, + .resume = rt5677_resume, + .set_bias_level = rt5677_set_bias_level, + .idle_bias_off = true, + .controls = rt5677_snd_controls, + .num_controls = ARRAY_SIZE(rt5677_snd_controls), + .dapm_widgets = rt5677_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5677_dapm_widgets), + .dapm_routes = rt5677_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5677_dapm_routes), +}; + +static const struct regmap_config rt5677_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = RT5677_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5677_ranges) * + RT5677_PR_SPACING), + + .volatile_reg = rt5677_volatile_register, + .readable_reg = rt5677_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5677_reg, + .num_reg_defaults = ARRAY_SIZE(rt5677_reg), + .ranges = rt5677_ranges, + .num_ranges = ARRAY_SIZE(rt5677_ranges), +}; + +static const struct i2c_device_id rt5677_i2c_id[] = { + { "rt5677", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); + +static int rt5677_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5677_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5677_priv *rt5677; + int ret; + unsigned int val; + + rt5677 = devm_kzalloc(&i2c->dev, sizeof(struct rt5677_priv), + GFP_KERNEL); + if (rt5677 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5677); + + if (pdata) + rt5677->pdata = *pdata; + + rt5677->regmap = devm_regmap_init_i2c(i2c, &rt5677_regmap); + if (IS_ERR(rt5677->regmap)) { + ret = PTR_ERR(rt5677->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5677->regmap, RT5677_VENDOR_ID2, &val); + if (val != RT5677_DEVICE_ID) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt5677\n", val); + return -ENODEV; + } + + regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); + + ret = regmap_register_patch(rt5677->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5677->pdata.in1_diff) + regmap_update_bits(rt5677->regmap, RT5677_IN1, + RT5677_IN_DF1, RT5677_IN_DF1); + + if (rt5677->pdata.in2_diff) + regmap_update_bits(rt5677->regmap, RT5677_IN1, + RT5677_IN_DF2, RT5677_IN_DF2); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677, + rt5677_dai, ARRAY_SIZE(rt5677_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt5677_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +static struct i2c_driver rt5677_i2c_driver = { + .driver = { + .name = "rt5677", + .owner = THIS_MODULE, + }, + .probe = rt5677_i2c_probe, + .remove = rt5677_i2c_remove, + .id_table = rt5677_i2c_id, +}; + +static int __init rt5677_modinit(void) +{ + return i2c_add_driver(&rt5677_i2c_driver); +} +module_init(rt5677_modinit); + +static void __exit rt5677_modexit(void) +{ + i2c_del_driver(&rt5677_i2c_driver); +} +module_exit(rt5677_modexit); + +MODULE_DESCRIPTION("ASoC RT5677 driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h new file mode 100644 index 0000000..af4e9c7 --- /dev/null +++ b/sound/soc/codecs/rt5677.h @@ -0,0 +1,1451 @@ +/* + * rt5677.h -- RT5677 ALSA SoC audio driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5677_H__ +#define __RT5677_H__ + +#include <sound/rt5677.h> + +/* Info */ +#define RT5677_RESET 0x00 +#define RT5677_VENDOR_ID 0xfd +#define RT5677_VENDOR_ID1 0xfe +#define RT5677_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5677_LOUT1 0x01 +/* I/O - Input */ +#define RT5677_IN1 0x03 +#define RT5677_MICBIAS 0x04 +/* I/O - SLIMBus */ +#define RT5677_SLIMBUS_PARAM 0x07 +#define RT5677_SLIMBUS_RX 0x08 +#define RT5677_SLIMBUS_CTRL 0x09 +/* I/O */ +#define RT5677_SIDETONE_CTRL 0x13 +/* I/O - ADC/DAC */ +#define RT5677_ANA_DAC1_2_3_SRC 0x15 +#define RT5677_IF_DSP_DAC3_4_MIXER 0x16 +#define RT5677_DAC4_DIG_VOL 0x17 +#define RT5677_DAC3_DIG_VOL 0x18 +#define RT5677_DAC1_DIG_VOL 0x19 +#define RT5677_DAC2_DIG_VOL 0x1a +#define RT5677_IF_DSP_DAC2_MIXER 0x1b +#define RT5677_STO1_ADC_DIG_VOL 0x1c +#define RT5677_MONO_ADC_DIG_VOL 0x1d +#define RT5677_STO1_2_ADC_BST 0x1e +#define RT5677_STO2_ADC_DIG_VOL 0x1f +/* Mixer - D-D */ +#define RT5677_ADC_BST_CTRL2 0x20 +#define RT5677_STO3_4_ADC_BST 0x21 +#define RT5677_STO3_ADC_DIG_VOL 0x22 +#define RT5677_STO4_ADC_DIG_VOL 0x23 +#define RT5677_STO4_ADC_MIXER 0x24 +#define RT5677_STO3_ADC_MIXER 0x25 +#define RT5677_STO2_ADC_MIXER 0x26 +#define RT5677_STO1_ADC_MIXER 0x27 +#define RT5677_MONO_ADC_MIXER 0x28 +#define RT5677_ADC_IF_DSP_DAC1_MIXER 0x29 +#define RT5677_STO1_DAC_MIXER 0x2a +#define RT5677_MONO_DAC_MIXER 0x2b +#define RT5677_DD1_MIXER 0x2c +#define RT5677_DD2_MIXER 0x2d +#define RT5677_IF3_DATA 0x2f +#define RT5677_IF4_DATA 0x30 +/* Mixer - PDM */ +#define RT5677_PDM_OUT_CTRL 0x31 +#define RT5677_PDM_DATA_CTRL1 0x32 +#define RT5677_PDM_DATA_CTRL2 0x33 +#define RT5677_PDM1_DATA_CTRL2 0x34 +#define RT5677_PDM1_DATA_CTRL3 0x35 +#define RT5677_PDM1_DATA_CTRL4 0x36 +#define RT5677_PDM2_DATA_CTRL2 0x37 +#define RT5677_PDM2_DATA_CTRL3 0x38 +#define RT5677_PDM2_DATA_CTRL4 0x39 +/* TDM */ +#define RT5677_TDM1_CTRL1 0x3b +#define RT5677_TDM1_CTRL2 0x3c +#define RT5677_TDM1_CTRL3 0x3d +#define RT5677_TDM1_CTRL4 0x3e +#define RT5677_TDM1_CTRL5 0x3f +#define RT5677_TDM2_CTRL1 0x40 +#define RT5677_TDM2_CTRL2 0x41 +#define RT5677_TDM2_CTRL3 0x42 +#define RT5677_TDM2_CTRL4 0x43 +#define RT5677_TDM2_CTRL5 0x44 +/* I2C_MASTER_CTRL */ +#define RT5677_I2C_MASTER_CTRL1 0x47 +#define RT5677_I2C_MASTER_CTRL2 0x48 +#define RT5677_I2C_MASTER_CTRL3 0x49 +#define RT5677_I2C_MASTER_CTRL4 0x4a +#define RT5677_I2C_MASTER_CTRL5 0x4b +#define RT5677_I2C_MASTER_CTRL6 0x4c +#define RT5677_I2C_MASTER_CTRL7 0x4d +#define RT5677_I2C_MASTER_CTRL8 0x4e +/* DMIC */ +#define RT5677_DMIC_CTRL1 0x50 +#define RT5677_DMIC_CTRL2 0x51 +/* Haptic Generator */ +#define RT5677_HAP_GENE_CTRL1 0x56 +#define RT5677_HAP_GENE_CTRL2 0x57 +#define RT5677_HAP_GENE_CTRL3 0x58 +#define RT5677_HAP_GENE_CTRL4 0x59 +#define RT5677_HAP_GENE_CTRL5 0x5a +#define RT5677_HAP_GENE_CTRL6 0x5b +#define RT5677_HAP_GENE_CTRL7 0x5c +#define RT5677_HAP_GENE_CTRL8 0x5d +#define RT5677_HAP_GENE_CTRL9 0x5e +#define RT5677_HAP_GENE_CTRL10 0x5f +/* Power */ +#define RT5677_PWR_DIG1 0x61 +#define RT5677_PWR_DIG2 0x62 +#define RT5677_PWR_ANLG1 0x63 +#define RT5677_PWR_ANLG2 0x64 +#define RT5677_PWR_DSP1 0x65 +#define RT5677_PWR_DSP_ST 0x66 +#define RT5677_PWR_DSP2 0x67 +#define RT5677_ADC_DAC_HPF_CTRL1 0x68 +/* Private Register Control */ +#define RT5677_PRIV_INDEX 0x6a +#define RT5677_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5677_I2S4_SDP 0x6f +#define RT5677_I2S1_SDP 0x70 +#define RT5677_I2S2_SDP 0x71 +#define RT5677_I2S3_SDP 0x72 +#define RT5677_CLK_TREE_CTRL1 0x73 +#define RT5677_CLK_TREE_CTRL2 0x74 +#define RT5677_CLK_TREE_CTRL3 0x75 +/* Function - Analog */ +#define RT5677_PLL1_CTRL1 0x7a +#define RT5677_PLL1_CTRL2 0x7b +#define RT5677_PLL2_CTRL1 0x7c +#define RT5677_PLL2_CTRL2 0x7d +#define RT5677_GLB_CLK1 0x80 +#define RT5677_GLB_CLK2 0x81 +#define RT5677_ASRC_1 0x83 +#define RT5677_ASRC_2 0x84 +#define RT5677_ASRC_3 0x85 +#define RT5677_ASRC_4 0x86 +#define RT5677_ASRC_5 0x87 +#define RT5677_ASRC_6 0x88 +#define RT5677_ASRC_7 0x89 +#define RT5677_ASRC_8 0x8a +#define RT5677_ASRC_9 0x8b +#define RT5677_ASRC_10 0x8c +#define RT5677_ASRC_11 0x8d +#define RT5677_ASRC_12 0x8e +#define RT5677_ASRC_13 0x8f +#define RT5677_ASRC_14 0x90 +#define RT5677_ASRC_15 0x91 +#define RT5677_ASRC_16 0x92 +#define RT5677_ASRC_17 0x93 +#define RT5677_ASRC_18 0x94 +#define RT5677_ASRC_19 0x95 +#define RT5677_ASRC_20 0x97 +#define RT5677_ASRC_21 0x98 +#define RT5677_ASRC_22 0x99 +#define RT5677_ASRC_23 0x9a +#define RT5677_VAD_CTRL1 0x9c +#define RT5677_VAD_CTRL2 0x9d +#define RT5677_VAD_CTRL3 0x9e +#define RT5677_VAD_CTRL4 0x9f +#define RT5677_VAD_CTRL5 0xa0 +/* Function - Digital */ +#define RT5677_DSP_INB_CTRL1 0xa3 +#define RT5677_DSP_INB_CTRL2 0xa4 +#define RT5677_DSP_IN_OUTB_CTRL 0xa5 +#define RT5677_DSP_OUTB0_1_DIG_VOL 0xa6 +#define RT5677_DSP_OUTB2_3_DIG_VOL 0xa7 +#define RT5677_DSP_OUTB4_5_DIG_VOL 0xa8 +#define RT5677_DSP_OUTB6_7_DIG_VOL 0xa9 +#define RT5677_ADC_EQ_CTRL1 0xae +#define RT5677_ADC_EQ_CTRL2 0xaf +#define RT5677_EQ_CTRL1 0xb0 +#define RT5677_EQ_CTRL2 0xb1 +#define RT5677_EQ_CTRL3 0xb2 +#define RT5677_SOFT_VOL_ZERO_CROSS1 0xb3 +#define RT5677_JD_CTRL1 0xb5 +#define RT5677_JD_CTRL2 0xb6 +#define RT5677_JD_CTRL3 0xb8 +#define RT5677_IRQ_CTRL1 0xbd +#define RT5677_IRQ_CTRL2 0xbe +#define RT5677_GPIO_ST 0xbf +#define RT5677_GPIO_CTRL1 0xc0 +#define RT5677_GPIO_CTRL2 0xc1 +#define RT5677_GPIO_CTRL3 0xc2 +#define RT5677_STO1_ADC_HI_FILTER1 0xc5 +#define RT5677_STO1_ADC_HI_FILTER2 0xc6 +#define RT5677_MONO_ADC_HI_FILTER1 0xc7 +#define RT5677_MONO_ADC_HI_FILTER2 0xc8 +#define RT5677_STO2_ADC_HI_FILTER1 0xc9 +#define RT5677_STO2_ADC_HI_FILTER2 0xca +#define RT5677_STO3_ADC_HI_FILTER1 0xcb +#define RT5677_STO3_ADC_HI_FILTER2 0xcc +#define RT5677_STO4_ADC_HI_FILTER1 0xcd +#define RT5677_STO4_ADC_HI_FILTER2 0xce +#define RT5677_MB_DRC_CTRL1 0xd0 +#define RT5677_DRC1_CTRL1 0xd2 +#define RT5677_DRC1_CTRL2 0xd3 +#define RT5677_DRC1_CTRL3 0xd4 +#define RT5677_DRC1_CTRL4 0xd5 +#define RT5677_DRC1_CTRL5 0xd6 +#define RT5677_DRC1_CTRL6 0xd7 +#define RT5677_DRC2_CTRL1 0xd8 +#define RT5677_DRC2_CTRL2 0xd9 +#define RT5677_DRC2_CTRL3 0xda +#define RT5677_DRC2_CTRL4 0xdb +#define RT5677_DRC2_CTRL5 0xdc +#define RT5677_DRC2_CTRL6 0xdd +#define RT5677_DRC1_HL_CTRL1 0xde +#define RT5677_DRC1_HL_CTRL2 0xdf +#define RT5677_DRC2_HL_CTRL1 0xe0 +#define RT5677_DRC2_HL_CTRL2 0xe1 +#define RT5677_DSP_INB1_SRC_CTRL1 0xe3 +#define RT5677_DSP_INB1_SRC_CTRL2 0xe4 +#define RT5677_DSP_INB1_SRC_CTRL3 0xe5 +#define RT5677_DSP_INB1_SRC_CTRL4 0xe6 +#define RT5677_DSP_INB2_SRC_CTRL1 0xe7 +#define RT5677_DSP_INB2_SRC_CTRL2 0xe8 +#define RT5677_DSP_INB2_SRC_CTRL3 0xe9 +#define RT5677_DSP_INB2_SRC_CTRL4 0xea +#define RT5677_DSP_INB3_SRC_CTRL1 0xeb +#define RT5677_DSP_INB3_SRC_CTRL2 0xec +#define RT5677_DSP_INB3_SRC_CTRL3 0xed +#define RT5677_DSP_INB3_SRC_CTRL4 0xee +#define RT5677_DSP_OUTB1_SRC_CTRL1 0xef +#define RT5677_DSP_OUTB1_SRC_CTRL2 0xf0 +#define RT5677_DSP_OUTB1_SRC_CTRL3 0xf1 +#define RT5677_DSP_OUTB1_SRC_CTRL4 0xf2 +#define RT5677_DSP_OUTB2_SRC_CTRL1 0xf3 +#define RT5677_DSP_OUTB2_SRC_CTRL2 0xf4 +#define RT5677_DSP_OUTB2_SRC_CTRL3 0xf5 +#define RT5677_DSP_OUTB2_SRC_CTRL4 0xf6 + +/* Virtual DSP Mixer Control */ +#define RT5677_DSP_OUTB_0123_MIXER_CTRL 0xf7 +#define RT5677_DSP_OUTB_45_MIXER_CTRL 0xf8 +#define RT5677_DSP_OUTB_67_MIXER_CTRL 0xf9 + +/* General Control */ +#define RT5677_DIG_MISC 0xfa +#define RT5677_GEN_CTRL1 0xfb +#define RT5677_GEN_CTRL2 0xfc + +/* DSP Mode I2C Control*/ +#define RT5677_DSP_I2C_OP_CODE 0x00 +#define RT5677_DSP_I2C_ADDR_LSB 0x01 +#define RT5677_DSP_I2C_ADDR_MSB 0x02 +#define RT5677_DSP_I2C_DATA_LSB 0x03 +#define RT5677_DSP_I2C_DATA_MSB 0x04 + +/* Index of Codec Private Register definition */ +#define RT5677_PR_DRC1_CTRL_1 0x01 +#define RT5677_PR_DRC1_CTRL_2 0x02 +#define RT5677_PR_DRC1_CTRL_3 0x03 +#define RT5677_PR_DRC1_CTRL_4 0x04 +#define RT5677_PR_DRC1_CTRL_5 0x05 +#define RT5677_PR_DRC1_CTRL_6 0x06 +#define RT5677_PR_DRC1_CTRL_7 0x07 +#define RT5677_PR_DRC2_CTRL_1 0x08 +#define RT5677_PR_DRC2_CTRL_2 0x09 +#define RT5677_PR_DRC2_CTRL_3 0x0a +#define RT5677_PR_DRC2_CTRL_4 0x0b +#define RT5677_PR_DRC2_CTRL_5 0x0c +#define RT5677_PR_DRC2_CTRL_6 0x0d +#define RT5677_PR_DRC2_CTRL_7 0x0e +#define RT5677_BIAS_CUR1 0x10 +#define RT5677_BIAS_CUR2 0x12 +#define RT5677_BIAS_CUR3 0x13 +#define RT5677_BIAS_CUR4 0x14 +#define RT5677_BIAS_CUR5 0x15 +#define RT5677_VREF_LOUT_CTRL 0x17 +#define RT5677_DIG_VOL_CTRL1 0x1a +#define RT5677_DIG_VOL_CTRL2 0x1b +#define RT5677_ANA_ADC_GAIN_CTRL 0x1e +#define RT5677_VAD_SRAM_TEST1 0x20 +#define RT5677_VAD_SRAM_TEST2 0x21 +#define RT5677_VAD_SRAM_TEST3 0x22 +#define RT5677_VAD_SRAM_TEST4 0x23 +#define RT5677_PAD_DRV_CTRL 0x26 +#define RT5677_DIG_IN_PIN_ST_CTRL1 0x29 +#define RT5677_DIG_IN_PIN_ST_CTRL2 0x2a +#define RT5677_DIG_IN_PIN_ST_CTRL3 0x2b +#define RT5677_PLL1_INT 0x38 +#define RT5677_PLL2_INT 0x39 +#define RT5677_TEST_CTRL1 0x3a +#define RT5677_TEST_CTRL2 0x3b +#define RT5677_TEST_CTRL3 0x3c +#define RT5677_CHOP_DAC_ADC 0x3d +#define RT5677_SOFT_DEPOP_DAC_CLK_CTRL 0x3e +#define RT5677_CROSS_OVER_FILTER1 0x90 +#define RT5677_CROSS_OVER_FILTER2 0x91 +#define RT5677_CROSS_OVER_FILTER3 0x92 +#define RT5677_CROSS_OVER_FILTER4 0x93 +#define RT5677_CROSS_OVER_FILTER5 0x94 +#define RT5677_CROSS_OVER_FILTER6 0x95 +#define RT5677_CROSS_OVER_FILTER7 0x96 +#define RT5677_CROSS_OVER_FILTER8 0x97 +#define RT5677_CROSS_OVER_FILTER9 0x98 +#define RT5677_CROSS_OVER_FILTER10 0x99 + +/* global definition */ +#define RT5677_L_MUTE (0x1 << 15) +#define RT5677_L_MUTE_SFT 15 +#define RT5677_VOL_L_MUTE (0x1 << 14) +#define RT5677_VOL_L_SFT 14 +#define RT5677_R_MUTE (0x1 << 7) +#define RT5677_R_MUTE_SFT 7 +#define RT5677_VOL_R_MUTE (0x1 << 6) +#define RT5677_VOL_R_SFT 6 +#define RT5677_L_VOL_MASK (0x3f << 8) +#define RT5677_L_VOL_SFT 8 +#define RT5677_R_VOL_MASK (0x3f) +#define RT5677_R_VOL_SFT 0 + +/* LOUT1 Control (0x01) */ +#define RT5677_LOUT1_L_MUTE (0x1 << 15) +#define RT5677_LOUT1_L_MUTE_SFT (15) +#define RT5677_LOUT1_L_DF (0x1 << 14) +#define RT5677_LOUT1_L_DF_SFT (14) +#define RT5677_LOUT2_L_MUTE (0x1 << 13) +#define RT5677_LOUT2_L_MUTE_SFT (13) +#define RT5677_LOUT2_L_DF (0x1 << 12) +#define RT5677_LOUT2_L_DF_SFT (12) +#define RT5677_LOUT3_L_MUTE (0x1 << 11) +#define RT5677_LOUT3_L_MUTE_SFT (11) +#define RT5677_LOUT3_L_DF (0x1 << 10) +#define RT5677_LOUT3_L_DF_SFT (10) +#define RT5677_LOUT1_ENH_DRV (0x1 << 9) +#define RT5677_LOUT1_ENH_DRV_SFT (9) +#define RT5677_LOUT2_ENH_DRV (0x1 << 8) +#define RT5677_LOUT2_ENH_DRV_SFT (8) +#define RT5677_LOUT3_ENH_DRV (0x1 << 7) +#define RT5677_LOUT3_ENH_DRV_SFT (7) + +/* IN1 Control (0x03) */ +#define RT5677_BST_MASK1 (0xf << 12) +#define RT5677_BST_SFT1 12 +#define RT5677_BST_MASK2 (0xf << 8) +#define RT5677_BST_SFT2 8 +#define RT5677_IN_DF1 (0x1 << 7) +#define RT5677_IN_DF1_SFT 7 +#define RT5677_IN_DF2 (0x1 << 6) +#define RT5677_IN_DF2_SFT 6 + +/* Micbias Control (0x04) */ +#define RT5677_MICBIAS1_OUTVOLT_MASK (0x1 << 15) +#define RT5677_MICBIAS1_OUTVOLT_SFT (15) +#define RT5677_MICBIAS1_OUTVOLT_2_7V (0x0 << 15) +#define RT5677_MICBIAS1_OUTVOLT_2_25V (0x1 << 15) +#define RT5677_MICBIAS1_CTRL_VDD_MASK (0x1 << 14) +#define RT5677_MICBIAS1_CTRL_VDD_SFT (14) +#define RT5677_MICBIAS1_CTRL_VDD_1_8V (0x0 << 14) +#define RT5677_MICBIAS1_CTRL_VDD_3_3V (0x1 << 14) +#define RT5677_MICBIAS1_OVCD_MASK (0x1 << 11) +#define RT5677_MICBIAS1_OVCD_SHIFT (11) +#define RT5677_MICBIAS1_OVCD_DIS (0x0 << 11) +#define RT5677_MICBIAS1_OVCD_EN (0x1 << 11) +#define RT5677_MICBIAS1_OVTH_MASK (0x3 << 9) +#define RT5677_MICBIAS1_OVTH_SFT 9 +#define RT5677_MICBIAS1_OVTH_640UA (0x0 << 9) +#define RT5677_MICBIAS1_OVTH_1280UA (0x1 << 9) +#define RT5677_MICBIAS1_OVTH_1920UA (0x2 << 9) + +/* SLIMbus Parameter (0x07) */ + +/* SLIMbus Rx (0x08) */ +#define RT5677_SLB_ADC4_MASK (0x3 << 6) +#define RT5677_SLB_ADC4_SFT 6 +#define RT5677_SLB_ADC3_MASK (0x3 << 4) +#define RT5677_SLB_ADC3_SFT 4 +#define RT5677_SLB_ADC2_MASK (0x3 << 2) +#define RT5677_SLB_ADC2_SFT 2 +#define RT5677_SLB_ADC1_MASK (0x3 << 0) +#define RT5677_SLB_ADC1_SFT 0 + +/* SLIMBus control (0x09) */ + +/* Sidetone Control (0x13) */ +#define RT5677_ST_HPF_SEL_MASK (0x7 << 13) +#define RT5677_ST_HPF_SEL_SFT 13 +#define RT5677_ST_HPF_PATH (0x1 << 12) +#define RT5677_ST_HPF_PATH_SFT 12 +#define RT5677_ST_SEL_MASK (0x7 << 9) +#define RT5677_ST_SEL_SFT 9 +#define RT5677_ST_EN (0x1 << 6) +#define RT5677_ST_EN_SFT 6 + +/* Analog DAC1/2/3 Source Control (0x15) */ +#define RT5677_ANA_DAC3_SRC_SEL_MASK (0x3 << 4) +#define RT5677_ANA_DAC3_SRC_SEL_SFT 4 +#define RT5677_ANA_DAC1_2_SRC_SEL_MASK (0x3 << 0) +#define RT5677_ANA_DAC1_2_SRC_SEL_SFT 0 + +/* IF/DSP to DAC3/4 Mixer Control (0x16) */ +#define RT5677_M_DAC4_L_VOL (0x1 << 15) +#define RT5677_M_DAC4_L_VOL_SFT 15 +#define RT5677_SEL_DAC4_L_SRC_MASK (0x7 << 12) +#define RT5677_SEL_DAC4_L_SRC_SFT 12 +#define RT5677_M_DAC4_R_VOL (0x1 << 11) +#define RT5677_M_DAC4_R_VOL_SFT 11 +#define RT5677_SEL_DAC4_R_SRC_MASK (0x7 << 8) +#define RT5677_SEL_DAC4_R_SRC_SFT 8 +#define RT5677_M_DAC3_L_VOL (0x1 << 7) +#define RT5677_M_DAC3_L_VOL_SFT 7 +#define RT5677_SEL_DAC3_L_SRC_MASK (0x7 << 4) +#define RT5677_SEL_DAC3_L_SRC_SFT 4 +#define RT5677_M_DAC3_R_VOL (0x1 << 3) +#define RT5677_M_DAC3_R_VOL_SFT 3 +#define RT5677_SEL_DAC3_R_SRC_MASK (0x7 << 0) +#define RT5677_SEL_DAC3_R_SRC_SFT 0 + +/* DAC4 Digital Volume (0x17) */ +#define RT5677_DAC4_L_VOL_MASK (0xff << 8) +#define RT5677_DAC4_L_VOL_SFT 8 +#define RT5677_DAC4_R_VOL_MASK (0xff) +#define RT5677_DAC4_R_VOL_SFT 0 + +/* DAC3 Digital Volume (0x18) */ +#define RT5677_DAC3_L_VOL_MASK (0xff << 8) +#define RT5677_DAC3_L_VOL_SFT 8 +#define RT5677_DAC3_R_VOL_MASK (0xff) +#define RT5677_DAC3_R_VOL_SFT 0 + +/* DAC3 Digital Volume (0x19) */ +#define RT5677_DAC1_L_VOL_MASK (0xff << 8) +#define RT5677_DAC1_L_VOL_SFT 8 +#define RT5677_DAC1_R_VOL_MASK (0xff) +#define RT5677_DAC1_R_VOL_SFT 0 + +/* DAC2 Digital Volume (0x1a) */ +#define RT5677_DAC2_L_VOL_MASK (0xff << 8) +#define RT5677_DAC2_L_VOL_SFT 8 +#define RT5677_DAC2_R_VOL_MASK (0xff) +#define RT5677_DAC2_R_VOL_SFT 0 + +/* IF/DSP to DAC2 Mixer Control (0x1b) */ +#define RT5677_M_DAC2_L_VOL (0x1 << 7) +#define RT5677_M_DAC2_L_VOL_SFT 7 +#define RT5677_SEL_DAC2_L_SRC_MASK (0x7 << 4) +#define RT5677_SEL_DAC2_L_SRC_SFT 4 +#define RT5677_M_DAC2_R_VOL (0x1 << 3) +#define RT5677_M_DAC2_R_VOL_SFT 3 +#define RT5677_SEL_DAC2_R_SRC_MASK (0x7 << 0) +#define RT5677_SEL_DAC2_R_SRC_SFT 0 + +/* Stereo1 ADC Digital Volume Control (0x1c) */ +#define RT5677_STO1_ADC_L_VOL_MASK (0x7f << 8) +#define RT5677_STO1_ADC_L_VOL_SFT 8 +#define RT5677_STO1_ADC_R_VOL_MASK (0x7f) +#define RT5677_STO1_ADC_R_VOL_SFT 0 + +/* Mono ADC Digital Volume Control (0x1d) */ +#define RT5677_MONO_ADC_L_VOL_MASK (0x7f << 8) +#define RT5677_MONO_ADC_L_VOL_SFT 8 +#define RT5677_MONO_ADC_R_VOL_MASK (0x7f) +#define RT5677_MONO_ADC_R_VOL_SFT 0 + +/* Stereo 1/2 ADC Boost Gain Control (0x1e) */ +#define RT5677_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5677_STO1_ADC_L_BST_SFT 14 +#define RT5677_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5677_STO1_ADC_R_BST_SFT 12 +#define RT5677_STO1_ADC_COMP_MASK (0x3 << 10) +#define RT5677_STO1_ADC_COMP_SFT 10 +#define RT5677_STO2_ADC_L_BST_MASK (0x3 << 8) +#define RT5677_STO2_ADC_L_BST_SFT 8 +#define RT5677_STO2_ADC_R_BST_MASK (0x3 << 6) +#define RT5677_STO2_ADC_R_BST_SFT 6 +#define RT5677_STO2_ADC_COMP_MASK (0x3 << 4) +#define RT5677_STO2_ADC_COMP_SFT 4 + +/* Stereo2 ADC Digital Volume Control (0x1f) */ +#define RT5677_STO2_ADC_L_VOL_MASK (0x7f << 8) +#define RT5677_STO2_ADC_L_VOL_SFT 8 +#define RT5677_STO2_ADC_R_VOL_MASK (0x7f) +#define RT5677_STO2_ADC_R_VOL_SFT 0 + +/* ADC Boost Gain Control 2 (0x20) */ +#define RT5677_MONO_ADC_L_BST_MASK (0x3 << 14) +#define RT5677_MONO_ADC_L_BST_SFT 14 +#define RT5677_MONO_ADC_R_BST_MASK (0x3 << 12) +#define RT5677_MONO_ADC_R_BST_SFT 12 +#define RT5677_MONO_ADC_COMP_MASK (0x3 << 10) +#define RT5677_MONO_ADC_COMP_SFT 10 + +/* Stereo 3/4 ADC Boost Gain Control (0x21) */ +#define RT5677_STO3_ADC_L_BST_MASK (0x3 << 14) +#define RT5677_STO3_ADC_L_BST_SFT 14 +#define RT5677_STO3_ADC_R_BST_MASK (0x3 << 12) +#define RT5677_STO3_ADC_R_BST_SFT 12 +#define RT5677_STO3_ADC_COMP_MASK (0x3 << 10) +#define RT5677_STO3_ADC_COMP_SFT 10 +#define RT5677_STO4_ADC_L_BST_MASK (0x3 << 8) +#define RT5677_STO4_ADC_L_BST_SFT 8 +#define RT5677_STO4_ADC_R_BST_MASK (0x3 << 6) +#define RT5677_STO4_ADC_R_BST_SFT 6 +#define RT5677_STO4_ADC_COMP_MASK (0x3 << 4) +#define RT5677_STO4_ADC_COMP_SFT 4 + +/* Stereo3 ADC Digital Volume Control (0x22) */ +#define RT5677_STO3_ADC_L_VOL_MASK (0x7f << 8) +#define RT5677_STO3_ADC_L_VOL_SFT 8 +#define RT5677_STO3_ADC_R_VOL_MASK (0x7f) +#define RT5677_STO3_ADC_R_VOL_SFT 0 + +/* Stereo4 ADC Digital Volume Control (0x23) */ +#define RT5677_STO4_ADC_L_VOL_MASK (0x7f << 8) +#define RT5677_STO4_ADC_L_VOL_SFT 8 +#define RT5677_STO4_ADC_R_VOL_MASK (0x7f) +#define RT5677_STO4_ADC_R_VOL_SFT 0 + +/* Stereo4 ADC Mixer control (0x24) */ +#define RT5677_M_STO4_ADC_L2 (0x1 << 15) +#define RT5677_M_STO4_ADC_L2_SFT 15 +#define RT5677_M_STO4_ADC_L1 (0x1 << 14) +#define RT5677_M_STO4_ADC_L1_SFT 14 +#define RT5677_SEL_STO4_ADC1_MASK (0x3 << 12) +#define RT5677_SEL_STO4_ADC1_SFT 12 +#define RT5677_SEL_STO4_ADC2_MASK (0x3 << 10) +#define RT5677_SEL_STO4_ADC2_SFT 10 +#define RT5677_SEL_STO4_DMIC_MASK (0x3 << 8) +#define RT5677_SEL_STO4_DMIC_SFT 8 +#define RT5677_M_STO4_ADC_R1 (0x1 << 7) +#define RT5677_M_STO4_ADC_R1_SFT 7 +#define RT5677_M_STO4_ADC_R2 (0x1 << 6) +#define RT5677_M_STO4_ADC_R2_SFT 6 + +/* Stereo3 ADC Mixer control (0x25) */ +#define RT5677_M_STO3_ADC_L2 (0x1 << 15) +#define RT5677_M_STO3_ADC_L2_SFT 15 +#define RT5677_M_STO3_ADC_L1 (0x1 << 14) +#define RT5677_M_STO3_ADC_L1_SFT 14 +#define RT5677_SEL_STO3_ADC1_MASK (0x3 << 12) +#define RT5677_SEL_STO3_ADC1_SFT 12 +#define RT5677_SEL_STO3_ADC2_MASK (0x3 << 10) +#define RT5677_SEL_STO3_ADC2_SFT 10 +#define RT5677_SEL_STO3_DMIC_MASK (0x3 << 8) +#define RT5677_SEL_STO3_DMIC_SFT 8 +#define RT5677_M_STO3_ADC_R1 (0x1 << 7) +#define RT5677_M_STO3_ADC_R1_SFT 7 +#define RT5677_M_STO3_ADC_R2 (0x1 << 6) +#define RT5677_M_STO3_ADC_R2_SFT 6 + +/* Stereo2 ADC Mixer Control (0x26) */ +#define RT5677_M_STO2_ADC_L2 (0x1 << 15) +#define RT5677_M_STO2_ADC_L2_SFT 15 +#define RT5677_M_STO2_ADC_L1 (0x1 << 14) +#define RT5677_M_STO2_ADC_L1_SFT 14 +#define RT5677_SEL_STO2_ADC1_MASK (0x3 << 12) +#define RT5677_SEL_STO2_ADC1_SFT 12 +#define RT5677_SEL_STO2_ADC2_MASK (0x3 << 10) +#define RT5677_SEL_STO2_ADC2_SFT 10 +#define RT5677_SEL_STO2_DMIC_MASK (0x3 << 8) +#define RT5677_SEL_STO2_DMIC_SFT 8 +#define RT5677_M_STO2_ADC_R1 (0x1 << 7) +#define RT5677_M_STO2_ADC_R1_SFT 7 +#define RT5677_M_STO2_ADC_R2 (0x1 << 6) +#define RT5677_M_STO2_ADC_R2_SFT 6 +#define RT5677_SEL_STO2_LR_MIX_MASK (0x1 << 0) +#define RT5677_SEL_STO2_LR_MIX_SFT 0 +#define RT5677_SEL_STO2_LR_MIX_L (0x0 << 0) +#define RT5677_SEL_STO2_LR_MIX_LR (0x1 << 0) + +/* Stereo1 ADC Mixer control (0x27) */ +#define RT5677_M_STO1_ADC_L2 (0x1 << 15) +#define RT5677_M_STO1_ADC_L2_SFT 15 +#define RT5677_M_STO1_ADC_L1 (0x1 << 14) +#define RT5677_M_STO1_ADC_L1_SFT 14 +#define RT5677_SEL_STO1_ADC1_MASK (0x3 << 12) +#define RT5677_SEL_STO1_ADC1_SFT 12 +#define RT5677_SEL_STO1_ADC2_MASK (0x3 << 10) +#define RT5677_SEL_STO1_ADC2_SFT 10 +#define RT5677_SEL_STO1_DMIC_MASK (0x3 << 8) +#define RT5677_SEL_STO1_DMIC_SFT 8 +#define RT5677_M_STO1_ADC_R1 (0x1 << 7) +#define RT5677_M_STO1_ADC_R1_SFT 7 +#define RT5677_M_STO1_ADC_R2 (0x1 << 6) +#define RT5677_M_STO1_ADC_R2_SFT 6 + +/* Mono ADC Mixer control (0x28) */ +#define RT5677_M_MONO_ADC_L2 (0x1 << 15) +#define RT5677_M_MONO_ADC_L2_SFT 15 +#define RT5677_M_MONO_ADC_L1 (0x1 << 14) +#define RT5677_M_MONO_ADC_L1_SFT 14 +#define RT5677_SEL_MONO_ADC_L1_MASK (0x3 << 12) +#define RT5677_SEL_MONO_ADC_L1_SFT 12 +#define RT5677_SEL_MONO_ADC_L2_MASK (0x3 << 10) +#define RT5677_SEL_MONO_ADC_L2_SFT 10 +#define RT5677_SEL_MONO_DMIC_L_MASK (0x3 << 8) +#define RT5677_SEL_MONO_DMIC_L_SFT 8 +#define RT5677_M_MONO_ADC_R1 (0x1 << 7) +#define RT5677_M_MONO_ADC_R1_SFT 7 +#define RT5677_M_MONO_ADC_R2 (0x1 << 6) +#define RT5677_M_MONO_ADC_R2_SFT 6 +#define RT5677_SEL_MONO_ADC_R1_MASK (0x3 << 4) +#define RT5677_SEL_MONO_ADC_R1_SFT 4 +#define RT5677_SEL_MONO_ADC_R2_MASK (0x3 << 2) +#define RT5677_SEL_MONO_ADC_R2_SFT 2 +#define RT5677_SEL_MONO_DMIC_R_MASK (0x3 << 0) +#define RT5677_SEL_MONO_DMIC_R_SFT 0 + +/* ADC/IF/DSP to DAC1 Mixer control (0x29) */ +#define RT5677_M_ADDA_MIXER1_L (0x1 << 15) +#define RT5677_M_ADDA_MIXER1_L_SFT 15 +#define RT5677_M_DAC1_L (0x1 << 14) +#define RT5677_M_DAC1_L_SFT 14 +#define RT5677_DAC1_L_SEL_MASK (0x7 << 8) +#define RT5677_DAC1_L_SEL_SFT 8 +#define RT5677_M_ADDA_MIXER1_R (0x1 << 7) +#define RT5677_M_ADDA_MIXER1_R_SFT 7 +#define RT5677_M_DAC1_R (0x1 << 6) +#define RT5677_M_DAC1_R_SFT 6 +#define RT5677_ADDA1_SEL_MASK (0x3 << 0) +#define RT5677_ADDA1_SEL_SFT 0 + +/* Stereo1 DAC Mixer L/R Control (0x2a) */ +#define RT5677_M_ST_DAC1_L (0x1 << 15) +#define RT5677_M_ST_DAC1_L_SFT 15 +#define RT5677_M_DAC1_L_STO_L (0x1 << 13) +#define RT5677_M_DAC1_L_STO_L_SFT 13 +#define RT5677_DAC1_L_STO_L_VOL_MASK (0x1 << 12) +#define RT5677_DAC1_L_STO_L_VOL_SFT 12 +#define RT5677_M_DAC2_L_STO_L (0x1 << 11) +#define RT5677_M_DAC2_L_STO_L_SFT 11 +#define RT5677_DAC2_L_STO_L_VOL_MASK (0x1 << 10) +#define RT5677_DAC2_L_STO_L_VOL_SFT 10 +#define RT5677_M_DAC1_R_STO_L (0x1 << 9) +#define RT5677_M_DAC1_R_STO_L_SFT 9 +#define RT5677_DAC1_R_STO_L_VOL_MASK (0x1 << 8) +#define RT5677_DAC1_R_STO_L_VOL_SFT 8 +#define RT5677_M_ST_DAC1_R (0x1 << 7) +#define RT5677_M_ST_DAC1_R_SFT 7 +#define RT5677_M_DAC1_R_STO_R (0x1 << 5) +#define RT5677_M_DAC1_R_STO_R_SFT 5 +#define RT5677_DAC1_R_STO_R_VOL_MASK (0x1 << 4) +#define RT5677_DAC1_R_STO_R_VOL_SFT 4 +#define RT5677_M_DAC2_R_STO_R (0x1 << 3) +#define RT5677_M_DAC2_R_STO_R_SFT 3 +#define RT5677_DAC2_R_STO_R_VOL_MASK (0x1 << 2) +#define RT5677_DAC2_R_STO_R_VOL_SFT 2 +#define RT5677_M_DAC1_L_STO_R (0x1 << 1) +#define RT5677_M_DAC1_L_STO_R_SFT 1 +#define RT5677_DAC1_L_STO_R_VOL_MASK (0x1 << 0) +#define RT5677_DAC1_L_STO_R_VOL_SFT 0 + +/* Mono DAC Mixer L/R Control (0x2b) */ +#define RT5677_M_ST_DAC2_L (0x1 << 15) +#define RT5677_M_ST_DAC2_L_SFT 15 +#define RT5677_M_DAC2_L_MONO_L (0x1 << 13) +#define RT5677_M_DAC2_L_MONO_L_SFT 13 +#define RT5677_DAC2_L_MONO_L_VOL_MASK (0x1 << 12) +#define RT5677_DAC2_L_MONO_L_VOL_SFT 12 +#define RT5677_M_DAC2_R_MONO_L (0x1 << 11) +#define RT5677_M_DAC2_R_MONO_L_SFT 11 +#define RT5677_DAC2_R_MONO_L_VOL_MASK (0x1 << 10) +#define RT5677_DAC2_R_MONO_L_VOL_SFT 10 +#define RT5677_M_DAC1_L_MONO_L (0x1 << 9) +#define RT5677_M_DAC1_L_MONO_L_SFT 9 +#define RT5677_DAC1_L_MONO_L_VOL_MASK (0x1 << 8) +#define RT5677_DAC1_L_MONO_L_VOL_SFT 8 +#define RT5677_M_ST_DAC2_R (0x1 << 7) +#define RT5677_M_ST_DAC2_R_SFT 7 +#define RT5677_M_DAC2_R_MONO_R (0x1 << 5) +#define RT5677_M_DAC2_R_MONO_R_SFT 5 +#define RT5677_DAC2_R_MONO_R_VOL_MASK (0x1 << 4) +#define RT5677_DAC2_R_MONO_R_VOL_SFT 4 +#define RT5677_M_DAC1_R_MONO_R (0x1 << 3) +#define RT5677_M_DAC1_R_MONO_R_SFT 3 +#define RT5677_DAC1_R_MONO_R_VOL_MASK (0x1 << 2) +#define RT5677_DAC1_R_MONO_R_VOL_SFT 2 +#define RT5677_M_DAC2_L_MONO_R (0x1 << 1) +#define RT5677_M_DAC2_L_MONO_R_SFT 1 +#define RT5677_DAC2_L_MONO_R_VOL_MASK (0x1 << 0) +#define RT5677_DAC2_L_MONO_R_VOL_SFT 0 + +/* DD Mixer 1 Control (0x2c) */ +#define RT5677_M_STO_L_DD1_L (0x1 << 15) +#define RT5677_M_STO_L_DD1_L_SFT 15 +#define RT5677_STO_L_DD1_L_VOL_MASK (0x1 << 14) +#define RT5677_STO_L_DD1_L_VOL_SFT 14 +#define RT5677_M_MONO_L_DD1_L (0x1 << 13) +#define RT5677_M_MONO_L_DD1_L_SFT 13 +#define RT5677_MONO_L_DD1_L_VOL_MASK (0x1 << 12) +#define RT5677_MONO_L_DD1_L_VOL_SFT 12 +#define RT5677_M_DAC3_L_DD1_L (0x1 << 11) +#define RT5677_M_DAC3_L_DD1_L_SFT 11 +#define RT5677_DAC3_L_DD1_L_VOL_MASK (0x1 << 10) +#define RT5677_DAC3_L_DD1_L_VOL_SFT 10 +#define RT5677_M_DAC3_R_DD1_L (0x1 << 9) +#define RT5677_M_DAC3_R_DD1_L_SFT 9 +#define RT5677_DAC3_R_DD1_L_VOL_MASK (0x1 << 8) +#define RT5677_DAC3_R_DD1_L_VOL_SFT 8 +#define RT5677_M_STO_R_DD1_R (0x1 << 7) +#define RT5677_M_STO_R_DD1_R_SFT 7 +#define RT5677_STO_R_DD1_R_VOL_MASK (0x1 << 6) +#define RT5677_STO_R_DD1_R_VOL_SFT 6 +#define RT5677_M_MONO_R_DD1_R (0x1 << 5) +#define RT5677_M_MONO_R_DD1_R_SFT 5 +#define RT5677_MONO_R_DD1_R_VOL_MASK (0x1 << 4) +#define RT5677_MONO_R_DD1_R_VOL_SFT 4 +#define RT5677_M_DAC3_R_DD1_R (0x1 << 3) +#define RT5677_M_DAC3_R_DD1_R_SFT 3 +#define RT5677_DAC3_R_DD1_R_VOL_MASK (0x1 << 2) +#define RT5677_DAC3_R_DD1_R_VOL_SFT 2 +#define RT5677_M_DAC3_L_DD1_R (0x1 << 1) +#define RT5677_M_DAC3_L_DD1_R_SFT 1 +#define RT5677_DAC3_L_DD1_R_VOL_MASK (0x1 << 0) +#define RT5677_DAC3_L_DD1_R_VOL_SFT 0 + +/* DD Mixer 2 Control (0x2d) */ +#define RT5677_M_STO_L_DD2_L (0x1 << 15) +#define RT5677_M_STO_L_DD2_L_SFT 15 +#define RT5677_STO_L_DD2_L_VOL_MASK (0x1 << 14) +#define RT5677_STO_L_DD2_L_VOL_SFT 14 +#define RT5677_M_MONO_L_DD2_L (0x1 << 13) +#define RT5677_M_MONO_L_DD2_L_SFT 13 +#define RT5677_MONO_L_DD2_L_VOL_MASK (0x1 << 12) +#define RT5677_MONO_L_DD2_L_VOL_SFT 12 +#define RT5677_M_DAC4_L_DD2_L (0x1 << 11) +#define RT5677_M_DAC4_L_DD2_L_SFT 11 +#define RT5677_DAC4_L_DD2_L_VOL_MASK (0x1 << 10) +#define RT5677_DAC4_L_DD2_L_VOL_SFT 10 +#define RT5677_M_DAC4_R_DD2_L (0x1 << 9) +#define RT5677_M_DAC4_R_DD2_L_SFT 9 +#define RT5677_DAC4_R_DD2_L_VOL_MASK (0x1 << 8) +#define RT5677_DAC4_R_DD2_L_VOL_SFT 8 +#define RT5677_M_STO_R_DD2_R (0x1 << 7) +#define RT5677_M_STO_R_DD2_R_SFT 7 +#define RT5677_STO_R_DD2_R_VOL_MASK (0x1 << 6) +#define RT5677_STO_R_DD2_R_VOL_SFT 6 +#define RT5677_M_MONO_R_DD2_R (0x1 << 5) +#define RT5677_M_MONO_R_DD2_R_SFT 5 +#define RT5677_MONO_R_DD2_R_VOL_MASK (0x1 << 4) +#define RT5677_MONO_R_DD2_R_VOL_SFT 4 +#define RT5677_M_DAC4_R_DD2_R (0x1 << 3) +#define RT5677_M_DAC4_R_DD2_R_SFT 3 +#define RT5677_DAC4_R_DD2_R_VOL_MASK (0x1 << 2) +#define RT5677_DAC4_R_DD2_R_VOL_SFT 2 +#define RT5677_M_DAC4_L_DD2_R (0x1 << 1) +#define RT5677_M_DAC4_L_DD2_R_SFT 1 +#define RT5677_DAC4_L_DD2_R_VOL_MASK (0x1 << 0) +#define RT5677_DAC4_L_DD2_R_VOL_SFT 0 + +/* IF3 data control (0x2f) */ +#define RT5677_IF3_DAC_SEL_MASK (0x3 << 6) +#define RT5677_IF3_DAC_SEL_SFT 6 +#define RT5677_IF3_ADC_SEL_MASK (0x3 << 4) +#define RT5677_IF3_ADC_SEL_SFT 4 +#define RT5677_IF3_ADC_IN_MASK (0xf << 0) +#define RT5677_IF3_ADC_IN_SFT 0 + +/* IF4 data control (0x30) */ +#define RT5677_IF4_ADC_IN_MASK (0xf << 4) +#define RT5677_IF4_ADC_IN_SFT 4 +#define RT5677_IF4_DAC_SEL_MASK (0x3 << 2) +#define RT5677_IF4_DAC_SEL_SFT 2 +#define RT5677_IF4_ADC_SEL_MASK (0x3 << 0) +#define RT5677_IF4_ADC_SEL_SFT 0 + +/* PDM Output Control (0x31) */ +#define RT5677_M_PDM1_L (0x1 << 15) +#define RT5677_M_PDM1_L_SFT 15 +#define RT5677_SEL_PDM1_L_MASK (0x3 << 12) +#define RT5677_SEL_PDM1_L_SFT 12 +#define RT5677_M_PDM1_R (0x1 << 11) +#define RT5677_M_PDM1_R_SFT 11 +#define RT5677_SEL_PDM1_R_MASK (0x3 << 8) +#define RT5677_SEL_PDM1_R_SFT 8 +#define RT5677_M_PDM2_L (0x1 << 7) +#define RT5677_M_PDM2_L_SFT 7 +#define RT5677_SEL_PDM2_L_MASK (0x3 << 4) +#define RT5677_SEL_PDM2_L_SFT 4 +#define RT5677_M_PDM2_R (0x1 << 3) +#define RT5677_M_PDM2_R_SFT 3 +#define RT5677_SEL_PDM2_R_MASK (0x3 << 0) +#define RT5677_SEL_PDM2_R_SFT 0 + +/* PDM I2C / Data Control 1 (0x32) */ +#define RT5677_PDM2_PW_DOWN (0x1 << 7) +#define RT5677_PDM1_PW_DOWN (0x1 << 6) +#define RT5677_PDM2_BUSY (0x1 << 5) +#define RT5677_PDM1_BUSY (0x1 << 4) +#define RT5677_PDM_PATTERN (0x1 << 3) +#define RT5677_PDM_GAIN (0x1 << 2) +#define RT5677_PDM_DIV_MASK (0x3 << 0) + +/* PDM I2C / Data Control 2 (0x33) */ +#define RT5677_PDM1_I2C_ID (0xf << 12) +#define RT5677_PDM1_EXE (0x1 << 11) +#define RT5677_PDM1_I2C_CMD (0x1 << 10) +#define RT5677_PDM1_I2C_EXE (0x1 << 9) +#define RT5677_PDM1_I2C_BUSY (0x1 << 8) +#define RT5677_PDM2_I2C_ID (0xf << 4) +#define RT5677_PDM2_EXE (0x1 << 3) +#define RT5677_PDM2_I2C_CMD (0x1 << 2) +#define RT5677_PDM2_I2C_EXE (0x1 << 1) +#define RT5677_PDM2_I2C_BUSY (0x1 << 0) + +/* MX3C TDM1 control 1 (0x3c) */ +#define RT5677_IF1_ADC4_MASK (0x3 << 10) +#define RT5677_IF1_ADC4_SFT 10 +#define RT5677_IF1_ADC3_MASK (0x3 << 8) +#define RT5677_IF1_ADC3_SFT 8 +#define RT5677_IF1_ADC2_MASK (0x3 << 6) +#define RT5677_IF1_ADC2_SFT 6 +#define RT5677_IF1_ADC1_MASK (0x3 << 4) +#define RT5677_IF1_ADC1_SFT 4 + +/* MX41 TDM2 control 1 (0x41) */ +#define RT5677_IF2_ADC4_MASK (0x3 << 10) +#define RT5677_IF2_ADC4_SFT 10 +#define RT5677_IF2_ADC3_MASK (0x3 << 8) +#define RT5677_IF2_ADC3_SFT 8 +#define RT5677_IF2_ADC2_MASK (0x3 << 6) +#define RT5677_IF2_ADC2_SFT 6 +#define RT5677_IF2_ADC1_MASK (0x3 << 4) +#define RT5677_IF2_ADC1_SFT 4 + +/* Digital Microphone Control 1 (0x50) */ +#define RT5677_DMIC_1_EN_MASK (0x1 << 15) +#define RT5677_DMIC_1_EN_SFT 15 +#define RT5677_DMIC_1_DIS (0x0 << 15) +#define RT5677_DMIC_1_EN (0x1 << 15) +#define RT5677_DMIC_2_EN_MASK (0x1 << 14) +#define RT5677_DMIC_2_EN_SFT 14 +#define RT5677_DMIC_2_DIS (0x0 << 14) +#define RT5677_DMIC_2_EN (0x1 << 14) +#define RT5677_DMIC_L_STO1_LH_MASK (0x1 << 13) +#define RT5677_DMIC_L_STO1_LH_SFT 13 +#define RT5677_DMIC_L_STO1_LH_FALLING (0x0 << 13) +#define RT5677_DMIC_L_STO1_LH_RISING (0x1 << 13) +#define RT5677_DMIC_R_STO1_LH_MASK (0x1 << 12) +#define RT5677_DMIC_R_STO1_LH_SFT 12 +#define RT5677_DMIC_R_STO1_LH_FALLING (0x0 << 12) +#define RT5677_DMIC_R_STO1_LH_RISING (0x1 << 12) +#define RT5677_DMIC_L_STO3_LH_MASK (0x1 << 11) +#define RT5677_DMIC_L_STO3_LH_SFT 11 +#define RT5677_DMIC_L_STO3_LH_FALLING (0x0 << 11) +#define RT5677_DMIC_L_STO3_LH_RISING (0x1 << 11) +#define RT5677_DMIC_R_STO3_LH_MASK (0x1 << 10) +#define RT5677_DMIC_R_STO3_LH_SFT 10 +#define RT5677_DMIC_R_STO3_LH_FALLING (0x0 << 10) +#define RT5677_DMIC_R_STO3_LH_RISING (0x1 << 10) +#define RT5677_DMIC_L_STO2_LH_MASK (0x1 << 9) +#define RT5677_DMIC_L_STO2_LH_SFT 9 +#define RT5677_DMIC_L_STO2_LH_FALLING (0x0 << 9) +#define RT5677_DMIC_L_STO2_LH_RISING (0x1 << 9) +#define RT5677_DMIC_R_STO2_LH_MASK (0x1 << 8) +#define RT5677_DMIC_R_STO2_LH_SFT 8 +#define RT5677_DMIC_R_STO2_LH_FALLING (0x0 << 8) +#define RT5677_DMIC_R_STO2_LH_RISING (0x1 << 8) +#define RT5677_DMIC_CLK_MASK (0x7 << 5) +#define RT5677_DMIC_CLK_SFT 5 +#define RT5677_DMIC_3_EN_MASK (0x1 << 4) +#define RT5677_DMIC_3_EN_SFT 4 +#define RT5677_DMIC_3_DIS (0x0 << 4) +#define RT5677_DMIC_3_EN (0x1 << 4) +#define RT5677_DMIC_R_MONO_LH_MASK (0x1 << 2) +#define RT5677_DMIC_R_MONO_LH_SFT 2 +#define RT5677_DMIC_R_MONO_LH_FALLING (0x0 << 2) +#define RT5677_DMIC_R_MONO_LH_RISING (0x1 << 2) +#define RT5677_DMIC_L_STO4_LH_MASK (0x1 << 1) +#define RT5677_DMIC_L_STO4_LH_SFT 1 +#define RT5677_DMIC_L_STO4_LH_FALLING (0x0 << 1) +#define RT5677_DMIC_L_STO4_LH_RISING (0x1 << 1) +#define RT5677_DMIC_R_STO4_LH_MASK (0x1 << 0) +#define RT5677_DMIC_R_STO4_LH_SFT 0 +#define RT5677_DMIC_R_STO4_LH_FALLING (0x0 << 0) +#define RT5677_DMIC_R_STO4_LH_RISING (0x1 << 0) + +/* Digital Microphone Control 2 (0x51) */ +#define RT5677_DMIC_4_EN_MASK (0x1 << 15) +#define RT5677_DMIC_4_EN_SFT 15 +#define RT5677_DMIC_4_DIS (0x0 << 15) +#define RT5677_DMIC_4_EN (0x1 << 15) +#define RT5677_DMIC_4L_LH_MASK (0x1 << 7) +#define RT5677_DMIC_4L_LH_SFT 7 +#define RT5677_DMIC_4L_LH_FALLING (0x0 << 7) +#define RT5677_DMIC_4L_LH_RISING (0x1 << 7) +#define RT5677_DMIC_4R_LH_MASK (0x1 << 6) +#define RT5677_DMIC_4R_LH_SFT 6 +#define RT5677_DMIC_4R_LH_FALLING (0x0 << 6) +#define RT5677_DMIC_4R_LH_RISING (0x1 << 6) +#define RT5677_DMIC_3L_LH_MASK (0x1 << 5) +#define RT5677_DMIC_3L_LH_SFT 5 +#define RT5677_DMIC_3L_LH_FALLING (0x0 << 5) +#define RT5677_DMIC_3L_LH_RISING (0x1 << 5) +#define RT5677_DMIC_3R_LH_MASK (0x1 << 4) +#define RT5677_DMIC_3R_LH_SFT 4 +#define RT5677_DMIC_3R_LH_FALLING (0x0 << 4) +#define RT5677_DMIC_3R_LH_RISING (0x1 << 4) +#define RT5677_DMIC_2L_LH_MASK (0x1 << 3) +#define RT5677_DMIC_2L_LH_SFT 3 +#define RT5677_DMIC_2L_LH_FALLING (0x0 << 3) +#define RT5677_DMIC_2L_LH_RISING (0x1 << 3) +#define RT5677_DMIC_2R_LH_MASK (0x1 << 2) +#define RT5677_DMIC_2R_LH_SFT 2 +#define RT5677_DMIC_2R_LH_FALLING (0x0 << 2) +#define RT5677_DMIC_2R_LH_RISING (0x1 << 2) +#define RT5677_DMIC_1L_LH_MASK (0x1 << 1) +#define RT5677_DMIC_1L_LH_SFT 1 +#define RT5677_DMIC_1L_LH_FALLING (0x0 << 1) +#define RT5677_DMIC_1L_LH_RISING (0x1 << 1) +#define RT5677_DMIC_1R_LH_MASK (0x1 << 0) +#define RT5677_DMIC_1R_LH_SFT 0 +#define RT5677_DMIC_1R_LH_FALLING (0x0 << 0) +#define RT5677_DMIC_1R_LH_RISING (0x1 << 0) + +/* Power Management for Digital 1 (0x61) */ +#define RT5677_PWR_I2S1 (0x1 << 15) +#define RT5677_PWR_I2S1_BIT 15 +#define RT5677_PWR_I2S2 (0x1 << 14) +#define RT5677_PWR_I2S2_BIT 14 +#define RT5677_PWR_I2S3 (0x1 << 13) +#define RT5677_PWR_I2S3_BIT 13 +#define RT5677_PWR_DAC1 (0x1 << 12) +#define RT5677_PWR_DAC1_BIT 12 +#define RT5677_PWR_DAC2 (0x1 << 11) +#define RT5677_PWR_DAC2_BIT 11 +#define RT5677_PWR_I2S4 (0x1 << 10) +#define RT5677_PWR_I2S4_BIT 10 +#define RT5677_PWR_SLB (0x1 << 9) +#define RT5677_PWR_SLB_BIT 9 +#define RT5677_PWR_DAC3 (0x1 << 7) +#define RT5677_PWR_DAC3_BIT 7 +#define RT5677_PWR_ADCFED2 (0x1 << 4) +#define RT5677_PWR_ADCFED2_BIT 4 +#define RT5677_PWR_ADCFED1 (0x1 << 3) +#define RT5677_PWR_ADCFED1_BIT 3 +#define RT5677_PWR_ADC_L (0x1 << 2) +#define RT5677_PWR_ADC_L_BIT 2 +#define RT5677_PWR_ADC_R (0x1 << 1) +#define RT5677_PWR_ADC_R_BIT 1 +#define RT5677_PWR_I2C_MASTER (0x1 << 0) +#define RT5677_PWR_I2C_MASTER_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5677_PWR_ADC_S1F (0x1 << 15) +#define RT5677_PWR_ADC_S1F_BIT 15 +#define RT5677_PWR_ADC_MF_L (0x1 << 14) +#define RT5677_PWR_ADC_MF_L_BIT 14 +#define RT5677_PWR_ADC_MF_R (0x1 << 13) +#define RT5677_PWR_ADC_MF_R_BIT 13 +#define RT5677_PWR_DAC_S1F (0x1 << 12) +#define RT5677_PWR_DAC_S1F_BIT 12 +#define RT5677_PWR_DAC_M2F_L (0x1 << 11) +#define RT5677_PWR_DAC_M2F_L_BIT 11 +#define RT5677_PWR_DAC_M2F_R (0x1 << 10) +#define RT5677_PWR_DAC_M2F_R_BIT 10 +#define RT5677_PWR_DAC_M3F_L (0x1 << 9) +#define RT5677_PWR_DAC_M3F_L_BIT 9 +#define RT5677_PWR_DAC_M3F_R (0x1 << 8) +#define RT5677_PWR_DAC_M3F_R_BIT 8 +#define RT5677_PWR_DAC_M4F_L (0x1 << 7) +#define RT5677_PWR_DAC_M4F_L_BIT 7 +#define RT5677_PWR_DAC_M4F_R (0x1 << 6) +#define RT5677_PWR_DAC_M4F_R_BIT 6 +#define RT5677_PWR_ADC_S2F (0x1 << 5) +#define RT5677_PWR_ADC_S2F_BIT 5 +#define RT5677_PWR_ADC_S3F (0x1 << 4) +#define RT5677_PWR_ADC_S3F_BIT 4 +#define RT5677_PWR_ADC_S4F (0x1 << 3) +#define RT5677_PWR_ADC_S4F_BIT 3 +#define RT5677_PWR_PDM1 (0x1 << 2) +#define RT5677_PWR_PDM1_BIT 2 +#define RT5677_PWR_PDM2 (0x1 << 1) +#define RT5677_PWR_PDM2_BIT 1 + +/* Power Management for Analog 1 (0x63) */ +#define RT5677_PWR_VREF1 (0x1 << 15) +#define RT5677_PWR_VREF1_BIT 15 +#define RT5677_PWR_FV1 (0x1 << 14) +#define RT5677_PWR_FV1_BIT 14 +#define RT5677_PWR_MB (0x1 << 13) +#define RT5677_PWR_MB_BIT 13 +#define RT5677_PWR_LO1 (0x1 << 12) +#define RT5677_PWR_LO1_BIT 12 +#define RT5677_PWR_BG (0x1 << 11) +#define RT5677_PWR_BG_BIT 11 +#define RT5677_PWR_LO2 (0x1 << 10) +#define RT5677_PWR_LO2_BIT 10 +#define RT5677_PWR_LO3 (0x1 << 9) +#define RT5677_PWR_LO3_BIT 9 +#define RT5677_PWR_VREF2 (0x1 << 8) +#define RT5677_PWR_VREF2_BIT 8 +#define RT5677_PWR_FV2 (0x1 << 7) +#define RT5677_PWR_FV2_BIT 7 +#define RT5677_LDO2_SEL_MASK (0x7 << 4) +#define RT5677_LDO2_SEL_SFT 4 +#define RT5677_LDO1_SEL_MASK (0x7 << 0) +#define RT5677_LDO1_SEL_SFT 0 + +/* Power Management for Analog 2 (0x64) */ +#define RT5677_PWR_BST1 (0x1 << 15) +#define RT5677_PWR_BST1_BIT 15 +#define RT5677_PWR_BST2 (0x1 << 14) +#define RT5677_PWR_BST2_BIT 14 +#define RT5677_PWR_CLK_MB1 (0x1 << 13) +#define RT5677_PWR_CLK_MB1_BIT 13 +#define RT5677_PWR_SLIM (0x1 << 12) +#define RT5677_PWR_SLIM_BIT 12 +#define RT5677_PWR_MB1 (0x1 << 11) +#define RT5677_PWR_MB1_BIT 11 +#define RT5677_PWR_PP_MB1 (0x1 << 10) +#define RT5677_PWR_PP_MB1_BIT 10 +#define RT5677_PWR_PLL1 (0x1 << 9) +#define RT5677_PWR_PLL1_BIT 9 +#define RT5677_PWR_PLL2 (0x1 << 8) +#define RT5677_PWR_PLL2_BIT 8 +#define RT5677_PWR_CORE (0x1 << 7) +#define RT5677_PWR_CORE_BIT 7 +#define RT5677_PWR_CLK_MB (0x1 << 6) +#define RT5677_PWR_CLK_MB_BIT 6 +#define RT5677_PWR_BST1_P (0x1 << 5) +#define RT5677_PWR_BST1_P_BIT 5 +#define RT5677_PWR_BST2_P (0x1 << 4) +#define RT5677_PWR_BST2_P_BIT 4 +#define RT5677_PWR_IPTV (0x1 << 3) +#define RT5677_PWR_IPTV_BIT 3 +#define RT5677_PWR_25M_CLK (0x1 << 1) +#define RT5677_PWR_25M_CLK_BIT 1 +#define RT5677_PWR_LDO1 (0x1 << 0) +#define RT5677_PWR_LDO1_BIT 0 + +/* Power Management for DSP (0x65) */ +#define RT5677_PWR_SR7 (0x1 << 10) +#define RT5677_PWR_SR7_BIT 10 +#define RT5677_PWR_SR6 (0x1 << 9) +#define RT5677_PWR_SR6_BIT 9 +#define RT5677_PWR_SR5 (0x1 << 8) +#define RT5677_PWR_SR5_BIT 8 +#define RT5677_PWR_SR4 (0x1 << 7) +#define RT5677_PWR_SR4_BIT 7 +#define RT5677_PWR_SR3 (0x1 << 6) +#define RT5677_PWR_SR3_BIT 6 +#define RT5677_PWR_SR2 (0x1 << 5) +#define RT5677_PWR_SR2_BIT 5 +#define RT5677_PWR_SR1 (0x1 << 4) +#define RT5677_PWR_SR1_BIT 4 +#define RT5677_PWR_SR0 (0x1 << 3) +#define RT5677_PWR_SR0_BIT 3 +#define RT5677_PWR_MLT (0x1 << 2) +#define RT5677_PWR_MLT_BIT 2 +#define RT5677_PWR_DSP (0x1 << 1) +#define RT5677_PWR_DSP_BIT 1 +#define RT5677_PWR_DSP_CPU (0x1 << 0) +#define RT5677_PWR_DSP_CPU_BIT 0 + +/* Power Status for DSP (0x66) */ +#define RT5677_PWR_SR7_RDY (0x1 << 9) +#define RT5677_PWR_SR7_RDY_BIT 9 +#define RT5677_PWR_SR6_RDY (0x1 << 8) +#define RT5677_PWR_SR6_RDY_BIT 8 +#define RT5677_PWR_SR5_RDY (0x1 << 7) +#define RT5677_PWR_SR5_RDY_BIT 7 +#define RT5677_PWR_SR4_RDY (0x1 << 6) +#define RT5677_PWR_SR4_RDY_BIT 6 +#define RT5677_PWR_SR3_RDY (0x1 << 5) +#define RT5677_PWR_SR3_RDY_BIT 5 +#define RT5677_PWR_SR2_RDY (0x1 << 4) +#define RT5677_PWR_SR2_RDY_BIT 4 +#define RT5677_PWR_SR1_RDY (0x1 << 3) +#define RT5677_PWR_SR1_RDY_BIT 3 +#define RT5677_PWR_SR0_RDY (0x1 << 2) +#define RT5677_PWR_SR0_RDY_BIT 2 +#define RT5677_PWR_MLT_RDY (0x1 << 1) +#define RT5677_PWR_MLT_RDY_BIT 1 +#define RT5677_PWR_DSP_RDY (0x1 << 0) +#define RT5677_PWR_DSP_RDY_BIT 0 + +/* Power Management for DSP (0x67) */ +#define RT5677_PWR_SLIM_ISO (0x1 << 11) +#define RT5677_PWR_SLIM_ISO_BIT 11 +#define RT5677_PWR_CORE_ISO (0x1 << 10) +#define RT5677_PWR_CORE_ISO_BIT 10 +#define RT5677_PWR_DSP_ISO (0x1 << 9) +#define RT5677_PWR_DSP_ISO_BIT 9 +#define RT5677_PWR_SR7_ISO (0x1 << 8) +#define RT5677_PWR_SR7_ISO_BIT 8 +#define RT5677_PWR_SR6_ISO (0x1 << 7) +#define RT5677_PWR_SR6_ISO_BIT 7 +#define RT5677_PWR_SR5_ISO (0x1 << 6) +#define RT5677_PWR_SR5_ISO_BIT 6 +#define RT5677_PWR_SR4_ISO (0x1 << 5) +#define RT5677_PWR_SR4_ISO_BIT 5 +#define RT5677_PWR_SR3_ISO (0x1 << 4) +#define RT5677_PWR_SR3_ISO_BIT 4 +#define RT5677_PWR_SR2_ISO (0x1 << 3) +#define RT5677_PWR_SR2_ISO_BIT 3 +#define RT5677_PWR_SR1_ISO (0x1 << 2) +#define RT5677_PWR_SR1_ISO_BIT 2 +#define RT5677_PWR_SR0_ISO (0x1 << 1) +#define RT5677_PWR_SR0_ISO_BIT 1 +#define RT5677_PWR_MLT_ISO (0x1 << 0) +#define RT5677_PWR_MLT_ISO_BIT 0 + +/* I2S1/2/3/4 Audio Serial Data Port Control (0x6f 0x70 0x71 0x72) */ +#define RT5677_I2S_MS_MASK (0x1 << 15) +#define RT5677_I2S_MS_SFT 15 +#define RT5677_I2S_MS_M (0x0 << 15) +#define RT5677_I2S_MS_S (0x1 << 15) +#define RT5677_I2S_O_CP_MASK (0x3 << 10) +#define RT5677_I2S_O_CP_SFT 10 +#define RT5677_I2S_O_CP_OFF (0x0 << 10) +#define RT5677_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5677_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5677_I2S_I_CP_MASK (0x3 << 8) +#define RT5677_I2S_I_CP_SFT 8 +#define RT5677_I2S_I_CP_OFF (0x0 << 8) +#define RT5677_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5677_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5677_I2S_BP_MASK (0x1 << 7) +#define RT5677_I2S_BP_SFT 7 +#define RT5677_I2S_BP_NOR (0x0 << 7) +#define RT5677_I2S_BP_INV (0x1 << 7) +#define RT5677_I2S_DL_MASK (0x3 << 2) +#define RT5677_I2S_DL_SFT 2 +#define RT5677_I2S_DL_16 (0x0 << 2) +#define RT5677_I2S_DL_20 (0x1 << 2) +#define RT5677_I2S_DL_24 (0x2 << 2) +#define RT5677_I2S_DL_8 (0x3 << 2) +#define RT5677_I2S_DF_MASK (0x3 << 0) +#define RT5677_I2S_DF_SFT 0 +#define RT5677_I2S_DF_I2S (0x0 << 0) +#define RT5677_I2S_DF_LEFT (0x1 << 0) +#define RT5677_I2S_DF_PCM_A (0x2 << 0) +#define RT5677_I2S_DF_PCM_B (0x3 << 0) + +/* Clock Tree Control 1 (0x73) */ +#define RT5677_I2S_PD1_MASK (0x7 << 12) +#define RT5677_I2S_PD1_SFT 12 +#define RT5677_I2S_PD1_1 (0x0 << 12) +#define RT5677_I2S_PD1_2 (0x1 << 12) +#define RT5677_I2S_PD1_3 (0x2 << 12) +#define RT5677_I2S_PD1_4 (0x3 << 12) +#define RT5677_I2S_PD1_6 (0x4 << 12) +#define RT5677_I2S_PD1_8 (0x5 << 12) +#define RT5677_I2S_PD1_12 (0x6 << 12) +#define RT5677_I2S_PD1_16 (0x7 << 12) +#define RT5677_I2S_BCLK_MS2_MASK (0x1 << 11) +#define RT5677_I2S_BCLK_MS2_SFT 11 +#define RT5677_I2S_BCLK_MS2_32 (0x0 << 11) +#define RT5677_I2S_BCLK_MS2_64 (0x1 << 11) +#define RT5677_I2S_PD2_MASK (0x7 << 8) +#define RT5677_I2S_PD2_SFT 8 +#define RT5677_I2S_PD2_1 (0x0 << 8) +#define RT5677_I2S_PD2_2 (0x1 << 8) +#define RT5677_I2S_PD2_3 (0x2 << 8) +#define RT5677_I2S_PD2_4 (0x3 << 8) +#define RT5677_I2S_PD2_6 (0x4 << 8) +#define RT5677_I2S_PD2_8 (0x5 << 8) +#define RT5677_I2S_PD2_12 (0x6 << 8) +#define RT5677_I2S_PD2_16 (0x7 << 8) +#define RT5677_I2S_BCLK_MS3_MASK (0x1 << 7) +#define RT5677_I2S_BCLK_MS3_SFT 7 +#define RT5677_I2S_BCLK_MS3_32 (0x0 << 7) +#define RT5677_I2S_BCLK_MS3_64 (0x1 << 7) +#define RT5677_I2S_PD3_MASK (0x7 << 4) +#define RT5677_I2S_PD3_SFT 4 +#define RT5677_I2S_PD3_1 (0x0 << 4) +#define RT5677_I2S_PD3_2 (0x1 << 4) +#define RT5677_I2S_PD3_3 (0x2 << 4) +#define RT5677_I2S_PD3_4 (0x3 << 4) +#define RT5677_I2S_PD3_6 (0x4 << 4) +#define RT5677_I2S_PD3_8 (0x5 << 4) +#define RT5677_I2S_PD3_12 (0x6 << 4) +#define RT5677_I2S_PD3_16 (0x7 << 4) +#define RT5677_I2S_BCLK_MS4_MASK (0x1 << 3) +#define RT5677_I2S_BCLK_MS4_SFT 3 +#define RT5677_I2S_BCLK_MS4_32 (0x0 << 3) +#define RT5677_I2S_BCLK_MS4_64 (0x1 << 3) +#define RT5677_I2S_PD4_MASK (0x7 << 0) +#define RT5677_I2S_PD4_SFT 0 +#define RT5677_I2S_PD4_1 (0x0 << 0) +#define RT5677_I2S_PD4_2 (0x1 << 0) +#define RT5677_I2S_PD4_3 (0x2 << 0) +#define RT5677_I2S_PD4_4 (0x3 << 0) +#define RT5677_I2S_PD4_6 (0x4 << 0) +#define RT5677_I2S_PD4_8 (0x5 << 0) +#define RT5677_I2S_PD4_12 (0x6 << 0) +#define RT5677_I2S_PD4_16 (0x7 << 0) + +/* Clock Tree Control 2 (0x74) */ +#define RT5677_I2S_PD5_MASK (0x7 << 12) +#define RT5677_I2S_PD5_SFT 12 +#define RT5677_I2S_PD5_1 (0x0 << 12) +#define RT5677_I2S_PD5_2 (0x1 << 12) +#define RT5677_I2S_PD5_3 (0x2 << 12) +#define RT5677_I2S_PD5_4 (0x3 << 12) +#define RT5677_I2S_PD5_6 (0x4 << 12) +#define RT5677_I2S_PD5_8 (0x5 << 12) +#define RT5677_I2S_PD5_12 (0x6 << 12) +#define RT5677_I2S_PD5_16 (0x7 << 12) +#define RT5677_I2S_PD6_MASK (0x7 << 8) +#define RT5677_I2S_PD6_SFT 8 +#define RT5677_I2S_PD6_1 (0x0 << 8) +#define RT5677_I2S_PD6_2 (0x1 << 8) +#define RT5677_I2S_PD6_3 (0x2 << 8) +#define RT5677_I2S_PD6_4 (0x3 << 8) +#define RT5677_I2S_PD6_6 (0x4 << 8) +#define RT5677_I2S_PD6_8 (0x5 << 8) +#define RT5677_I2S_PD6_12 (0x6 << 8) +#define RT5677_I2S_PD6_16 (0x7 << 8) +#define RT5677_I2S_PD7_MASK (0x7 << 4) +#define RT5677_I2S_PD7_SFT 4 +#define RT5677_I2S_PD7_1 (0x0 << 4) +#define RT5677_I2S_PD7_2 (0x1 << 4) +#define RT5677_I2S_PD7_3 (0x2 << 4) +#define RT5677_I2S_PD7_4 (0x3 << 4) +#define RT5677_I2S_PD7_6 (0x4 << 4) +#define RT5677_I2S_PD7_8 (0x5 << 4) +#define RT5677_I2S_PD7_12 (0x6 << 4) +#define RT5677_I2S_PD7_16 (0x7 << 4) +#define RT5677_I2S_PD8_MASK (0x7 << 0) +#define RT5677_I2S_PD8_SFT 0 +#define RT5677_I2S_PD8_1 (0x0 << 0) +#define RT5677_I2S_PD8_2 (0x1 << 0) +#define RT5677_I2S_PD8_3 (0x2 << 0) +#define RT5677_I2S_PD8_4 (0x3 << 0) +#define RT5677_I2S_PD8_6 (0x4 << 0) +#define RT5677_I2S_PD8_8 (0x5 << 0) +#define RT5677_I2S_PD8_12 (0x6 << 0) +#define RT5677_I2S_PD8_16 (0x7 << 0) + +/* Clock Tree Control 3 (0x75) */ +#define RT5677_DSP_ASRC_O_MASK (0x3 << 6) +#define RT5677_DSP_ASRC_O_SFT 6 +#define RT5677_DSP_ASRC_O_1_0 (0x0 << 6) +#define RT5677_DSP_ASRC_O_1_5 (0x1 << 6) +#define RT5677_DSP_ASRC_O_2_0 (0x2 << 6) +#define RT5677_DSP_ASRC_O_3_0 (0x3 << 6) +#define RT5677_DSP_ASRC_I_MASK (0x3 << 4) +#define RT5677_DSP_ASRC_I_SFT 4 +#define RT5677_DSP_ASRC_I_1_0 (0x0 << 4) +#define RT5677_DSP_ASRC_I_1_5 (0x1 << 4) +#define RT5677_DSP_ASRC_I_2_0 (0x2 << 4) +#define RT5677_DSP_ASRC_I_3_0 (0x3 << 4) +#define RT5677_DSP_BUS_PD_MASK (0x7 << 0) +#define RT5677_DSP_BUS_PD_SFT 0 +#define RT5677_DSP_BUS_PD_1 (0x0 << 0) +#define RT5677_DSP_BUS_PD_2 (0x1 << 0) +#define RT5677_DSP_BUS_PD_3 (0x2 << 0) +#define RT5677_DSP_BUS_PD_4 (0x3 << 0) +#define RT5677_DSP_BUS_PD_6 (0x4 << 0) +#define RT5677_DSP_BUS_PD_8 (0x5 << 0) +#define RT5677_DSP_BUS_PD_12 (0x6 << 0) +#define RT5677_DSP_BUS_PD_16 (0x7 << 0) + +#define RT5677_PLL_INP_MAX 40000000 +#define RT5677_PLL_INP_MIN 2048000 +/* PLL M/N/K Code Control 1 (0x7a 0x7c) */ +#define RT5677_PLL_N_MAX 0x1ff +#define RT5677_PLL_N_MASK (RT5677_PLL_N_MAX << 7) +#define RT5677_PLL_N_SFT 7 +#define RT5677_PLL_K_BP (0x1 << 5) +#define RT5677_PLL_K_BP_SFT 5 +#define RT5677_PLL_K_MAX 0x1f +#define RT5677_PLL_K_MASK (RT5677_PLL_K_MAX) +#define RT5677_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x7b 0x7d) */ +#define RT5677_PLL_M_MAX 0xf +#define RT5677_PLL_M_MASK (RT5677_PLL_M_MAX << 12) +#define RT5677_PLL_M_SFT 12 +#define RT5677_PLL_M_BP (0x1 << 11) +#define RT5677_PLL_M_BP_SFT 11 + +/* Global Clock Control 1 (0x80) */ +#define RT5677_SCLK_SRC_MASK (0x3 << 14) +#define RT5677_SCLK_SRC_SFT 14 +#define RT5677_SCLK_SRC_MCLK (0x0 << 14) +#define RT5677_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5677_SCLK_SRC_RCCLK (0x2 << 14) /* 25MHz */ +#define RT5677_SCLK_SRC_SLIM (0x3 << 14) +#define RT5677_PLL1_SRC_MASK (0x7 << 11) +#define RT5677_PLL1_SRC_SFT 11 +#define RT5677_PLL1_SRC_MCLK (0x0 << 11) +#define RT5677_PLL1_SRC_BCLK1 (0x1 << 11) +#define RT5677_PLL1_SRC_BCLK2 (0x2 << 11) +#define RT5677_PLL1_SRC_BCLK3 (0x3 << 11) +#define RT5677_PLL1_SRC_BCLK4 (0x4 << 11) +#define RT5677_PLL1_SRC_RCCLK (0x5 << 11) +#define RT5677_PLL1_SRC_SLIM (0x6 << 11) +#define RT5677_MCLK_SRC_MASK (0x1 << 10) +#define RT5677_MCLK_SRC_SFT 10 +#define RT5677_MCLK1_SRC (0x0 << 10) +#define RT5677_MCLK2_SRC (0x1 << 10) +#define RT5677_PLL1_PD_MASK (0x1 << 8) +#define RT5677_PLL1_PD_SFT 8 +#define RT5677_PLL1_PD_1 (0x0 << 8) +#define RT5677_PLL1_PD_2 (0x1 << 8) +#define RT5671_DAC_OSR_MASK (0x3 << 6) +#define RT5671_DAC_OSR_SFT 6 +#define RT5671_DAC_OSR_128 (0x0 << 6) +#define RT5671_DAC_OSR_64 (0x1 << 6) +#define RT5671_DAC_OSR_32 (0x2 << 6) +#define RT5671_ADC_OSR_MASK (0x3 << 4) +#define RT5671_ADC_OSR_SFT 4 +#define RT5671_ADC_OSR_128 (0x0 << 4) +#define RT5671_ADC_OSR_64 (0x1 << 4) +#define RT5671_ADC_OSR_32 (0x2 << 4) + +/* Global Clock Control 2 (0x81) */ +#define RT5677_PLL2_PR_SRC_MASK (0x1 << 15) +#define RT5677_PLL2_PR_SRC_SFT 15 +#define RT5677_PLL2_PR_SRC_MCLK1 (0x0 << 15) +#define RT5677_PLL2_PR_SRC_MCLK2 (0x1 << 15) +#define RT5677_PLL2_SRC_MASK (0x7 << 12) +#define RT5677_PLL2_SRC_SFT 12 +#define RT5677_PLL2_SRC_MCLK (0x0 << 12) +#define RT5677_PLL2_SRC_BCLK1 (0x1 << 12) +#define RT5677_PLL2_SRC_BCLK2 (0x2 << 12) +#define RT5677_PLL2_SRC_BCLK3 (0x3 << 12) +#define RT5677_PLL2_SRC_BCLK4 (0x4 << 12) +#define RT5677_PLL2_SRC_RCCLK (0x5 << 12) +#define RT5677_PLL2_SRC_SLIM (0x6 << 12) +#define RT5671_DSP_ASRC_O_SRC (0x3 << 10) +#define RT5671_DSP_ASRC_O_SRC_SFT 10 +#define RT5671_DSP_ASRC_O_MCLK (0x0 << 10) +#define RT5671_DSP_ASRC_O_PLL1 (0x1 << 10) +#define RT5671_DSP_ASRC_O_SLIM (0x2 << 10) +#define RT5671_DSP_ASRC_O_RCCLK (0x3 << 10) +#define RT5671_DSP_ASRC_I_SRC (0x3 << 8) +#define RT5671_DSP_ASRC_I_SRC_SFT 8 +#define RT5671_DSP_ASRC_I_MCLK (0x0 << 8) +#define RT5671_DSP_ASRC_I_PLL1 (0x1 << 8) +#define RT5671_DSP_ASRC_I_SLIM (0x2 << 8) +#define RT5671_DSP_ASRC_I_RCCLK (0x3 << 8) +#define RT5677_DSP_CLK_SRC_MASK (0x1 << 7) +#define RT5677_DSP_CLK_SRC_SFT 7 +#define RT5677_DSP_CLK_SRC_PLL2 (0x0 << 7) +#define RT5677_DSP_CLK_SRC_BYPASS (0x1 << 7) + +/* VAD Function Control 4 (0x9f) */ +#define RT5677_VAD_SRC_MASK (0x7 << 8) +#define RT5677_VAD_SRC_SFT 8 + +/* DSP InBound Control (0xa3) */ +#define RT5677_IB01_SRC_MASK (0x7 << 12) +#define RT5677_IB01_SRC_SFT 12 +#define RT5677_IB23_SRC_MASK (0x7 << 8) +#define RT5677_IB23_SRC_SFT 8 +#define RT5677_IB45_SRC_MASK (0x7 << 4) +#define RT5677_IB45_SRC_SFT 4 +#define RT5677_IB6_SRC_MASK (0x7 << 0) +#define RT5677_IB6_SRC_SFT 0 + +/* DSP InBound Control (0xa4) */ +#define RT5677_IB7_SRC_MASK (0x7 << 12) +#define RT5677_IB7_SRC_SFT 12 +#define RT5677_IB8_SRC_MASK (0x7 << 8) +#define RT5677_IB8_SRC_SFT 8 +#define RT5677_IB9_SRC_MASK (0x7 << 4) +#define RT5677_IB9_SRC_SFT 4 + +/* DSP In/OutBound Control (0xa5) */ +#define RT5677_SEL_SRC_OB23 (0x1 << 4) +#define RT5677_SEL_SRC_OB23_SFT 4 +#define RT5677_SEL_SRC_OB01 (0x1 << 3) +#define RT5677_SEL_SRC_OB01_SFT 3 +#define RT5677_SEL_SRC_IB45 (0x1 << 2) +#define RT5677_SEL_SRC_IB45_SFT 2 +#define RT5677_SEL_SRC_IB23 (0x1 << 1) +#define RT5677_SEL_SRC_IB23_SFT 1 +#define RT5677_SEL_SRC_IB01 (0x1 << 0) +#define RT5677_SEL_SRC_IB01_SFT 0 + +/* Virtual DSP Mixer Control (0xf7 0xf8 0xf9) */ +#define RT5677_DSP_IB_01_H (0x1 << 15) +#define RT5677_DSP_IB_01_H_SFT 15 +#define RT5677_DSP_IB_23_H (0x1 << 14) +#define RT5677_DSP_IB_23_H_SFT 14 +#define RT5677_DSP_IB_45_H (0x1 << 13) +#define RT5677_DSP_IB_45_H_SFT 13 +#define RT5677_DSP_IB_6_H (0x1 << 12) +#define RT5677_DSP_IB_6_H_SFT 12 +#define RT5677_DSP_IB_7_H (0x1 << 11) +#define RT5677_DSP_IB_7_H_SFT 11 +#define RT5677_DSP_IB_8_H (0x1 << 10) +#define RT5677_DSP_IB_8_H_SFT 10 +#define RT5677_DSP_IB_9_H (0x1 << 9) +#define RT5677_DSP_IB_9_H_SFT 9 +#define RT5677_DSP_IB_01_L (0x1 << 7) +#define RT5677_DSP_IB_01_L_SFT 7 +#define RT5677_DSP_IB_23_L (0x1 << 6) +#define RT5677_DSP_IB_23_L_SFT 6 +#define RT5677_DSP_IB_45_L (0x1 << 5) +#define RT5677_DSP_IB_45_L_SFT 5 +#define RT5677_DSP_IB_6_L (0x1 << 4) +#define RT5677_DSP_IB_6_L_SFT 4 +#define RT5677_DSP_IB_7_L (0x1 << 3) +#define RT5677_DSP_IB_7_L_SFT 3 +#define RT5677_DSP_IB_8_L (0x1 << 2) +#define RT5677_DSP_IB_8_L_SFT 2 +#define RT5677_DSP_IB_9_L (0x1 << 1) +#define RT5677_DSP_IB_9_L_SFT 1 + +/* Debug String Length */ +#define RT5677_REG_DISP_LEN 23 + +#define RT5677_NO_JACK BIT(0) +#define RT5677_HEADSET_DET BIT(1) +#define RT5677_HEADPHO_DET BIT(2) + +/* System Clock Source */ +enum { + RT5677_SCLK_S_MCLK, + RT5677_SCLK_S_PLL1, + RT5677_SCLK_S_RCCLK, +}; + +/* PLL1 Source */ +enum { + RT5677_PLL1_S_MCLK, + RT5677_PLL1_S_BCLK1, + RT5677_PLL1_S_BCLK2, + RT5677_PLL1_S_BCLK3, + RT5677_PLL1_S_BCLK4, +}; + +enum { + RT5677_AIF1, + RT5677_AIF2, + RT5677_AIF3, + RT5677_AIF4, + RT5677_AIF5, + RT5677_AIFS, +}; + +struct rt5677_pll_code { + bool m_bp; /* Indicates bypass m code or not. */ + bool k_bp; /* Indicates bypass k code or not. */ + int m_code; + int n_code; + int k_code; +}; + +struct rt5677_priv { + struct snd_soc_codec *codec; + struct rt5677_platform_data pdata; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck[RT5677_AIFS]; + int bclk[RT5677_AIFS]; + int master[RT5677_AIFS]; + int pll_src; + int pll_in; + int pll_out; +}; + +#endif /* __RT5677_H__ */ -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [Bug?] alsa-lib: Plug fails to convert float mono to float stereo
by Sergey 08 Jun '14

08 Jun '14

Plug fails to convert float mono to float stereo, which is required when plug is used for float-only slaves like "jack" or "ladspa". Steps to reproduce: #~/.asoundrc pcm.!default { type plug slave.pcm { type null } slave.channels 2 slave.format FLOAT_LE } # aplay -f FLOAT_LE /dev/zero A possible workaround is to use two plugs: pcm.!default { type plug slave.pcm { type plug slave.pcm { type null } slave.channels 2 slave.format FLOAT_LE } slave.format S32_LE } But shouldn't "plug" do that automatically? PS: the workaround makes `aplay -fFLOAT_LE /dev/zero` freeze on Ctrl+C, but that's a different alsa 1.0.27 regression that I have not tracked yet. -- Sergey

1 0

[alsa-devel] [PATCH] pcm: Fix channel map query when there are lots of maps
by Anssi Hannula 08 Jun '14

08 Jun '14

On most HDA Intel HDMI devices, the channel map list is approx. 500 ints in size, making the 256-sized buffer used by snd_pcm_query_chmaps_from_hw() too small and causing the query to fail (NULL result to caller, ENOMEM). Bump the buffer size to 2048 ints. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- Found while writing XBMC support. I don't like just bumping the buffer size, so I hope you have some idea of a better solution? I guess one way would be just to retry with a dynamically allocated bigger buffer on ENOMEM, if there is nothing better... src/pcm/pcm_hw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 9ff83a9..ed83197 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1051,7 +1051,7 @@ snd_pcm_query_chmaps_from_hw(int card, int dev, int subdev, { snd_ctl_t *ctl; snd_ctl_elem_id_t *id; - unsigned int tlv[256], *start; + unsigned int tlv[2048], *start; snd_pcm_chmap_query_t **map; int i, ret, nums; -- 1.8.4.5

2 2

[alsa-devel] On HD codec valid command working only on hwdep hda-verb and embedded on function snd_hda_multi_out_analog_open
by asd 08 Jun '14

08 Jun '14

Hi On patch_realtek can't run snd_hda_write.Only situations when run snd_hda_write is to run inside off snd_hda_multi_out_analog_open function (suppose to when sream is on codec is weak up and become online.

2 1

[alsa-devel] [PATCH] ASoC: Intel: Show Baytrail SST DSP firmware details during init
by Jarkko Nikula 06 Jun '14

06 Jun '14

DSP initialization complete message IPC_IA_FW_INIT_CMPLT is a large message carrying firmware details in mailbox. Read and show those details during init in order to be able to get that information to QA reports. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-ipc.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index d207b22ea330..67673a2c0f41 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -122,6 +122,26 @@ struct sst_byt_tstamp { u32 channel_peak[8]; } __packed; +struct sst_byt_fw_version { + u8 build; + u8 minor; + u8 major; + u8 type; +} __packed; + +struct sst_byt_fw_build_info { + u8 date[16]; + u8 time[16]; +} __packed; + +struct sst_byt_fw_init { + struct sst_byt_fw_version fw_version; + struct sst_byt_fw_build_info build_info; + u16 result; + u8 module_id; + u8 debug_info; +} __packed; + /* driver internal IPC message structure */ struct ipc_message { struct list_head list; @@ -868,6 +888,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) { struct sst_byt *byt; struct sst_fw *byt_sst_fw; + struct sst_byt_fw_init init; int err; dev_dbg(dev, "initialising Byt DSP IPC\n"); @@ -929,6 +950,15 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) goto boot_err; } + /* show firmware information */ + sst_dsp_inbox_read(byt->dsp, &init, sizeof(init)); + dev_info(byt->dev, "FW version: %02x.%02x.%02x.%02x\n", + init.fw_version.major, init.fw_version.minor, + init.fw_version.build, init.fw_version.type); + dev_info(byt->dev, "Build type: %x\n", init.fw_version.type); + dev_info(byt->dev, "Build date: %s %s\n", + init.build_info.date, init.build_info.time); + pdata->dsp = byt; byt->fw = byt_sst_fw; -- 2.0.0.rc2

2 2

[alsa-devel] Add support of alc667 and more codec rename
by Kailang 06 Jun '14

06 Jun '14

Hi Takashi, Attach has 3 patches. Many Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH 0/5] ARM: atmel: switch audio to CCF
by Bo Shen 06 Jun '14

06 Jun '14

As the sama5d3xek and at91sam9n12ek have switch to CCF, in order to support audio, also switch it to CCF. This patch series based on next-2014-0604 tag. Bo Shen (5): ASoC: atmel_wm8904: switch to CCF ASoC: wm8904: switch to CCF dt: wm8904: add device tree binding document ARM: atmel: sama5d3xek: switch sound to CCF ARM: atmel: at91sam9n12ek: switch sound to CCF Documentation/devicetree/bindings/sound/wm8904.txt | 32 ++++++++++++++ arch/arm/boot/dts/at91sam9n12ek.dts | 2 + arch/arm/boot/dts/sama5d3xmb.dtsi | 2 + sound/soc/atmel/atmel_wm8904.c | 50 ---------------------- sound/soc/codecs/wm8904.c | 13 ++++++ 5 files changed, 49 insertions(+), 50 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8904.txt -- 1.8.5.2

2 10

[alsa-devel] [PATCH v2 1/2] ASoC: export dapm_kcontrol_set/get_value functions
by Vinod Koul 06 Jun '14

06 Jun '14

The DSPs like Intel ones use the DPCM to represent the DSP toplogy. So when DAPM triggers the widget, the DSP needs to know the kcontrol values and pass on to DSP firmware, so export these for driver use Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/soc-dapm.h | 3 +++ sound/soc/soc-dapm.c | 6 ++++-- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 75020f5..abda467 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -368,6 +368,9 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_clock_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, + unsigned int value); +unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol); /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 98c1dc6..d831e73 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -327,14 +327,15 @@ static struct list_head *dapm_kcontrol_get_path_list( list_for_each_entry(path, dapm_kcontrol_get_path_list(kcontrol), \ list_kcontrol) -static unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) +unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); return data->value; } +EXPORT_SYMBOL_GPL(dapm_kcontrol_get_value); -static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, +bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, unsigned int value) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); @@ -349,6 +350,7 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, return true; } +EXPORT_SYMBOL_GPL(dapm_kcontrol_set_value); /** * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol -- 1.7.0.4

3 12

[alsa-devel] r8a7790 only has audio when clock is forced on
by Violeta Menendez Gonzalez 06 Jun '14

06 Jun '14

Hi all, We have a problem where audio on r8a7790 is not working due to it not sending interrupts. We've verified that the pins and the clocks are working and correctly configured. The codec is starting and producing correct bit and word clock signals We've made a hack to force /mstp10_clks /on that makes the audio work. Further investigation showed that it's only needed to force SSI(ALL) on, but, as far as we know, the manual doesn't say anything about it having to be kept on. We've also debugged this hack and saw that the registers have correct values. These are some of the debugging messages we have when the audio fails: Playing WAVE '../audio/r.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo cpg_mstp_clock_endisable: index 5, enable 1 cpg_mstp_clock_endisable: index 5, value read from group->smstpcr 0002ffe0 cpg_mstp_clock_endisable: index 5, value write in group->smstpcr 0002ffc0 cpg_mstp_clock_endisable: index 15, enable 1 cpg_mstp_clock_endisable: index 15, value read from group->smstpcr 0002ffc0 cpg_mstp_clock_endisable: index 15, value written in group->smstpcr 00027fc0 aplay: pcm_writecpg_mstp_clock_endisable: index 15, enable 0 cpg_mstp_clock_endisable: index 15, value read from group->smstpcr 00027fc0 cpg_mstp_clock_endisable: index 15, value write in group->smstpcr 0002ffc0 :1710: write errcpg_mstp_clock_endisable: index 5, enable 0 cpg_mstp_clock_endisable: index 5, value read from group->smstpcr 0002ffc0 cpg_mstp_clock_endisable: index 5, value write in group->smstpcr 0002ffe0 or: Input/output error This shows that the SSI (ALL) clock in bit 5 being enabled before the SSI0 and that it is also disabled in the reverse sequence as expected. Not sure what to do next or what the problem could be. Any new ideas are welcome, Thanks! -- Violeta Menéndez González http://www.codethink.co.uk/ Software Engineer Codethink - Providing Genius

6 13

[alsa-devel] [PATCH 00/11] ASoC: Add support for DAPM at the component level
by Lars-Peter Clausen 06 Jun '14

06 Jun '14

Hi, This series is the next step in componentization. It moves the DAPM support to the component level. This will allow any component to have DAPM widgets and routes, which was previously only possible for CODECs, and will allow any component to have DAPM widgets with controls (i.e. Mixers and MUXs), which was previously only possible for CODECs. At the end of this patchset the DAPM code will have (almost) no dependency on snd_soc_codec or snd_soc_platform and the dependency on snd_soc_component is confined to 4 small helper functions. The only reason why there is still snd_soc_codec/snd_soc_platform dependent code is because there are still a lot of drivers which use widget->codec/platform and dapm->codec/platform. So those still need to be initialized. This will be cleaned up in a separate series and once that's done the codec and platform fields will be removed from the widget and DAPM context struct. One of the major problems of moving DAPM to the component level is that the DAPM context is directly embedded in the snd_soc_codec and snd_soc_platform structs and there are many drivers directly accessing that embedded field. This means it is not possible to move the field without causing a bit of code churn in the drivers. The patch series solves this by adding two fields to the component struct. One pointer to a DAPM context, which points to the DAPM context that is used by the component. And one DAPM context, which is used as the default if no other DAPM context was provided. Neither of the fields is accessed directly, instead a inline helper function is added that returns the DAPM context for a component. This will make it possible to eventually seamlessly switch over once all direct references to CODEC and platform DAPM contexts have been eliminated. Depends on topic/component, topic/core, topic/dapm and topic/dt - Lars Lars-Peter Clausen (11): ASoC: Move name_prefix from CODEC to component ASoC: Move name and id from CODEC/platform to component ASoC: Split component registration into two steps ASoC: Auto disconnect pins from all DAPM contexts ASoC: Add helper functions to cast from DAPM context to CODEC/platform ASoC: Add a set_bias_level() callback to the DAPM context struct ASoC: Add DAPM support at the component level ASoC: Use component DAPM context for platforms ASoC: Add component level stream_event() and seq_notifier() support ASoC: Add component level set_bias_level() support ASoC: dapm: Remove DAI DAPM context include/sound/soc-dai.h | 1 - include/sound/soc-dapm.h | 6 +- include/sound/soc.h | 78 ++++++++-- include/trace/events/asoc.h | 6 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/soc-cache.c | 7 +- sound/soc/soc-compress.c | 8 +- sound/soc/soc-core.c | 328 ++++++++++++++++++++++------------------- sound/soc/soc-dapm.c | 180 ++++++++++++---------- sound/soc/soc-pcm.c | 4 +- 10 files changed, 365 insertions(+), 255 deletions(-) -- 1.8.0

3 15

[alsa-devel] [PATCH] ALSA: hda - add two new pin tables
by Hui Wang 06 Jun '14

06 Jun '14

These two new pin tables can fix headset mic problems for several new Dell machines. And also delete some machines from old quirk table since the existing pin talbes already cover them. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 47 +++++++++++++++++++++++++++++++++++++------ 1 file changed, 41 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 12fb411..003ba10 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4741,18 +4741,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), - SND_PCI_QUIRK(0x1028, 0x062c, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x064d, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0668, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0669, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0674, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x067e, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x067f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0680, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0684, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), @@ -4977,6 +4971,26 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { #endif .pins = (const struct hda_pintbl[]) { {0x12, 0x90a60160}, + {0x14, 0x90170120}, + {0x17, 0x90170140}, + {0x18, 0x40000000}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x411111f0}, + {0x1d, 0x41163b05}, + {0x1e, 0x411111f0}, + {0x21, 0x0321102f}, + }, + .value = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + }, + { + .codec = 0x10ec0255, + .subvendor = 0x1028, +#ifdef CONFIG_SND_DEBUG_VERBOSE + .name = "Dell", +#endif + .pins = (const struct hda_pintbl[]) { + {0x12, 0x90a60160}, {0x14, 0x90170130}, {0x17, 0x40000000}, {0x18, 0x411111f0}, @@ -6014,6 +6028,27 @@ static const struct snd_hda_pin_quirk alc662_pin_fixup_tbl[] = { .name = "Dell", #endif .pins = (const struct hda_pintbl[]) { + {0x12, 0x99a30140}, + {0x14, 0x90170110}, + {0x15, 0x0321101f}, + {0x16, 0x03011020}, + {0x18, 0x40000008}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x411111f0}, + {0x1d, 0x41000001}, + {0x1e, 0x411111f0}, + {0x1f, 0x411111f0}, + }, + .value = ALC668_FIXUP_AUTO_MUTE, + }, + { + .codec = 0x10ec0668, + .subvendor = 0x1028, +#ifdef CONFIG_SND_DEBUG_VERBOSE + .name = "Dell", +#endif + .pins = (const struct hda_pintbl[]) { {0x12, 0x99a30150}, {0x14, 0x90170110}, {0x15, 0x0321101f}, -- 1.8.1.2

2 1

[alsa-devel] Recording support for Roland SD-50 USB Audio device
by François Isabelle 05 Jun '14

05 Jun '14

Hi. I'm unable to record using the Roland SD-50 on Linux. My alsa configuration is available at the following address: http://www.alsa-project.org/db/?f=19aeffd0d30e952fdd12d589929fbe2a0f98ad7f There is something odd with playback as well. With the sound control panel ( unity-control-center ). The test output only works when the SD-50 input is selected from the input tab. So, I'm pretty sure there is simply some mix up in the mixer configuration... and I'd like to contribute making it work. ( I have some programming stills :-) ) Although Alsa with Jack and Ardour makes a pretty decent and promising environment for audio production on Linux. It's hard to believe that device support is still an issue after more than 20 years of Linux, darn manufacturers ! On another note, if anyone has a good recommendation for USB Midi+Audio + Sound Module to replace this SD-50 on Linux , I'd be willing to hear about it... I want the sound module to be able to use it when not connected to a computer. I want USB because I don't want to take it off with the laptop as well. I really though the SD-50 would make my day... Thank you. François

3 8

[alsa-devel] [PATCH] ASoC: sigmadsp: remove I2C support
by Arnd Bergmann 05 Jun '14

05 Jun '14

The newly added ADAU1781 codec comes with separate drivers for I2C and SPI modes, which gives us a new variation of the dependency problems: If SPI is enabled and the I2C core is a loadable module, selecting SND_SOC_ADAU1781_SPI as built-in leads to a link error because the common base code from sigmadsp.c now also gets built-in: sound/built-in.o: In function `sigma_action_write_i2c': :(.text+0x5d8d4): undefined reference to `i2c_master_send' Rather than adding yet more complex Kconfig logic, this converts the last user of process_sigma_firmware() to process_sigma_firmware_regmap(), which seems trivial after the driver itself now uses regmap as well. With that done, there is no more linker dependency from SND_SOC_ADAU1781_SPI to i2c_master_send. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Lars-Peter Clausen <lars(a)metafoo.de> --- I hope this is the last ASoC patch I have for the moment, sorry for spamming everybody ;-) sound/soc/codecs/adau1701.c | 5 +++-- sound/soc/codecs/sigmadsp.c | 27 --------------------------- sound/soc/codecs/sigmadsp.h | 3 --- 3 files changed, 3 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index d71c59c..1b5e306 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -239,7 +239,6 @@ static int adau1701_reg_read(void *context, unsigned int reg, static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) { struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *client = to_i2c_client(codec->dev); int ret; if (clkdiv != ADAU1707_CLKDIV_UNSET && @@ -282,7 +281,9 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) * know the correct PLL setup */ if (clkdiv != ADAU1707_CLKDIV_UNSET) { - ret = process_sigma_firmware(client, ADAU1701_FIRMWARE); + ret = process_sigma_firmware_regmap(codec->dev, + adau1701->regmap, + ADAU1701_FIRMWARE); if (ret) { dev_warn(codec->dev, "Failed to load firmware\n"); return ret; diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 4068f24..209f98f 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -10,7 +10,6 @@ #include <linux/delay.h> #include <linux/firmware.h> #include <linux/kernel.h> -#include <linux/i2c.h> #include <linux/regmap.h> #include <linux/module.h> @@ -198,30 +197,6 @@ static int _process_sigma_firmware(struct device *dev, return ret; } -#if IS_ENABLED(CONFIG_I2C) - -static int sigma_action_write_i2c(void *control_data, - const struct sigma_action *sa, size_t len) -{ - return i2c_master_send(control_data, (const unsigned char *)&sa->addr, - len); -} - -int process_sigma_firmware(struct i2c_client *client, const char *name) -{ - struct sigma_firmware ssfw; - - ssfw.control_data = client; - ssfw.write = sigma_action_write_i2c; - - return _process_sigma_firmware(&client->dev, &ssfw, name); -} -EXPORT_SYMBOL(process_sigma_firmware); - -#endif - -#if IS_ENABLED(CONFIG_REGMAP) - static int sigma_action_write_regmap(void *control_data, const struct sigma_action *sa, size_t len) { @@ -241,6 +216,4 @@ int process_sigma_firmware_regmap(struct device *dev, struct regmap *regmap, } EXPORT_SYMBOL(process_sigma_firmware_regmap); -#endif - MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp.h b/sound/soc/codecs/sigmadsp.h index e439cbd..38a455b 100644 --- a/sound/soc/codecs/sigmadsp.h +++ b/sound/soc/codecs/sigmadsp.h @@ -12,9 +12,6 @@ #include <linux/device.h> #include <linux/regmap.h> -struct i2c_client; - -extern int process_sigma_firmware(struct i2c_client *client, const char *name); extern int process_sigma_firmware_regmap(struct device *dev, struct regmap *regmap, const char *name);

2 4

[alsa-devel] [PATCH 1/2] ALSA: soundcore: coding style fixes
by Daniel Mack 05 Jun '14

05 Jun '14

sound/sounc_core.c has a larger number of style issues. Fix them to make the code more readable. No functional change. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/sound_core.c | 192 ++++++++++++++++++++++++----------------------------- 1 file changed, 85 insertions(+), 107 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 11e953a..bf9e41e 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -68,21 +68,16 @@ module_exit(cleanup_soundcore); #ifdef CONFIG_SOUND_OSS_CORE /* * OSS sound core handling. Breaks out sound functions to submodules - * - * Author: Alan Cox <alan(a)lxorguk.ukuu.org.uk> - * - * Fixes: * + * Author: Alan Cox <alan(a)lxorguk.ukuu.org.uk> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * as published by the Free Software Foundation; either version * 2 of the License, or (at your option) any later version. * - * -------------------- - * * Top level handler for the sound subsystem. Various devices can - * plug into this. The fact they don't all go via OSS doesn't mean + * plug into this. The fact they don't all go via OSS doesn't mean * they don't have to implement the OSS API. There is a lot of logic * to keeping much of the OSS weight out of the code in a compatibility * module, but it's up to the driver to rember to load it... @@ -111,8 +106,7 @@ module_exit(cleanup_soundcore); #define SOUND_STEP 16 -struct sound_unit -{ +struct sound_unit { int unit_minor; const struct file_operations *unit_fops; struct sound_unit *next; @@ -158,8 +152,7 @@ module_param(preclaim_oss, int, 0444); static int soundcore_open(struct inode *, struct file *); -static const struct file_operations soundcore_fops = -{ +static const struct file_operations soundcore_fops = { /* We must have an owner or the module locking fails */ .owner = THIS_MODULE, .open = soundcore_open, @@ -171,71 +164,66 @@ static const struct file_operations soundcore_fops = * join into it. Called with the lock asserted */ -static int __sound_insert_unit(struct sound_unit * s, struct sound_unit **list, const struct file_operations *fops, int index, int low, int top) +static int __sound_insert_unit(struct sound_unit *s, struct sound_unit **list, + const struct file_operations *fops, + int index, int low, int top) { - int n=low; + int n = low; if (index < 0) { /* first free */ + while (*list && (*list)->unit_minor < n) + list = &((*list)->next); - while (*list && (*list)->unit_minor<n) - list=&((*list)->next); - - while(n<top) - { + while (n < top) { /* Found a hole ? */ - if(*list==NULL || (*list)->unit_minor>n) + if (*list == NULL || (*list)->unit_minor > n) break; - list=&((*list)->next); - n+=SOUND_STEP; + + list = &((*list)->next); + n += SOUND_STEP; } - if(n>=top) + if (n >= top) return -ENOENT; } else { - n = low+(index*16); + n = low + (index * 16); while (*list) { - if ((*list)->unit_minor==n) + if ((*list)->unit_minor == n) return -EBUSY; - if ((*list)->unit_minor>n) + if ((*list)->unit_minor > n) break; - list=&((*list)->next); + list = &((*list)->next); } - } - - /* - * Fill it in - */ - - s->unit_minor=n; - s->unit_fops=fops; - - /* - * Link it - */ - - s->next=*list; - *list=s; - - + } + + /* Fill it in */ + s->unit_minor = n; + s->unit_fops = fops; + + /* Link it */ + s->next = *list; + *list = s; + return n; } /* * Remove a node from the chain. Called with the lock asserted */ - -static struct sound_unit *__sound_remove_unit(struct sound_unit **list, int unit) +static struct sound_unit *__sound_remove_unit(struct sound_unit **list, + int unit) { - while(*list) - { - struct sound_unit *p=*list; - if(p->unit_minor==unit) - { - *list=p->next; + while (*list) { + struct sound_unit *p = *list; + + if (p->unit_minor == unit) { + *list = p->next; return p; } - list=&(p->next); + + list = &p->next; } + printk(KERN_ERR "Sound device %d went missing!\n", unit); return NULL; } @@ -250,12 +238,15 @@ static DEFINE_SPINLOCK(sound_loader_lock); * Allocate the controlling structure and add it to the sound driver * list. Acquires locks as needed */ - -static int sound_insert_unit(struct sound_unit **list, const struct file_operations *fops, int index, int low, int top, const char *name, umode_t mode, struct device *dev) +static int sound_insert_unit(struct sound_unit **list, + const struct file_operations *fops, + int index, int low, int top, const char *name, + umode_t mode, struct device *dev) { - struct sound_unit *s = kmalloc(sizeof(*s), GFP_KERNEL); + struct sound_unit *s; int r; + s = kmalloc(sizeof(*s), GFP_KERNEL); if (!s) return -ENOMEM; @@ -263,7 +254,7 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati retry: r = __sound_insert_unit(s, list, fops, index, low, top); spin_unlock(&sound_loader_lock); - + if (r < 0) goto fail; else if (r < SOUND_STEP) @@ -292,7 +283,8 @@ retry: } device_create(sound_class, dev, MKDEV(SOUND_MAJOR, s->unit_minor), - NULL, "%s", s->name+6); + NULL, "%s", s->name + 6); + return s->unit_minor; fail: @@ -305,7 +297,6 @@ fail: * completed the removal before their file operations become * invalid. */ - static void sound_remove_unit(struct sound_unit **list, int unit) { struct sound_unit *p; @@ -357,7 +348,6 @@ static struct sound_unit *chains[SOUND_STEP]; * Return: The allocated number is returned on success. On failure, * a negative error code is returned. */ - int register_sound_special_device(const struct file_operations *fops, int unit, struct device *dev) { @@ -367,69 +357,66 @@ int register_sound_special_device(const struct file_operations *fops, int unit, char _name[16]; switch (chain) { - case 0: + case 0: name = "mixer"; break; - case 1: + case 1: name = "sequencer"; if (unit >= SOUND_STEP) goto __unknown; max_unit = unit + 1; break; - case 2: + case 2: name = "midi"; break; - case 3: + case 3: name = "dsp"; break; - case 4: + case 4: name = "audio"; break; - case 5: + case 5: name = "dspW"; break; - case 8: + case 8: name = "sequencer2"; if (unit >= SOUND_STEP) goto __unknown; max_unit = unit + 1; break; - case 9: + case 9: name = "dmmidi"; break; - case 10: + case 10: name = "dmfm"; break; - case 12: + case 12: name = "adsp"; break; - case 13: + case 13: name = "amidi"; break; - case 14: + case 14: name = "admmidi"; break; - default: - { - __unknown: - sprintf(_name, "unknown%d", chain); - if (unit >= SOUND_STEP) - strcat(_name, "-"); - name = _name; - } + default: +__unknown: + sprintf(_name, "unknown%d", chain); + if (unit >= SOUND_STEP) + strcat(_name, "-"); + name = _name; break; } + return sound_insert_unit(&chains[chain], fops, -1, unit, max_unit, name, S_IRUSR | S_IWUSR, dev); } - EXPORT_SYMBOL(register_sound_special_device); int register_sound_special(const struct file_operations *fops, int unit) { return register_sound_special_device(fops, unit, NULL); } - EXPORT_SYMBOL(register_sound_special); /** @@ -443,13 +430,11 @@ EXPORT_SYMBOL(register_sound_special); * Return: On success, the allocated number is returned. On failure, * a negative error code is returned. */ - int register_sound_mixer(const struct file_operations *fops, int dev) { return sound_insert_unit(&chains[0], fops, dev, 0, 128, "mixer", S_IRUSR | S_IWUSR, NULL); } - EXPORT_SYMBOL(register_sound_mixer); /** @@ -469,14 +454,13 @@ int register_sound_midi(const struct file_operations *fops, int dev) return sound_insert_unit(&chains[2], fops, dev, 2, 130, "midi", S_IRUSR | S_IWUSR, NULL); } - EXPORT_SYMBOL(register_sound_midi); /* * DSP's are registered as a triple. Register only one and cheat * in open - see below. */ - + /** * register_sound_dsp - register a DSP device * @fops: File operations for the driver @@ -497,7 +481,6 @@ int register_sound_dsp(const struct file_operations *fops, int dev) return sound_insert_unit(&chains[3], fops, dev, 3, 131, "dsp", S_IWUSR | S_IRUSR, NULL); } - EXPORT_SYMBOL(register_sound_dsp); /** @@ -514,7 +497,6 @@ void unregister_sound_special(int unit) { sound_remove_unit(&chains[unit % SOUND_STEP], unit); } - EXPORT_SYMBOL(unregister_sound_special); /** @@ -529,7 +511,6 @@ void unregister_sound_mixer(int unit) { sound_remove_unit(&chains[0], unit); } - EXPORT_SYMBOL(unregister_sound_mixer); /** @@ -544,7 +525,6 @@ void unregister_sound_midi(int unit) { sound_remove_unit(&chains[2], unit); } - EXPORT_SYMBOL(unregister_sound_midi); /** @@ -561,21 +541,19 @@ void unregister_sound_dsp(int unit) { sound_remove_unit(&chains[3], unit); } - - EXPORT_SYMBOL(unregister_sound_dsp); static struct sound_unit *__look_for_unit(int chain, int unit) { struct sound_unit *s; - - s=chains[chain]; - while(s && s->unit_minor <= unit) - { - if(s->unit_minor==unit) + + s = chains[chain]; + while (s && s->unit_minor <= unit) { + if (s->unit_minor == unit) return s; - s=s->next; + s = s->next; } + return NULL; } @@ -586,14 +564,13 @@ static int soundcore_open(struct inode *inode, struct file *file) struct sound_unit *s; const struct file_operations *new_fops = NULL; - chain=unit&0x0F; - if(chain==4 || chain==5) /* dsp/audio/dsp16 */ - { - unit&=0xF0; - unit|=3; - chain=3; + chain = unit & 0x0F; + if (chain == 4 || chain == 5) { /* dsp/audio/dsp16 */ + unit &= 0xF0; + unit |= 3; + chain = 3; } - + spin_lock(&sound_loader_lock); s = __look_for_unit(chain, unit); if (s) @@ -608,8 +585,8 @@ static int soundcore_open(struct inode *inode, struct file *file) * ALSA toplevel modules for soundcards, thus we need * load them at first. [Jaroslav Kysela <perex(a)jcu.cz>] */ - request_module("sound-slot-%i", unit>>4); - request_module("sound-service-%i-%i", unit>>4, chain); + request_module("sound-slot-%i", unit >> 4); + request_module("sound-service-%i-%i", unit >> 4, chain); /* * sound-slot/service-* module aliases are scheduled @@ -636,10 +613,11 @@ static int soundcore_open(struct inode *inode, struct file *file) replace_fops(file, new_fops); if (file->f_op->open) - err = file->f_op->open(inode,file); + err = file->f_op->open(inode, file); return err; } + return -ENODEV; } -- 1.9.3

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[alsa-devel] New codec support for ALC891
by Kailang 05 Jun '14

05 Jun '14

Hi Takashi, We have a new codec ALC891. The support patch as attach. Many Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH - ucm 1/1] ucm: set active_verb when set_verb was successful
by nrubinstein＠aldebaran-robotics.com 05 Jun '14

05 Jun '14

From: Noé Rubinstein <nrubinstein(a)aldebaran-robotics.com> This seems to be a bug: execute_sequence returns zero on success, therefore the current code changes the active_verb iff execute_sequence failed. Signed-off-by: Noé Rubinstein <nrubinstein(a)aldebaran-robotics.com> diff --git a/src/ucm/main.c b/src/ucm/main.c index 0494728..3c0d7e4 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -623,7 +623,7 @@ static int set_verb(snd_use_case_mgr_t *uc_mgr, &verb->value_list, &uc_mgr->value_list, NULL); - if (enable && err >= 0) + if (enable && err == 0) uc_mgr->active_verb = verb; return err; } -- 1.9.2

2 1

[alsa-devel] Rate or plug plugin issue with snd-pcsp
by Alexander E. Patrakov 05 Jun '14

05 Jun '14

Hello. While testing application support for weird-rate cards (such as snd-pcsp), I found a bug in alsa-lib. Alsa-plugins are installed, so the "libspeex (builtin)" rate converter is used. This bug already exists in alsa-plugins 1.0.27.2, so not a regression and thus should not block 1.0.28. $ file test.wav test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 32000 Hz $ aplay -vvv -D plughw:pcsp test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Stereo aplay: set_params:1297: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: NONE PERIOD_TIME: (125006 125007) PERIOD_SIZE: NONE PERIOD_BYTES: (16000 16004) PERIODS: (3 4) BUFFER_TIME: 500000 BUFFER_SIZE: 16000 BUFFER_BYTES: 64000 TICK_TIME: 0 The pcsp device supports the sample rate of 37286 Hz. Files with 44100 Hz sample rate work fine. $ aplay -vv -D plughw:pcsp -f cd /dev/zero Playing raw data '/dev/zero' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0*0,5 + 1*0,5 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22050 period_size : 5512 period_time : 125006 tstamp_mode : NONE period_step : 1 avail_min : 5512 period_event : 0 start_threshold : 22050 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 6206523236469964800 Slave: Rate conversion PCM (37286, sformat=S16_LE) Converter: libspeex (builtin) Protocol version: 10002 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22050 period_size : 5512 period_time : 125006 tstamp_mode : NONE period_step : 1 avail_min : 5512 period_event : 0 start_threshold : 22050 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 6206523236469964800 Slave: Hardware PCM card 2 'pcsp' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 37286 exact rate : 37286 (37286/1) msbits : 16 buffer_size : 18644 period_size : 4661 period_time : 125006 tstamp_mode : NONE period_step : 1 avail_min : 4661 period_event : 0 start_threshold : 18644 stop_threshold : 18644 silence_threshold: 0 silence_size : 0 boundary : 5247819465793470464 appl_ptr : 0 hw_ptr : 0 #+ | 00%^C -- Alexander E. Patrakov

2 1

[alsa-devel] [PATCH] ASoC: pxa: add I2C dependencies as needed
by Arnd Bergmann 04 Jun '14

04 Jun '14

We have in the past added 'depends on I2C' for some of the PXA boards after hitting randconfig build bugs. I have seens a couple of new bugs in this area during the linux-next cycle for 3.16, after it became possible to build some more PXA machines with I2C disabled. To shut this up for good, this adds the dependency to every board that uses I2C as the interface to the codec. I have gone through all board files and verified that they all either use AC97 or I2C, and this annotates the latter. Some of these already enable I2C from mach-pxa/Kconfig, but since that can change it's better to be explicit here. The link error that can result otherwise happens when CONFIG_I2C is set to 'm' and the codec driver is built-in as a result of being selected by the platform specific glue. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index dcf0c08..e195b038 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -40,7 +40,7 @@ config SND_MMP_SOC_SSPA config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" - depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx + depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8731 help @@ -49,7 +49,7 @@ config SND_PXA2XX_SOC_CORGI config SND_PXA2XX_SOC_SPITZ tristate "SoC Audio support for Sharp Zaurus SL-Cxx00" - depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 + depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8750 help @@ -58,7 +58,7 @@ config SND_PXA2XX_SOC_SPITZ config SND_PXA2XX_SOC_Z2 tristate "SoC Audio support for Zipit Z2" - depends on SND_PXA2XX_SOC && MACH_ZIPIT2 + depends on SND_PXA2XX_SOC && MACH_ZIPIT2 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8750 help @@ -66,7 +66,7 @@ config SND_PXA2XX_SOC_Z2 config SND_PXA2XX_SOC_POODLE tristate "SoC Audio support for Poodle" - depends on SND_PXA2XX_SOC && MACH_POODLE + depends on SND_PXA2XX_SOC && MACH_POODLE && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8731 help @@ -181,7 +181,7 @@ config SND_PXA2XX_SOC_HX4700 config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" - depends on SND_PXA2XX_SOC && MACH_MAGICIAN + depends on SND_PXA2XX_SOC && MACH_MAGICIAN && I2C select SND_PXA2XX_SOC_I2S select SND_PXA_SOC_SSP select SND_SOC_UDA1380

3 4

[alsa-devel] [PATCH] sound: tlv320aic32x4: Fixed Coding style issues
by ShaRab Shaik 04 Jun '14

04 Jun '14

>From 617d7a25f3c0c08e0b95b24de2d1ea0f3f4b98c7 Mon Sep 17 00:00:00 2001 From: Shahina Shaik <sharab.shaik(a)gmail.com> Date: Wed, 4 Jun 2014 16:27:19 +0530 Subject: [PATCH] sound: tlv320aic32x4: Fixed Coding style issues. Fixed brace coding style, lines morethan 80 characters issues. Replaced printk with pr_err as pr_err is prefered than printk. Signed-off-by: Shahina Shaik <sharab.shaik(a)gmail.com> --- sound/soc/codecs/tlv320aic32x4.c | 34 +++++++++++++++++++--------------- 1 files changed, 19 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 1d9b117..9e45490 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -306,7 +306,7 @@ static inline int aic32x4_get_divs(int mclk, int rate) return i; } } - printk(KERN_ERR "aic32x4: master clock and sample rate is not supported\n"); + pr_err(KERN_ERR "aic32x4: master clock and sample rate is not supported\n"); return -EINVAL; } @@ -323,7 +323,7 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, aic32x4->sysclk = freq; return 0; } - printk(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n"); + pr_err(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n"); return -EINVAL; } @@ -349,7 +349,7 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) case SND_SOC_DAIFMT_CBS_CFS: break; default: - printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); + pr_err(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); return -EINVAL; } @@ -374,7 +374,7 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) (AIC32X4_LEFT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT); break; default: - printk(KERN_ERR "aic32x4: invalid DAI interface format\n"); + pr_err(KERN_ERR "aic32x4: invalid DAI interface format\n"); return -EINVAL; } @@ -395,7 +395,7 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream, i = aic32x4_get_divs(aic32x4->sysclk, params_rate(params)); if (i < 0) { - printk(KERN_ERR "aic32x4: sampling rate not supported\n"); + pr_err(KERN_ERR "aic32x4: sampling rate not supported\n"); return i; } @@ -626,32 +626,36 @@ static int aic32x4_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN | AIC32X4_MICBIAS_2075V); } - if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) { + if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); - } + tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ? AIC32X4_LDOCTLEN : 0; snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg); tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE); - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) { + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) tmp_reg |= AIC32X4_LDOIN_18_36; - } - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) { + + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) tmp_reg |= AIC32X4_LDOIN2HP; - } + snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg); /* Mic PGA routing */ if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) - snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K); + snd_soc_write(codec, AIC32X4_LMICPGANIN, + AIC32X4_LMICPGANIN_IN2R_10K); else - snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_CM1L_10K); + snd_soc_write(codec, AIC32X4_LMICPGANIN, + AIC32X4_LMICPGANIN_CM1L_10K); if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) - snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K); + snd_soc_write(codec, AIC32X4_RMICPGANIN, + AIC32X4_RMICPGANIN_IN1L_10K); else - snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_CM1R_10K); + snd_soc_write(codec, AIC32X4_RMICPGANIN, + AIC32X4_RMICPGANIN_CM1R_10K); aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

2 2

[alsa-devel] [PATCH] ASoC: samsung: Add I2C dependency for snow
by Arnd Bergmann 04 Jun '14

04 Jun '14

Both codecs used by snow, max98090 and max98095 require the use of I2C, so we can only select this driver if I2C is there, otherwise we get a build error like: codecs/max98090.c:2494:1: warning: data definition has no type or storage class [enabled by default] module_i2c_driver(max98090_i2c_driver); ^ codecs/max98095.c:2443:1: warning: data definition has no type or storage class [enabled by default] module_i2c_driver(max98095_i2c_driver); ^ This adds one more I2C dependency to the hundreds we already have. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Tushar Behera <tushar.behera(a)linaro.org> --- diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index a7cb8f6..cb7be4f 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -232,7 +232,7 @@ config SND_SOC_LITTLEMILL config SND_SOC_SNOW tristate "Audio support for Google Snow boards" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SOC_MAX98090 select SND_SOC_MAX98095 select SND_SAMSUNG_I2S

3 2

[alsa-devel] [PATCH v2] ASoC: arizona: Implement TDM support for Arizona devices
by Charles Keepax 04 Jun '14

04 Jun '14

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Removed check of TDM width to requested stream word length include/linux/mfd/arizona/core.h | 3 + sound/soc/codecs/arizona.c | 78 +++++++++++++++++++++++++++++++++++--- 2 files changed, 75 insertions(+), 6 deletions(-) diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index 5cf8b91..11783b5 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -110,6 +110,9 @@ struct arizona { int clk32k_ref; struct snd_soc_dapm_context *dapm; + + int tdm_width[ARIZONA_MAX_AIF]; + int tdm_slots[ARIZONA_MAX_AIF]; }; int arizona_clk32k_enable(struct arizona *arizona); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 29e198f..e77f61c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1185,7 +1185,10 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, int base = dai->driver->base; const int *rates; int i, ret, val; + int channels = params_channels(params); int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1]; + int tdm_width = arizona->tdm_width[dai->id - 1]; + int tdm_slots = arizona->tdm_slots[dai->id - 1]; int bclk, lrclk, wl, frame, bclk_target; if (params_rate(params) % 8000) @@ -1193,18 +1196,27 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, else rates = &arizona_48k_bclk_rates[0]; - bclk_target = snd_soc_params_to_bclk(params); - if (chan_limit && chan_limit < params_channels(params)) { + if (tdm_slots) { + arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", + tdm_slots, tdm_width); + bclk_target = tdm_slots * tdm_width * params_rate(params); + channels = tdm_slots; + } else { + bclk_target = snd_soc_params_to_bclk(params); + } + + if (chan_limit && chan_limit < channels) { arizona_aif_dbg(dai, "Limiting to %d channels\n", chan_limit); - bclk_target /= params_channels(params); + bclk_target /= channels; bclk_target *= chan_limit; } - /* Force stereo for I2S mode */ + /* Force multiple of 2 channels for I2S mode */ val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); - if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) { + if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) { arizona_aif_dbg(dai, "Forcing stereo mode\n"); - bclk_target *= 2; + bclk_target /= channels; + bclk_target *= channels + 1; } for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { @@ -1324,9 +1336,63 @@ static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) ARIZONA_AIF1_TRI, reg); } +static void arizona_set_channels_to_mask(struct snd_soc_dai *dai, + unsigned int base, + int channels, unsigned int mask) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int slot, i; + + for (i = 0; i < channels; ++i) { + slot = ffs(mask) - 1; + if (slot < 0) + return; + + regmap_write(arizona->regmap, base + i, slot); + + mask &= ~(1 << slot); + } + + if (mask) + arizona_aif_warn(dai, "Too many channels in TDM mask\n"); +} + +static int arizona_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int base = dai->driver->base; + int rx_max_chan = dai->driver->playback.channels_max; + int tx_max_chan = dai->driver->capture.channels_max; + + /* Only support TDM for the physical AIFs */ + if (dai->id > ARIZONA_MAX_AIF) + return -ENOTSUPP; + + if (slots == 0) { + tx_mask = (1 << tx_max_chan) - 1; + rx_mask = (1 << rx_max_chan) - 1; + } + + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_3, + tx_max_chan, tx_mask); + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_11, + rx_max_chan, rx_mask); + + arizona->tdm_width[dai->id - 1] = slot_width; + arizona->tdm_slots[dai->id - 1] = slots; + + return 0; +} + const struct snd_soc_dai_ops arizona_dai_ops = { .startup = arizona_startup, .set_fmt = arizona_set_fmt, + .set_tdm_slot = arizona_set_tdm_slot, .hw_params = arizona_hw_params, .set_sysclk = arizona_dai_set_sysclk, .set_tristate = arizona_set_tristate, -- 1.7.2.5

2 1

[alsa-devel] [PATCH v2 1/1] ALSA: seq: Continue broadcasting events to ports if one of them fails
by Adam Goode 04 Jun '14

04 Jun '14

Sometimes PORT_EXIT messages are lost when a process is exiting. This happens if you subscribe to the announce port with client A, then subscribe to the announce port with client B, then kill client A. Client B will not see the PORT_EXIT message because client A's port is closing and is earlier in the announce port subscription list. The for each loop will try to send the announcement to client A and fail, then will stop trying to broadcast to other ports. Killing B works fine since the announcement will already have gone to A. The CLIENT_EXIT message does not get lost. How to reproduce problem: *** termA $ aseqdump -p 0:1 0:1 Port subscribed 0:1 -> 128:0 *** termB $ aseqdump -p 0:1 *** termA 0:1 Client start client 129 0:1 Port start 129:0 0:1 Port subscribed 0:1 -> 129:0 *** termB 0:1 Port subscribed 0:1 -> 129:0 *** termA ^C *** termB 0:1 Client exit client 128 <--- expected Port exit as well (before client exit) Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 9ca5e64..225c7315 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -660,7 +660,7 @@ static int deliver_to_subscribers(struct snd_seq_client *client, int atomic, int hop) { struct snd_seq_subscribers *subs; - int err = 0, num_ev = 0; + int err, result = 0, num_ev = 0; struct snd_seq_event event_saved; struct snd_seq_client_port *src_port; struct snd_seq_port_subs_info *grp; @@ -685,8 +685,12 @@ static int deliver_to_subscribers(struct snd_seq_client *client, subs->info.flags & SNDRV_SEQ_PORT_SUBS_TIME_REAL); err = snd_seq_deliver_single_event(client, event, 0, atomic, hop); - if (err < 0) - break; + if (err < 0) { + /* save first error that occurs and continue */ + if (!result) + result = err; + continue; + } num_ev++; /* restore original event record */ *event = event_saved; @@ -697,7 +701,7 @@ static int deliver_to_subscribers(struct snd_seq_client *client, up_read(&grp->list_mutex); *event = event_saved; /* restore */ snd_seq_port_unlock(src_port); - return (err < 0) ? err : num_ev; + return (result < 0) ? result : num_ev; } @@ -709,7 +713,7 @@ static int port_broadcast_event(struct snd_seq_client *client, struct snd_seq_event *event, int atomic, int hop) { - int num_ev = 0, err = 0; + int num_ev = 0, err, result = 0; struct snd_seq_client *dest_client; struct snd_seq_client_port *port; @@ -724,14 +728,18 @@ static int port_broadcast_event(struct snd_seq_client *client, err = snd_seq_deliver_single_event(NULL, event, SNDRV_SEQ_FILTER_BROADCAST, atomic, hop); - if (err < 0) - break; + if (err < 0) { + /* save first error that occurs and continue */ + if (!result) + result = err; + continue; + } num_ev++; } read_unlock(&dest_client->ports_lock); snd_seq_client_unlock(dest_client); event->dest.port = SNDRV_SEQ_ADDRESS_BROADCAST; /* restore */ - return (err < 0) ? err : num_ev; + return (result < 0) ? result : num_ev; } /* @@ -741,7 +749,7 @@ static int port_broadcast_event(struct snd_seq_client *client, static int broadcast_event(struct snd_seq_client *client, struct snd_seq_event *event, int atomic, int hop) { - int err = 0, num_ev = 0; + int err, result = 0, num_ev = 0; int dest; struct snd_seq_addr addr; @@ -760,12 +768,16 @@ static int broadcast_event(struct snd_seq_client *client, err = snd_seq_deliver_single_event(NULL, event, SNDRV_SEQ_FILTER_BROADCAST, atomic, hop); - if (err < 0) - break; + if (err < 0) { + /* save first error that occurs and continue */ + if (!result) + result = err; + continue; + } num_ev += err; } event->dest = addr; /* restore */ - return (err < 0) ? err : num_ev; + return (result < 0) ? result : num_ev; } -- 2.0.0.526.g5318336

2 1

[alsa-devel] Correction for comment about firewire stack for tag 'sound-3.16-rc1'
by Takashi Sakamoto 04 Jun '14

04 Jun '14

Iwai-san, I find a typo in the comment. If it's impossible to correct it, please ignore this message. > FireWire-audio: > - FireWire audio stack enhancement (AMDTP, MIDI) > - BeBbB-based devices support > - Fireworks-based device support s/BeBbB/BeBoB/ ~ ~ You can see "what's BeBoB" in sound/firewire/bebob/bebob.c:10. I use BeBoB as a name of module, names of directory/files and Kconfig so it's better to correct this typo. And I strongly hope to describe a support for 'incoming isochronous stream' and 'duplex streams with timestamp synchronization'. They're main mechanisms which BeBoB/Fireworks need but previous stack didn't have. If it's too detailed for comment of tag, please keep it what it is. Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

2 1

[alsa-devel] [PATCH 0/6] Fix some minor issues for Fireworks/BeBoB drivers
by Takashi Sakamoto 04 Jun '14

04 Jun '14

Remove some unused or meaningless functions and improve some codes and comments. Takashi Sakamoto (6): fireworks/bebob: Shorten critical section for stream_stop_duplex() fireworks: Use safer way to arrange ring buffer pointer fireworks: Improve comments about Fireworks transaction fireworks: Remove a constant over width to which it's applied fireworks: Remove meaningless mutex_unlock() bebob: Remove unused function prototype sound/firewire/bebob/bebob.h | 2 -- sound/firewire/bebob/bebob_stream.c | 4 ++-- sound/firewire/fireworks/fireworks.c | 1 - sound/firewire/fireworks/fireworks.h | 1 - sound/firewire/fireworks/fireworks_hwdep.c | 2 +- sound/firewire/fireworks/fireworks_stream.c | 4 ++-- sound/firewire/fireworks/fireworks_transaction.c | 18 +++++++++--------- 7 files changed, 14 insertions(+), 18 deletions(-) -- 1.8.3.2

2 13

[alsa-devel] [PATCH v6] ASoC: add RT286 CODEC driver
by bardliao＠realtek.com 04 Jun '14

04 Jun '14

From: Bard Liao <bardliao(a)realtek.com> ALC286 is a dual mode codec, which can run as HD-A or I2S mode. It is controlled by HD-A verb commands via I2C protocol. The following is the I/O difference between ALC286 and general I2S codecs. 1. A HD-A verb command contains three parts, NID, VID, and PID. And an I2S command contains only two parts: address and data. 2. Not only the register address is written, but the read command also includes the entire write command. 3. rt286 uses different registers for read and write the same bits. As a result, regmap is difficult to be used on ALC286. We don't request a standard I/O by snd_soc_codec_set_cache_io anymore. Now we have specific write and read functions for ALC286's I/O. A verb command contains three part: NID, VID, and PID. We combine NID and VID as reg address, and PID as reg value. And do corresponding change in rt286_hw_read to get the data properly. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski(a)intel.com> --- The jack detection function is done by Lewandowski Gustaw. So I add Gustaw's Signed-off-by line in this patch The difference between this version and previous version is listed below. * Implement specific read/write functions rather than use regmap. * Use dapm widget to control power bit instead of setting it in _event * Add device ID check Verb command use two bit to set power state(D0, D1, D2, D3). But dapm widget use only one bit to set power up/down. So we use dapm widget to switch each node between D0 and D1 power state. And will set audio group to D3 when bias level = STANDBY. --- include/sound/rt286.h | 19 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt286.c | 1054 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 186 ++++++++ 5 files changed, 1265 insertions(+) create mode 100644 include/sound/rt286.h create mode 100644 sound/soc/codecs/rt286.c create mode 100644 sound/soc/codecs/rt286.h diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 0000000..eb773d1 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool gpio2_en; /*GPIO2 enable*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0e8401..5818391 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -69,6 +69,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RT286 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C @@ -390,6 +391,9 @@ config SND_SOC_PCM512x_SPI select SND_SOC_PCM512x select REGMAP_SPI +config SND_SOC_RT286 + tristate + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3c4d275..5ef8adc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -58,6 +58,7 @@ snd-soc-pcm3008-objs := pcm3008.o snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o +snd-soc-rt286-objs := rt286.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -209,6 +210,7 @@ obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 0000000..a88b3500 --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1054 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <linux/workqueue.h> +#include <sound/rt286.h> +#include <sound/hda_verbs.h> + +#include "rt286.h" + +#define RT286_VENDOR_ID 0x10ec0286 + +#define RT286_INDEX_REG_BASE 0x100 +#define RT286_INDEX_REG_RANGE 0xff +#define RT286_INDEX_REG_MAX (RT286_INDEX_REG_BASE + RT286_INDEX_REG_RANGE) + +struct rt286_priv { + struct snd_soc_codec *codec; + struct rt286_platform_data pdata; + struct i2c_client *i2c; + struct snd_soc_jack *jack; + struct delayed_work jack_detect_work; + int sys_clk; +}; + +static unsigned int rt286_hw_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = rt286->i2c; + struct i2c_msg xfer[2]; + int ret; + __be32 be_reg; + unsigned int index, vid, buf = 0x0; + + /*index registers*/ + if (reg >= RT286_INDEX_REG_BASE && reg < RT286_INDEX_REG_MAX) { + ret = snd_soc_write(codec, + VERB_CMD(AC_VERB_SET_COEF_INDEX, + RT286_VENDOR_REGISTERS, 0), + reg - RT286_INDEX_REG_BASE); + + if (ret < 0) { + dev_err(codec->dev, + "Failed to set private addr: %d\n", ret); + return ret; + } + + reg = VERB_CMD(AC_VERB_GET_PROC_COEF, + RT286_VENDOR_REGISTERS, 0); + } + reg = reg | 0x80000; + vid = (reg >> 8) & 0xfff; + + if (AC_VERB_GET_AMP_GAIN_MUTE == (vid & 0xf00)) { + index = (reg >> 8) & 0xf; + reg = (reg & ~0xf0f) | index; + } + be_reg = cpu_to_be32(reg); + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)&be_reg; + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + return be32_to_cpu(buf); +} + +static int rt286_hw_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = rt286->i2c; + u8 data[4]; + int ret; + + /*index registers*/ + if (reg >= RT286_INDEX_REG_BASE && reg < RT286_INDEX_REG_MAX) { + ret = snd_soc_write(codec, + VERB_CMD(AC_VERB_SET_COEF_INDEX, + RT286_VENDOR_REGISTERS, 0), + reg - RT286_INDEX_REG_BASE); + if (ret < 0) { + dev_err(codec->dev, + "Failed to set private addr: %d\n", ret); + return -EIO; + } + reg = VERB_CMD(AC_VERB_SET_PROC_COEF, + RT286_VENDOR_REGISTERS, 0); + } + + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + /* + * 4 bit VID: reg should be 0 + * 12 bit VID: value should be 0 + * So we use an OR operator to handle it rather than use if condition. + */ + data[2] = ((reg >> 8) & 0xff) | ((value >> 8) & 0xff); + data[3] = value & 0xff; + + ret = i2c_master_send(client, data, 4); + if (ret == 4) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static int rt286_support_power_controls[] = { + RT286_DAC_OUT1, + RT286_DAC_OUT2, + RT286_ADC_IN1, + RT286_ADC_IN2, + RT286_MIC1, + RT286_DMIC1, + RT286_DMIC2, + RT286_SPK_OUT, + RT286_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val, buf; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + *hp = buf & 0x80000000; + if (*hp) { + /* power on HV,VERF */ + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_POWER_CTRL1, 0x1001, 0x0); + /* power LDO1 */ + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_POWER_CTRL2, 0x4, 0x4); + snd_soc_write(codec, RT286_SET_MIC1, 0x24); + val = snd_soc_read(codec, RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL2); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + val = snd_soc_read(codec, + RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL2); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + snd_soc_write(codec, + RT286_SET_MIC1, 0x20); + /* power off HV,VERF */ + snd_soc_update_bits(codec, + RT286_INDEX_REG_BASE + + RT286_POWER_CTRL1, 0x1001, 0x1001); + snd_soc_update_bits(codec, + RT286_INDEX_REG_BASE + + RT286_A_BIAS_CTRL3, 0xc000, 0x0000); + snd_soc_update_bits(codec, + RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL1, 0x0030, 0x0000); + snd_soc_update_bits(codec, + RT286_INDEX_REG_BASE + + RT286_A_BIAS_CTRL2, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + snd_soc_update_bits(codec, + RT286_INDEX_REG_BASE + + RT286_A_BIAS_CTRL3, 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL1, 0x0030, 0x0020); + snd_soc_update_bits(codec, + RT286_INDEX_REG_BASE + + RT286_A_BIAS_CTRL2, 0xc000, 0x8000); + } else { + *mic = false; + } + + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_MISC_CTRL1, + 0x0060, 0x0000); + } else { + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_MISC_CTRL1, + 0x0060, 0x0020); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_A_BIAS_CTRL3, + 0xc000, 0x8000); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL1, + 0x0030, 0x0020); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_A_BIAS_CTRL2, + 0xc000, 0x8000); + + *mic = false; + } + } else { + buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + *hp = buf & 0x80000000; + buf = snd_soc_read(codec, RT286_GET_MIC1_SENSE); + *mic = buf & 0x80000000; + } + + return 0; +} + +static void rt286_jack_detect_work(struct work_struct *work) +{ + struct rt286_priv *rt286 = + container_of(work, struct rt286_priv, jack_detect_work.work); + int status = 0; + + bool hp = false; + bool mic = false; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); +} + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + rt286->jack = jack; + + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, 0, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + return 0; +} +EXPORT_SYMBOL_GPL(rt286_mic_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + + + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN, + RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN, + RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN, + 0, 0x3, 0, mic_vol_tlv), + SOC_DOUBLE_R("Speaker Playback Switch", RT286_SPOL_GAIN, + RT286_SPOR_GAIN, RT286_MUTE_SFT, 1, 1), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT286_F_DAC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", RT286_F_RECMIX_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Mic1 Switch", RT286_REC_MIC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", RT286_REC_I2S_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", RT286_REC_LINE_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Beep Switch", RT286_REC_BEEP_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spo_enable_control = + SOC_DAPM_SINGLE("Switch", RT286_SET_PIN_SPK, + RT286_SET_PIN_SFT, 1, 0); + +static const struct snd_kcontrol_new hpol_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOL_GAIN, + RT286_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hpor_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOR_GAIN, + RT286_MUTE_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "RECMIX", "Dmic" +}; + +static const int rt286_adc_values[] = { + 0, 4, 5, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, RT286_ADC0_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_VALUE_ENUM("ADC 0 source", rt286_adc0_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, RT286_ADC1_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_VALUE_ENUM("ADC 1 source", rt286_adc1_enum); + +static const char * const rt286_dac_src[] = { + "Front", "Surround" +}; +/* HP-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, RT286_HPO_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_spo_enum, RT286_SPK_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +static int rt286_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_HIGH); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_LOW); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0x20); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int nid; + + nid = (w->reg >> 20) & 0xff; + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7080); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", RT286_SET_POWER(RT286_DMIC1), 0, 1, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA("DMIC2", RT286_SET_POWER(RT286_DMIC2), 0, 1, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* Output Mux */ + SND_SOC_DAPM_MUX("SPK Mux", SND_SOC_NOPM, 0, 0, &rt286_spo_mux), + SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt286_hpo_mux), + + SND_SOC_DAPM_SUPPLY("HP Power", RT286_SET_PIN_HPO, + RT286_SET_PIN_SFT, 0, NULL, 0), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", RT286_SET_POWER(RT286_DAC_OUT1), 0, 1, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", RT286_SET_POWER(RT286_DAC_OUT2), 0, 1, + NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0, + &spo_enable_control, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0, + &hpol_enable_control), + SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0, + &hpor_enable_control), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "RECMIX", "RECMIX"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "RECMIX", "RECMIX"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", "Switch", "SPK Mux"}, + {"HPO L", "Switch", "HPO Mux"}, + {"HPO R", "Switch", "HPO Mux"}, + {"HPO L", NULL, "HP Power"}, + {"HPO R", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO L"}, + {"HPO Pin", NULL, "HPO R"}, +}; + +static int rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_format(params)) { + /* bit 6:4 Bits per Sample */ + case SNDRV_PCM_FORMAT_S16_LE: + d_len_code = 0; + val |= (0x1 << 4); + break; + case SNDRV_PCM_FORMAT_S32_LE: + d_len_code = 2; + val |= (0x4 << 4); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + d_len_code = 1; + val |= (0x2 << 4); + break; + case SNDRV_PCM_FORMAT_S24_LE: + d_len_code = 2; + val |= (0x3 << 4); + break; + case SNDRV_PCM_FORMAT_S8: + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x0018, d_len_code << 3); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); + else + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x8000, 0); + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x8000, 0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL2, 0x0100, 0x0); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_PLL_CTRL1, 0x20, 0x20); + } else { + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL2, 0x0100, 0x0100); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_PLL_CTRL, 0x4, 0x4); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_PLL_CTRL1, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL2, 0x40, 0x40); + break; + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL2, 0x40, 0x0); + break; + case 12288000: + case 11289600: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL2, 0x8, 0x0); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_CLK_DIV, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL2, 0x8, 0x8); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_CLK_DIV, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (50 == ratio) + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x1000, 0x1000); + else + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D0); + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D3); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static irqreturn_t rt286_irq(int irq, void *data) +{ + struct rt286_priv *rt286 = data; + bool hp = false; + bool mic = false; + int status = 0; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + /* Clear IRQ */ + snd_soc_update_bits(rt286->codec, RT286_INDEX_REG_BASE + + RT286_IRQ_CTRL, 0x1, 0x1); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt286->i2c->dev, 300); + + return IRQ_HANDLED; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i, ret; + + ret = snd_soc_read(codec, + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID)); + if (ret != RT286_VENDOR_ID) { + dev_err(codec->dev, + "Device with ID register %x is not rt286\n", ret); + return -ENODEV; + } + + snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); + + for (i = 0; i < RT286_POWER_REG_LEN; i++) + snd_soc_write(codec, + RT286_SET_POWER(rt286_support_power_controls[i]), + AC_PWRST_D1); + + if (!rt286->pdata.cbj_en) { + snd_soc_write(codec, RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL2, 0x0000); + snd_soc_write(codec, RT286_INDEX_REG_BASE + + RT286_MIC1_DET_CTRL, 0x0816); + snd_soc_write(codec, RT286_INDEX_REG_BASE + + RT286_MISC_CTRL1, 0x0000); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL1, 0xf000, 0xb000); + } else { + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL1, 0xf000, 0x5000); + } + + mdelay(10); + + if (!rt286->pdata.gpio2_en) + snd_soc_write(codec, RT286_SET_DMIC2_DEFAULT, 0x4000); + else + snd_soc_write(codec, RT286_SET_DMIC2_DEFAULT, 0); + + mdelay(10); + + /*Power down LDO2*/ + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_POWER_CTRL2, 0x8, 0x0); + + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + rt286->codec = codec; + + if (rt286->i2c->irq) { + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_IRQ_CTRL, 0x2, 0x2); + + INIT_DELAYED_WORK(&rt286->jack_detect_work, + rt286_jack_detect_work); + schedule_delayed_work(&rt286->jack_detect_work, + msecs_to_jiffies(1250)); + + ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); + if (ret != 0) { + dev_err(codec->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int rt286_remove(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + cancel_delayed_work_sync(&rt286->jack_detect_work); + + return 0; +} + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_bclk_ratio = rt286_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "rt286-aif2", + .id = RT286_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .remove = rt286_remove, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .write = rt286_hw_write, + .read = rt286_hw_read, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt286", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static const struct acpi_device_id rt286_acpi_match[] = { + { "INT343A", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + int ret; + + rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->i2c = i2c; + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + + return ret; +} + +static int rt286_i2c_remove(struct i2c_client *i2c) +{ + struct rt286_priv *rt286 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt286); + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + + +struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt286", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .remove = rt286_i2c_remove, + .id_table = rt286_i2c_id, +}; + +module_i2c_driver(rt286_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); + +/* + * ALC286 is a dual mode codec, which can run as HD-A or I2S mode. + * It is controlled by HD-A verb commands via I2C protocol. + * The following is the I/O difference between ALC286 and general I2S codecs. + * 1. A HD-A verb command contains three parts, NID, VID, and PID. + * And an I2S command contains only two parts: address and data. + * 2. Not only the register address is written, but the read command also + includes the entire write command. + * 3. rt286 uses different registers for read and write the same bits. + * As a result, regmap is difficult to be used on ALC286. + * We don't request a standard I/O by snd_soc_codec_set_cache_io anymore. + * Now we have specific write and read functions for ALC286's I/O. + * A verb command contains three part: NID, VID, and PID. + * We combine NID and VID as reg address, and PID as reg value. + * And do corresponding change in rt286_hw_read to get the data properly. + */ diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 0000000..eaa093f --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,186 @@ +/* + * rt286.h -- RT286 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT286_H__ +#define __RT286_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT286_AUDIO_FUNCTION_GROUP 0x01 +#define RT286_DAC_OUT1 0x02 +#define RT286_DAC_OUT2 0x03 +#define RT286_ADC_IN1 0x09 +#define RT286_ADC_IN2 0x08 +#define RT286_MIXER_IN 0x0b +#define RT286_MIXER_OUT1 0x0c +#define RT286_MIXER_OUT2 0x0d +#define RT286_DMIC1 0x12 +#define RT286_DMIC2 0x13 +#define RT286_SPK_OUT 0x14 +#define RT286_MIC1 0x18 +#define RT286_LINE1 0x1a +#define RT286_BEEP 0x1d +#define RT286_SPDIF 0x1e +#define RT286_VENDOR_REGISTERS 0x20 +#define RT286_HP_OUT 0x21 +#define RT286_MIXER_IN1 0x22 +#define RT286_MIXER_IN2 0x23 + +#define RT286_SET_PIN_SFT 6 +#define RT286_SET_PIN_ENABLE 0x40 +#define RT286_SET_PIN_DISABLE 0 +#define RT286_SET_EAPD_HIGH 0x2 +#define RT286_SET_EAPD_LOW 0 + +#define RT286_MUTE_SFT 7 + +/* Verb commands */ +#define RT286_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM) +#define RT286_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0) +#define RT286_SET_AUDIO_POWER RT286_SET_POWER(RT286_AUDIO_FUNCTION_GROUP) +#define RT286_SET_HPO_POWER RT286_SET_POWER(RT286_HP_OUT) +#define RT286_SET_SPK_POWER RT286_SET_POWER(RT286_SPK_OUT) +#define RT286_SET_DMIC1_POWER RT286_SET_POWER(RT286_DMIC1) +#define RT286_SPK_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_SPK_OUT, 0) +#define RT286_HPO_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_HP_OUT, 0) +#define RT286_ADC0_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN1, 0) +#define RT286_ADC1_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN2, 0) +#define RT286_SET_MIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0) +#define RT286_SET_PIN_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0) +#define RT286_SET_PIN_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0) +#define RT286_SET_PIN_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0) +#define RT286_SPK_EAPD\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0) +#define RT286_SET_AMP_GAIN_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0) +#define RT286_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0) +#define RT286_GET_MIC1_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0) +#define RT286_SET_DMIC2_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0) +#define RT286_DACL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0xa000) +#define RT286_DACR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0x9000) +#define RT286_ADCL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x6000) +#define RT286_ADCR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x5000) +#define RT286_MIC_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIC1, 0x7000) +#define RT286_SPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0xa000) +#define RT286_SPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0x9000) +#define RT286_HPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000) +#define RT286_HPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000) +#define RT286_F_DAC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7000) +#define RT286_F_RECMIX_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7100) +#define RT286_REC_MIC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7000) +#define RT286_REC_I2S_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7100) +#define RT286_REC_LINE_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7200) +#define RT286_REC_BEEP_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7300) +#define RT286_DAC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_DAC_OUT1, 0) +#define RT286_ADC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_ADC_IN1, 0) + + +/* Index registers */ +#define RT286_A_BIAS_CTRL1 0x01 +#define RT286_A_BIAS_CTRL2 0x02 +#define RT286_POWER_CTRL1 0x03 +#define RT286_A_BIAS_CTRL3 0x04 +#define RT286_POWER_CTRL2 0x08 +#define RT286_I2S_CTRL1 0x09 +#define RT286_I2S_CTRL2 0x0a +#define RT286_CLK_DIV 0x0b +#define RT286_POWER_CTRL3 0x0f +#define RT286_MIC1_DET_CTRL 0x19 +#define RT286_MISC_CTRL1 0x20 +#define RT286_IRQ_CTRL 0x33 +#define RT286_PLL_CTRL1 0x49 +#define RT286_CBJ_CTRL1 0x4f +#define RT286_CBJ_CTRL2 0x50 +#define RT286_PLL_CTRL 0x63 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_M_SPK_MUX_SFT 14 +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_M_HP_MUX_SFT 14 +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIF2, + RT286_AIFS, +}; + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +#endif /* __RT286_H__ */ + -- 1.8.1.1.439.g50a6b54

5 19

[alsa-devel] [PATCH] ASoC: arizona: Implement TDM support for Arizona devices
by Charles Keepax 04 Jun '14

04 Jun '14

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/linux/mfd/arizona/core.h | 3 + sound/soc/codecs/arizona.c | 81 +++++++++++++++++++++++++++++++++++--- 2 files changed, 78 insertions(+), 6 deletions(-) diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index 5cf8b91..11783b5 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -110,6 +110,9 @@ struct arizona { int clk32k_ref; struct snd_soc_dapm_context *dapm; + + int tdm_width[ARIZONA_MAX_AIF]; + int tdm_slots[ARIZONA_MAX_AIF]; }; int arizona_clk32k_enable(struct arizona *arizona); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 29e198f..d98bd77 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1185,7 +1185,10 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, int base = dai->driver->base; const int *rates; int i, ret, val; + int channels = params_channels(params); int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1]; + int tdm_width = arizona->tdm_width[dai->id - 1]; + int tdm_slots = arizona->tdm_slots[dai->id - 1]; int bclk, lrclk, wl, frame, bclk_target; if (params_rate(params) % 8000) @@ -1193,18 +1196,27 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, else rates = &arizona_48k_bclk_rates[0]; - bclk_target = snd_soc_params_to_bclk(params); - if (chan_limit && chan_limit < params_channels(params)) { + if (tdm_slots) { + arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", + tdm_slots, tdm_width); + bclk_target = tdm_slots * tdm_width * params_rate(params); + channels = tdm_slots; + } else { + bclk_target = snd_soc_params_to_bclk(params); + } + + if (chan_limit && chan_limit < channels) { arizona_aif_dbg(dai, "Limiting to %d channels\n", chan_limit); - bclk_target /= params_channels(params); + bclk_target /= channels; bclk_target *= chan_limit; } - /* Force stereo for I2S mode */ + /* Force multiple of 2 channels for I2S mode */ val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); - if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) { + if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) { arizona_aif_dbg(dai, "Forcing stereo mode\n"); - bclk_target *= 2; + bclk_target /= channels; + bclk_target *= channels + 1; } for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { @@ -1228,6 +1240,9 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, wl = snd_pcm_format_width(params_format(params)); frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + if (tdm_width && wl != tdm_width) + arizona_aif_warn(dai, "Word length not equal to TDM width\n"); + ret = arizona_hw_params_rate(substream, params, dai); if (ret != 0) return ret; @@ -1324,9 +1339,63 @@ static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) ARIZONA_AIF1_TRI, reg); } +static void arizona_set_channels_to_mask(struct snd_soc_dai *dai, + unsigned int base, + int channels, unsigned int mask) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int slot, i; + + for (i = 0; i < channels; ++i) { + slot = ffs(mask) - 1; + if (slot < 0) + return; + + regmap_write(arizona->regmap, base + i, slot); + + mask &= ~(1 << slot); + } + + if (mask) + arizona_aif_warn(dai, "Too many channels in TDM mask\n"); +} + +static int arizona_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int base = dai->driver->base; + int rx_max_chan = dai->driver->playback.channels_max; + int tx_max_chan = dai->driver->capture.channels_max; + + /* Only support TDM for the physical AIFs */ + if (dai->id > ARIZONA_MAX_AIF) + return -ENOTSUPP; + + if (slots == 0) { + tx_mask = (1 << tx_max_chan) - 1; + rx_mask = (1 << rx_max_chan) - 1; + } + + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_3, + tx_max_chan, tx_mask); + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_11, + rx_max_chan, rx_mask); + + arizona->tdm_width[dai->id - 1] = slot_width; + arizona->tdm_slots[dai->id - 1] = slots; + + return 0; +} + const struct snd_soc_dai_ops arizona_dai_ops = { .startup = arizona_startup, .set_fmt = arizona_set_fmt, + .set_tdm_slot = arizona_set_tdm_slot, .hw_params = arizona_hw_params, .set_sysclk = arizona_dai_set_sysclk, .set_tristate = arizona_set_tristate, -- 1.7.2.5

2 2

[alsa-devel] PROBLEM: [Lenovo ThinkPad X61s] Speaker volume mutes itself in random intervals after upgrading from 3.11 to 3.15.
by Alex Fedora 04 Jun '14

04 Jun '14

Hi, Please see problem description below. I've captured alsa info output for the latest kernel. Kernel just before the regression. And info from the kernel with the offending commit. But I had to calculate diff from first one to fit to 100K limit. If you want, I can send all 3 files as an attachments. Also the latest version of the kernel was took from http://kernel.ubuntu.com/~kernel-ppa/mainline/daily/current/linux-image-3.1… . Thank you, Alex [1.] One line summary of the problem: [Lenovo ThinkPad X61s] Speaker volume mutes itself in random intervals after upgrade to latest kernel [2.] Full description of the problem/report: Speaker mutes itself in random intervals. Headphones are working fine at that moment. I can see what volume of the Speaker drops to 0% and then back to 100% using alsamixer. It seems like Internal Mic Boost also jumps from 100% to 0% and back. It is regression from 5ccc618fee67f0f0b2122dd4b32a02fd2b6a1569 (ALSA: hda - Remove static quirks for AD1884/1984 & variants). I've tested previous commit aa95d61b43e0fcb0b2ce68e5efa37174fd9e5cd3 (ALSA: hda - Remove static quirks for AD1882) and audio works as expected. New kernel shows two controls "Headphone Playback Volume" and "Speaker Playback Volume". But old one shows single control "PCM Playback Volume". "Speaker Playback Volume" is the one which gets muted randomly. [3.] Keywords (i.e., modules, networking, kernel): [4.] Kernel version (from /proc/version): Linux version 3.15.0-999-generic (apw@gomeisa) (gcc version 4.6.3 (Ubuntu/Linaro 4.6.3-1ubuntu5) ) #201405310205 SMP Sat May 31 06:06:08 UTC 2014 [5.] Output of Oops.. message (if applicable) with symbolic information resolved (see Documentation/oops-tracing.txt) [6.] A small shell script or example program which triggers the problem (if possible) Tested manually. [7.] Environment Description: Ubuntu 14.04 LTS Release: 14.04 [7.1.] Software (add the output of the ver_linux script here) If some fields are empty or look unusual you may have an old version. Compare to the current minimal requirements in Documentation/Changes. Linux x61s 3.15.0-999-generic #201405310205 SMP Sat May 31 06:06:08 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux Gnu C 4.8 Gnu make 3.81 binutils 2.24 util-linux 2.20.1 mount support module-init-tools 15 e2fsprogs 1.42.9 pcmciautils 018 Linux C Library 2.19 Dynamic linker (ldd) 2.19 Procps 3.3.9 Net-tools 1.60 Kbd 1.15.5 Sh-utils 8.21 wireless-tools 30 Modules Loaded ctr ccm nls_iso8859_1 ipt_MASQUERADE iptable_nat nf_nat_ipv4 nf_nat nf_conntrack_ipv4 nf_defrag_ipv4 xt_conntrack nf_conntrack ipt_REJECT xt_CHECKSUM iptable_mangle xt_tcpudp bridge stp llc ip6table_filter ip6_tables iptable_filter ip_tables ebtable_nat ebtables x_tables rfcomm bnep bluetooth 6lowpan_iphc binfmt_misc snd_hda_codec_analog snd_hda_codec_generic snd_hda_intel snd_hda_controller snd_hda_codec snd_hwdep thinkpad_acpi dm_multipath coretemp snd_pcm nvram kvm_intel scsi_dh snd_seq_midi arc4 kvm snd_seq_midi_event iwl4965 iwlegacy mac80211 snd_rawmidi snd_seq cfg80211 snd_seq_device pcmcia snd_timer microcode snd serio_raw yenta_socket soundcore mac_hid pcmcia_rsrc ppdev lpc_ich pcmcia_core parport_pc lp parport xts gf128mul dm_crypt mmc_block ahci sdhci_pci psmouse libahci sdhci i915 video i2c_algo_bit e1000e drm_kms_helper drm ptp pps_core [7.2.] Processor information (from /proc/cpuinfo): processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 15 model name : Intel(R) Core(TM)2 Duo CPU L7500 @ 1.60GHz stepping : 11 microcode : 0xba cpu MHz : 800.000 cache size : 4096 KB physical id : 0 siblings : 2 core id : 0 cpu cores : 2 apicid : 0 initial apicid : 0 fpu : yes fpu_exception : yes cpuid level : 10 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe syscall nx lm constant_tsc arch_perfmon pebs bts rep_good nopl aperfmperf pni dtes64 monitor ds_cpl vmx est tm2 ssse3 cx16 xtpr pdcm lahf_lm ida dtherm tpr_shadow vnmi flexpriority bogomips : 3191.89 clflush size : 64 cache_alignment : 64 address sizes : 36 bits physical, 48 bits virtual power management: processor : 1 vendor_id : GenuineIntel cpu family : 6 model : 15 model name : Intel(R) Core(TM)2 Duo CPU L7500 @ 1.60GHz stepping : 11 microcode : 0xba cpu MHz : 800.000 cache size : 4096 KB physical id : 0 siblings : 2 core id : 1 cpu cores : 2 apicid : 1 initial apicid : 1 fpu : yes fpu_exception : yes cpuid level : 10 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe syscall nx lm constant_tsc arch_perfmon pebs bts rep_good nopl aperfmperf pni dtes64 monitor ds_cpl vmx est tm2 ssse3 cx16 xtpr pdcm lahf_lm ida dtherm tpr_shadow vnmi flexpriority bogomips : 3191.89 clflush size : 64 cache_alignment : 64 address sizes : 36 bits physical, 48 bits virtual power management: [7.3.] Module information (from /proc/modules): ctr 13193 2 - Live 0x0000000000000000 ccm 17856 2 - Live 0x0000000000000000 nls_iso8859_1 12713 1 - Live 0x0000000000000000 ipt_MASQUERADE 12880 3 - Live 0x0000000000000000 iptable_nat 13151 1 - Live 0x0000000000000000 nf_nat_ipv4 13316 1 iptable_nat, Live 0x0000000000000000 nf_nat 26147 3 ipt_MASQUERADE,iptable_nat,nf_nat_ipv4, Live 0x0000000000000000 nf_conntrack_ipv4 14857 2 - Live 0x0000000000000000 nf_defrag_ipv4 12758 1 nf_conntrack_ipv4, Live 0x0000000000000000 xt_conntrack 12760 1 - Live 0x0000000000000000 nf_conntrack 105687 6 ipt_MASQUERADE,iptable_nat,nf_nat_ipv4,nf_nat,nf_conntrack_ipv4,xt_conntrack, Live 0x0000000000000000 ipt_REJECT 12579 2 - Live 0x0000000000000000 xt_CHECKSUM 12549 1 - Live 0x0000000000000000 iptable_mangle 12734 1 - Live 0x0000000000000000 xt_tcpudp 12924 6 - Live 0x0000000000000000 bridge 116335 0 - Live 0x0000000000000000 stp 12976 1 bridge, Live 0x0000000000000000 llc 14441 2 bridge,stp, Live 0x0000000000000000 ip6table_filter 12815 0 - Live 0x0000000000000000 ip6_tables 27504 1 ip6table_filter, Live 0x0000000000000000 iptable_filter 12810 1 - Live 0x0000000000000000 ip_tables 27718 3 iptable_nat,iptable_mangle,iptable_filter, Live 0x0000000000000000 ebtable_nat 12807 0 - Live 0x0000000000000000 ebtables 35305 1 ebtable_nat, Live 0x0000000000000000 x_tables 34194 11 ipt_MASQUERADE,xt_conntrack,ipt_REJECT,xt_CHECKSUM,iptable_mangle,xt_tcpudp,ip6table_filter,ip6_tables,iptable_filter,ip_tables,ebtables, Live 0x0000000000000000 rfcomm 75078 0 - Live 0x0000000000000000 bnep 19884 2 - Live 0x0000000000000000 bluetooth 461775 10 rfcomm,bnep, Live 0x0000000000000000 6lowpan_iphc 18968 1 bluetooth, Live 0x0000000000000000 binfmt_misc 17508 1 - Live 0x0000000000000000 snd_hda_codec_analog 15049 1 - Live 0x0000000000000000 snd_hda_codec_generic 70087 1 snd_hda_codec_analog, Live 0x0000000000000000 snd_hda_intel 30608 6 - Live 0x0000000000000000 snd_hda_controller 35518 1 snd_hda_intel, Live 0x0000000000000000 snd_hda_codec 144671 4 snd_hda_codec_analog,snd_hda_codec_generic,snd_hda_intel,snd_hda_controller, Live 0x0000000000000000 snd_hwdep 13613 1 snd_hda_codec, Live 0x0000000000000000 thinkpad_acpi 82198 0 - Live 0x0000000000000000 dm_multipath 23188 0 - Live 0x0000000000000000 coretemp 13638 0 - Live 0x0000000000000000 snd_pcm 113863 3 snd_hda_intel,snd_hda_controller,snd_hda_codec, Live 0x0000000000000000 nvram 14462 1 thinkpad_acpi, Live 0x0000000000000000 kvm_intel 148919 0 - Live 0x0000000000000000 scsi_dh 14873 1 dm_multipath, Live 0x0000000000000000 snd_seq_midi 13564 0 - Live 0x0000000000000000 arc4 12573 2 - Live 0x0000000000000000 kvm 463855 1 kvm_intel, Live 0x0000000000000000 snd_seq_midi_event 14899 1 snd_seq_midi, Live 0x0000000000000000 iwl4965 118803 0 - Live 0x0000000000000000 iwlegacy 105101 1 iwl4965, Live 0x0000000000000000 mac80211 663883 2 iwl4965,iwlegacy, Live 0x0000000000000000 snd_rawmidi 30865 1 snd_seq_midi, Live 0x0000000000000000 snd_seq 67636 2 snd_seq_midi,snd_seq_midi_event, Live 0x0000000000000000 cfg80211 515506 3 iwl4965,iwlegacy,mac80211, Live 0x0000000000000000 snd_seq_device 14497 3 snd_seq_midi,snd_rawmidi,snd_seq, Live 0x0000000000000000 pcmcia 62760 0 - Live 0x0000000000000000 snd_timer 30118 2 snd_pcm,snd_seq, Live 0x0000000000000000 microcode 24391 0 - Live 0x0000000000000000 snd 74195 23 snd_hda_codec_analog,snd_hda_codec_generic,snd_hda_intel,snd_hda_codec,snd_hwdep,thinkpad_acpi,snd_pcm,snd_rawmidi,snd_seq,snd_seq_device,snd_timer, Live 0x0000000000000000 serio_raw 13483 0 - Live 0x0000000000000000 yenta_socket 45447 0 - Live 0x0000000000000000 soundcore 12680 2 snd_hda_codec,snd, Live 0x0000000000000000 mac_hid 13275 0 - Live 0x0000000000000000 pcmcia_rsrc 18591 1 yenta_socket, Live 0x0000000000000000 ppdev 17711 0 - Live 0x0000000000000000 lpc_ich 21176 0 - Live 0x0000000000000000 pcmcia_core 23715 3 pcmcia,yenta_socket,pcmcia_rsrc, Live 0x0000000000000000 parport_pc 32906 0 - Live 0x0000000000000000 lp 17799 0 - Live 0x0000000000000000 parport 42481 3 ppdev,parport_pc,lp, Live 0x0000000000000000 xts 12951 1 - Live 0x0000000000000000 gf128mul 14951 1 xts, Live 0x0000000000000000 dm_crypt 23456 1 - Live 0x0000000000000000 mmc_block 36596 2 - Live 0x0000000000000000 ahci 30167 3 - Live 0x0000000000000000 sdhci_pci 23347 0 - Live 0x0000000000000000 psmouse 113095 0 - Live 0x0000000000000000 libahci 32191 1 ahci, Live 0x0000000000000000 sdhci 43409 1 sdhci_pci, Live 0x0000000000000000 i915 873516 3 - Live 0x0000000000000000 video 19932 1 i915, Live 0x0000000000000000 i2c_algo_bit 13564 1 i915, Live 0x0000000000000000 e1000e 262482 0 - Live 0x0000000000000000 drm_kms_helper 59729 1 i915, Live 0x0000000000000000 drm 310655 5 i915,drm_kms_helper, Live 0x0000000000000000 ptp 19534 1 e1000e, Live 0x0000000000000000 pps_core 19381 1 ptp, Live 0x0000000000000000 [7.4.] Loaded driver and hardware information (/proc/ioports, /proc/iomem) ioports: 0000-0cf7 : PCI Bus 0000:00 0000-001f : dma1 0020-0021 : pic1 0040-0043 : timer0 0050-0053 : timer1 0060-0060 : keyboard 0062-0062 : EC data 0064-0064 : keyboard 0066-0066 : EC cmd 0070-0071 : rtc0 0080-008f : dma page reg 00a0-00a1 : pic2 00c0-00df : dma2 00f0-00ff : fpu 03c0-03df : vesafb 0800-080f : pnp 00:01 0cf8-0cff : PCI conf1 0d00-ffff : PCI Bus 0000:00 1000-1003 : ACPI PM1a_EVT_BLK 1004-1005 : ACPI PM1a_CNT_BLK 1008-100b : ACPI PM_TMR 1010-1015 : ACPI CPU throttle 1020-1020 : ACPI PM2_CNT_BLK 1028-102f : ACPI GPE0_BLK 1030-1033 : iTCO_wdt 1060-107f : iTCO_wdt 1180-11bf : 0000:00:1f.0 1180-11bf : pnp 00:01 15e0-15ef : pnp 00:01 164e-164f : pnp 00:01 1800-1807 : 0000:00:02.0 1838-183b : 0000:00:1f.2 1838-183b : ahci 183c-183f : 0000:00:1f.2 183c-183f : ahci 1840-185f : 0000:00:19.0 1860-187f : 0000:00:1a.0 1860-187f : uhci_hcd 1880-189f : 0000:00:1a.1 1880-189f : uhci_hcd 18a0-18bf : 0000:00:1d.0 18a0-18bf : uhci_hcd 18c0-18df : 0000:00:1d.1 18c0-18df : uhci_hcd 18e0-18ff : 0000:00:1f.2 18e0-18ff : ahci 1c00-1c07 : 0000:00:1f.2 1c00-1c07 : ahci 1c08-1c0f : 0000:00:1f.2 1c08-1c0f : ahci 1c20-1c3f : 0000:00:1f.3 2000-2fff : PCI Bus 0000:02 3000-3fff : PCI Bus 0000:03 4000-7fff : PCI Bus 0000:05 4000-40ff : PCI CardBus 0000:06 4400-44ff : PCI CardBus 0000:06 iomem: 00000000-00000fff : reserved 00001000-0009d7ff : System RAM 0009d800-0009ffff : reserved 000a0000-000bffff : PCI Bus 0000:00 000c0000-000c7fff : Video ROM 000c8000-000cbfff : pnp 00:00 000cc000-000cffff : pnp 00:00 000d0000-000d3fff : PCI Bus 0000:00 000d4000-000d7fff : PCI Bus 0000:00 000d8000-000dbfff : PCI Bus 0000:00 000dc000-000dffff : PCI Bus 0000:00 000e0000-000fffff : reserved 000f0000-000fffff : System ROM 00100000-bf6affff : System RAM 01000000-0177cf58 : Kernel code 0177cf59-01d20f7f : Kernel data 01e7d000-01fe2fff : Kernel bss 2c000000-33ffffff : Crash kernel bf6b0000-bf6cbfff : ACPI Tables bf6cc000-bf6fffff : ACPI Non-volatile Storage bf700000-bfffffff : reserved bf800000-bfffffff : Graphics Stolen Memory c0000000-febfffff : PCI Bus 0000:00 c0000000-c3ffffff : PCI CardBus 0000:06 c4000000-c4000fff : Intel Flush Page d4000000-d7efffff : PCI Bus 0000:05 d7efec00-d7efecff : 0000:05:00.2 d7efec00-d7efecff : mmc0 d7eff000-d7efffff : 0000:05:00.0 d7eff000-d7efffff : yenta_socket d8000000-dbffffff : PCI Bus 0000:05 d8000000-dbffffff : PCI CardBus 0000:06 e0000000-efffffff : 0000:00:02.0 f0000000-f3ffffff : PCI MMCONFIG 0000 [bus 00-3f] f0000000-f3ffffff : reserved f0000000-f3ffffff : pnp 00:01 f4000000-f5ffffff : PCI Bus 0000:02 f6000000-f7ffffff : PCI Bus 0000:03 f7f00000-f7f01fff : 0000:03:00.0 f7f00000-f7f01fff : iwl4965 f8000000-f80fffff : 0000:00:02.0 f8100000-f81fffff : 0000:00:02.1 f8200000-f821ffff : 0000:00:19.0 f8200000-f821ffff : e1000e f8220000-f8223fff : 0000:00:1b.0 f8220000-f8223fff : ICH HD audio f8225000-f8225fff : 0000:00:19.0 f8225000-f8225fff : e1000e f8426000-f84267ff : 0000:00:1f.2 f8426000-f84267ff : ahci f8426c00-f8426fff : 0000:00:1a.7 f8426c00-f8426fff : ehci_hcd f8427000-f84273ff : 0000:00:1d.7 f8427000-f84273ff : ehci_hcd f8427400-f84274ff : 0000:00:1f.3 f8500000-f85fffff : PCI Bus 0000:02 f8600000-f86fffff : PCI Bus 0000:03 fec00000-fec0ffff : reserved fec00000-fec003ff : IOAPIC 0 fed00000-fed003ff : HPET 0 fed00000-fed003ff : reserved fed14000-fed19fff : reserved fed14000-fed17fff : pnp 00:01 fed18000-fed18fff : pnp 00:01 fed19000-fed19fff : pnp 00:01 fed1c000-fed8ffff : reserved fed1c000-fed1ffff : pnp 00:01 fed1f410-fed1f414 : iTCO_wdt fed40000-fed4bfff : PCI Bus 0000:00 fed45000-fed4bfff : pnp 00:01 fee00000-fee00fff : Local APIC fee00000-fee00fff : reserved ff000000-ffffffff : reserved 100000000-13bffffff : System RAM [7.5.] PCI information ('lspci -vvv' as root) 00:00.0 Host bridge: Intel Corporation Mobile PM965/GM965/GL960 Memory Controller Hub (rev 0c) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort+ >SERR- <PERR- INTx- Latency: 0 Capabilities: [e0] Vendor Specific Information: Len=0a <?> Kernel driver in use: agpgart-intel 00:02.0 VGA compatible controller: Intel Corporation Mobile GM965/GL960 Integrated Graphics Controller (primary) (rev 0c) (prog-if 00 [VGA controller]) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 43 Region 0: Memory at f8000000 (64-bit, non-prefetchable) [size=1M] Region 2: Memory at e0000000 (64-bit, prefetchable) [size=256M] Region 4: I/O ports at 1800 [size=8] Expansion ROM at <unassigned> [disabled] Capabilities: [90] MSI: Enable+ Count=1/1 Maskable- 64bit- Address: fee0300c Data: 41e1 Capabilities: [d0] Power Management version 3 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Bridge: PM- B3+ Kernel driver in use: i915 00:02.1 Display controller: Intel Corporation Mobile GM965/GL960 Integrated Graphics Controller (secondary) (rev 0c) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Region 0: Memory at f8100000 (64-bit, non-prefetchable) [size=1M] Capabilities: [d0] Power Management version 3 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Bridge: PM- B3+ 00:19.0 Ethernet controller: Intel Corporation 82566MM Gigabit Network Connection (rev 03) Subsystem: Lenovo Device 20de Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 42 Region 0: Memory at f8200000 (32-bit, non-prefetchable) [size=128K] Region 1: Memory at f8225000 (32-bit, non-prefetchable) [size=4K] Region 2: I/O ports at 1840 [size=32] Capabilities: [c8] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=1 PME- Capabilities: [d0] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0300c Data: 4172 Kernel driver in use: e1000e 00:1a.0 USB controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #4 (rev 03) (prog-if 00 [UHCI]) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 20 Region 4: I/O ports at 1860 [size=32] Kernel driver in use: uhci_hcd 00:1a.1 USB controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #5 (rev 03) (prog-if 00 [UHCI]) Subsystem: Lenovo Thinkpad T61/R61 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 21 Region 4: I/O ports at 1880 [size=32] Kernel driver in use: uhci_hcd 00:1a.7 USB controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #2 (rev 03) (prog-if 20 [EHCI]) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin C routed to IRQ 22 Region 0: Memory at f8426c00 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: ehci-pci 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin B routed to IRQ 46 Region 0: Memory at f8220000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0300c Data: 4152 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- Capabilities: [100 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed- WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01 Status: NegoPending- InProgress- VC1: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=1 ArbSelect=Fixed TC/VC=80 Status: NegoPending- InProgress- Capabilities: [130 v1] Root Complex Link Desc: PortNumber=0f ComponentID=02 EltType=Config Link0: Desc: TargetPort=00 TargetComponent=02 AssocRCRB- LinkType=MemMapped LinkValid+ Addr: 00000000fed1c000 Kernel driver in use: snd_hda_intel 00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 1 (rev 03) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 I/O behind bridge: 00002000-00002fff Memory behind bridge: f4000000-f5ffffff Prefetchable memory behind bridge: 00000000f8500000-00000000f85fffff Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA+ VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [40] Express (v1) Root Port (Slot+), MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #1, Speed 2.5GT/s, Width x1, ASPM L0s L1, Exit Latency L0s <1us, L1 <4us ClockPM- Surprise- LLActRep+ BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x0, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- SltCap: AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug+ Surprise+ Slot #2, PowerLimit 6.500W; Interlock- NoCompl- SltCtl: Enable: AttnBtn- PwrFlt- MRL- PresDet+ CmdCplt+ HPIrq+ LinkChg+ Control: AttnInd Unknown, PwrInd Unknown, Power- Interlock- SltSta: Status: AttnBtn- PowerFlt- MRL- CmdCplt- PresDet- Interlock- Changed: MRL- PresDet- LinkState- RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- PMEIntEna+ CRSVisible- RootCap: CRSVisible- RootSta: PME ReqID 0000, PMEStatus- PMEPending- Capabilities: [80] MSI: Enable+ Count=1/1 Maskable- 64bit- Address: fee0300c Data: 41a1 Capabilities: [90] Subsystem: Lenovo ThinkPad T61/R61 Capabilities: [a0] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [100 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed+ WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed+ WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=ff Status: NegoPending- InProgress- Capabilities: [180 v1] Root Complex Link Desc: PortNumber=01 ComponentID=02 EltType=Config Link0: Desc: TargetPort=00 TargetComponent=02 AssocRCRB- LinkType=MemMapped LinkValid+ Addr: 00000000fed1c001 Kernel driver in use: pcieport 00:1c.1 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 2 (rev 03) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Bus: primary=00, secondary=03, subordinate=03, sec-latency=0 I/O behind bridge: 00003000-00003fff Memory behind bridge: f6000000-f7ffffff Prefetchable memory behind bridge: 00000000f8600000-00000000f86fffff Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort+ <SERR+ <PERR- BridgeCtl: Parity- SERR- NoISA+ VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [40] Express (v1) Root Port (Slot+), MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #2, Speed 2.5GT/s, Width x1, ASPM L0s L1, Exit Latency L0s <256ns, L1 <4us ClockPM- Surprise- LLActRep+ BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive+ BWMgmt- ABWMgmt- SltCap: AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug+ Surprise+ Slot #3, PowerLimit 6.500W; Interlock- NoCompl- SltCtl: Enable: AttnBtn- PwrFlt- MRL- PresDet+ CmdCplt+ HPIrq+ LinkChg+ Control: AttnInd Unknown, PwrInd Unknown, Power- Interlock- SltSta: Status: AttnBtn- PowerFlt- MRL- CmdCplt- PresDet+ Interlock- Changed: MRL- PresDet- LinkState+ RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- PMEIntEna+ CRSVisible- RootCap: CRSVisible- RootSta: PME ReqID 0000, PMEStatus- PMEPending- Capabilities: [80] MSI: Enable+ Count=1/1 Maskable- 64bit- Address: fee0300c Data: 41b1 Capabilities: [90] Subsystem: Lenovo ThinkPad T61/R61 Capabilities: [a0] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [100 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed+ WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed+ WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01 Status: NegoPending- InProgress- Capabilities: [180 v1] Root Complex Link Desc: PortNumber=02 ComponentID=02 EltType=Config Link0: Desc: TargetPort=00 TargetComponent=02 AssocRCRB- LinkType=MemMapped LinkValid+ Addr: 00000000fed1c001 Kernel driver in use: pcieport 00:1d.0 USB controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #1 (rev 03) (prog-if 00 [UHCI]) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 16 Region 4: I/O ports at 18a0 [size=32] Kernel driver in use: uhci_hcd 00:1d.1 USB controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #2 (rev 03) (prog-if 00 [UHCI]) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 17 Region 4: I/O ports at 18c0 [size=32] Kernel driver in use: uhci_hcd 00:1d.7 USB controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #1 (rev 03) (prog-if 20 [EHCI]) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin D routed to IRQ 19 Region 0: Memory at f8427000 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [58] Debug port: BAR=1 offset=00a0 Kernel driver in use: ehci-pci 00:1e.0 PCI bridge: Intel Corporation 82801 Mobile PCI Bridge (rev f3) (prog-if 01 [Subtractive decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Bus: primary=00, secondary=05, subordinate=08, sec-latency=32 I/O behind bridge: 00004000-00007fff Memory behind bridge: d4000000-d7efffff Prefetchable memory behind bridge: 00000000d8000000-00000000dbffffff Secondary status: 66MHz- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA- VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [50] Subsystem: Lenovo ThinkPad T61/R61 00:1f.0 ISA bridge: Intel Corporation 82801HEM (ICH8M-E) LPC Interface Controller (rev 03) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Capabilities: [e0] Vendor Specific Information: Len=0c <?> Kernel driver in use: lpc_ich 00:1f.2 SATA controller: Intel Corporation 82801HM/HEM (ICH8M/ICH8M-E) SATA Controller [AHCI mode] (rev 03) (prog-if 01 [AHCI 1.0]) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 44 Region 0: I/O ports at 1c08 [size=8] Region 1: I/O ports at 183c [size=4] Region 2: I/O ports at 1c00 [size=8] Region 3: I/O ports at 1838 [size=4] Region 4: I/O ports at 18e0 [size=32] Region 5: Memory at f8426000 (32-bit, non-prefetchable) [size=2K] Capabilities: [80] MSI: Enable+ Count=1/4 Maskable- 64bit- Address: fee0300c Data: 4122 Capabilities: [70] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold-) Status: D0 NoSoftRst+ PME-Enable- DSel=0 DScale=0 PME- Capabilities: [a8] SATA HBA v1.0 BAR4 Offset=00000004 Kernel driver in use: ahci 00:1f.3 SMBus: Intel Corporation 82801H (ICH8 Family) SMBus Controller (rev 03) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin A routed to IRQ 11 Region 0: Memory at f8427400 (32-bit, non-prefetchable) [size=256] Region 4: I/O ports at 1c20 [size=32] 03:00.0 Network controller: Intel Corporation PRO/Wireless 4965 AG or AGN [Kedron] Network Connection (rev 61) Subsystem: Intel Corporation Lenovo ThinkPad T51 Physical Slot: 3 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR+ <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 45 Region 0: Memory at f7f00000 (64-bit, non-prefetchable) [size=8K] Capabilities: [c8] Power Management version 3 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [d0] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0300c Data: 4142 Capabilities: [e0] Express (v1) Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <512ns, L1 unlimited ExtTag- AttnBtn- AttnInd- PwrInd- RBE+ FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd+ ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr+ UncorrErr+ FatalErr- UnsuppReq+ AuxPwr+ TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x1, ASPM L0s L1, Exit Latency L0s <128ns, L1 <64us ClockPM+ Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- Capabilities: [100 v1] Advanced Error Reporting UESta: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq+ ACSViol- UEMsk: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UESvrt: DLP+ SDES- TLP- FCP+ CmpltTO- CmpltAbrt- UnxCmplt- RxOF+ MalfTLP+ ECRC- UnsupReq- ACSViol- CESta: RxErr+ BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr+ CEMsk: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr+ AERCap: First Error Pointer: 14, GenCap- CGenEn- ChkCap- ChkEn- Capabilities: [140 v1] Device Serial Number 00-1d-e0-ff-ff-6b-9c-3f Kernel driver in use: iwl4965 05:00.0 CardBus bridge: Ricoh Co Ltd RL5c476 II (rev ba) Subsystem: Lenovo ThinkPad R61 Physical Slot: 1 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 168, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 16 Region 0: Memory at d7eff000 (32-bit, non-prefetchable) [size=4K] Bus: primary=05, secondary=06, subordinate=07, sec-latency=176 Memory window 0: d8000000-dbffffff (prefetchable) Memory window 1: c0000000-c3ffffff I/O window 0: 00004000-000040ff I/O window 1: 00004400-000044ff BridgeCtl: Parity- SERR- ISA- VGA- MAbort- >Reset- 16bInt+ PostWrite+ 16-bit legacy interface ports at 0001 Capabilities: [dc] Power Management version 2 Flags: PMEClk+ DSI- D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=2 PME- Bridge: PM- B3+ Kernel driver in use: yenta_cardbus 05:00.2 SD Host controller: Ricoh Co Ltd R5C822 SD/SDIO/MMC/MS/MSPro Host Adapter (rev 21) Subsystem: Lenovo ThinkPad W500 Physical Slot: 1 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 Interrupt: pin C routed to IRQ 18 Region 0: Memory at d7efec00 (32-bit, non-prefetchable) [size=256] Capabilities: [80] Power Management version 2 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=2 PME- Kernel driver in use: sdhci-pci [7.6.] SCSI information (from /proc/scsi/scsi) Attached devices: Host: scsi0 Channel: 00 Id: 00 Lun: 00 Vendor: ATA Model: ST9200420AS Rev: 3.AA Type: Direct-Access ANSI SCSI revision: 05 [7.7.] Other information that might be relevant to the problem (please look in /proc and include all information that you think to be relevant): [X.] Other notes, patches, fixes, workarounds: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1324384 ---- ALSA info for 5ccc618fee67f0f0b2122dd4b32a02fd2b6a1569 BEGIN ---- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.60 !!################################ !!Script ran on: Sun Jun 1 02:56:49 UTC 2014 !!Linux Distribution Ubuntu 14.04 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 14.04 LTS" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 14.04 LTS" HOME_URL="http://www.ubuntu.com/" SUPPORT_URL="http://help.ubuntu.com/" BUG_REPORT_URL="http://bugs.launchpad.net/ubuntu/" !!DMI Information Manufacturer: LENOVO Product Name: 76693JG Product Version: ThinkPad X61s !!Kernel Information Kernel release: 3.10-rc7-359-g5ccc618-generic Operating System: GNU/Linux Architecture: x86_64 Processor: x86_64 SMP Enabled: Yes !!ALSA Version Driver version: k3.10-rc7-359-g5ccc618-generic Library version: 1.0.27.2 Utilities version: 1.0.27.2 !!Loaded ALSA modules snd_hda_intel thinkpad_acpi !!Sound Servers on this system Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes !!Soundcards recognised by ALSA 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf8220000 irq 46 4 [ThinkPadEC ]: ThinkPad EC - ThinkPad Console Audio Control ThinkPad Console Audio Control at EC reg 0x30, fw 7MHT25WW-1.03 !!PCI Soundcards installed in the system 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) !!Advanced information - PCI Vendor/Device/Subsystem ID's 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 17aa:20ac !!Modprobe options (Sound related) snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 snd_usb_audio: index=-2 snd_usb_caiaq: index=-2 snd_usb_ua101: index=-2 snd_usb_us122l: index=-2 snd_usb_usx2y: index=-2 snd_cmipci: mpu_port=0x330 fm_port=0x388 snd_pcsp: index=-2 snd_usb_audio: index=-2 !!Loaded sound module options !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 1 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!Module: thinkpad_acpi brightness_enable : 2 brightness_mode : 4 dbg_bluetoothemul : 0 dbg_uwbemul : 0 dbg_wlswemul : 0 dbg_wwanemul : 0 enable : Y experimental : 0 fan_control : N force_load : N hotkey_report_mode : 0 id : ThinkPadEC index : -536870912 volume_capabilities : 0 volume_control : N volume_mode : 3 !!HDA-Intel Codec information --startcollapse-- Codec: Analog Devices AD1984 Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x11d41984 Subsystem Id: 0x17aa20d6 Revision Id: 0x100400 No Modem Function Group found Default PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Default Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 State of AFG node 0x01: Power states: D0 D3 Power: setting=D0, actual=D0 GPIO: io=3, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=1, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 3 samples Connection: 3 0x01* 0x08 0x09 Node 0x03 [Audio Output] wcaps 0x405: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="AD1984 Analog", type="Audio", device=0 Device: name="AD1984 Alt Analog", type="Audio", device=2 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 Amp-Out vals: [0x27 0x27] Converter: stream=0, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x405: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0 Node 0x05 [Audio Input] wcaps 0x10050b: Stereo Amp-In Amp-In caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-In vals: [0xa7 0xa7] Converter: stream=0, channel=0 SDI-Select: 0 Power states: D0 D3 Power: setting=D0, actual=D0 Connection: 1 0x17 Node 0x06 [Audio Input] wcaps 0x10050b: Stereo Amp-In Amp-In caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-In vals: [0xa7 0xa7] Converter: stream=0, channel=0 SDI-Select: 0 Power states: D0 D3 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x07 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x22 0x21 Node 0x08 [Audio Input] wcaps 0x100501: Stereo Device: name="AD1984 Analog", type="Audio", device=0 Converter: stream=0, channel=0 SDI-Select: 0 Power states: D0 D3 Power: setting=D0, actual=D0 Connection: 1 0x0c Node 0x09 [Audio Input] wcaps 0x100501: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power states: D0 D3 Power: setting=D0, actual=D0 Connection: 1 0x0d Node 0x0a [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x04 0x21 Node 0x0b [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x0f 0x21 Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0x28 0x28] Connection: 5 0x14 0x15* 0x16 0x20 0x25 Node 0x0d [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0xa7 0xa7] Connection: 5 0x14* 0x15 0x16 0x20 0x25 Node 0x0e [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03* 0x04 Node 0x0f [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03* 0x04 Node 0x10 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x0f] Node 0x11 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000001f: OUT HP Detect Trigger ImpSense Pin Default 0x0421401f: [Jack] HP Out at Ext Right Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Connection: 1 0x07 Node 0x12 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001001f: OUT HP EAPD Detect Trigger ImpSense EAPD 0x0: Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 Power: setting=D0, actual=D0 Connection: 1 0x0a Node 0x13 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f] Pincap 0x00010010: OUT EAPD EAPD 0x0: Pin Default 0x511301f0: [N/A] Speaker at Int Rear Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D3 Power: setting=D0, actual=D0 Connection: 1 0x1f Node 0x14 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003727: IN Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 100 Pin Default 0x04a15021: [Jack] Mic at Ext Right Conn = 1/8, Color = Red DefAssociation = 0x2, Sequence = 0x1 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=03, enabled=1 Node 0x15 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003727: IN Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 100 Pin Default 0x90a7012e: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x2, Sequence = 0xe Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x0b Node 0x17 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x55a601f0: [N/A] Mic at Int Top Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN Node 0x18 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x55a601f0: [N/A] Mic at Int Top Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x19 [Power Widget] wcaps 0x500500: Mono Power states: D0 D3 Power: setting=D0, actual=D0 Connection: 2 0x20 0x21 Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x91f311f0: [Fixed] Other at Int Rear Conn = ATAPI, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1b [Pin Complex] wcaps 0x40030d: Stereo Digital Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=1 Amp-Out vals: [0xa7 0xa7] Pincap 0x00000010: OUT Pin Default 0x614411f0: [N/A] SPDIF Out at Sep Rear Conn = RCA, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x1c [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Dock Mic Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 100 Pin Default 0x21a15020: [Jack] Mic at Sep Rear Conn = 1/8, Color = Red DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 1 0x24 Node 0x1d [Vendor Defined Widget] wcaps 0xf00100: Mono Connection: 25 0x07* 0x19 0x0a 0x0b 0x0c 0x0d 0x0e 0x0f 0x1a 0x1c 0x11 0x12 0x13 0x14 0x15 0x16 0x1e 0x1f 0x20 0x21 0x22 0x23 0x24 0x25 0x26 Node 0x1e [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x0e 0x21 Node 0x1f [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x1e Node 0x20 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Internal Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Internal Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Dock Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Dock Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x9f 0x9f] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 5 0x14 0x15 0x16 0x1a 0x25 Node 0x21 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Connection: 1 0x20 Node 0x22 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03* 0x04 Node 0x23 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03* 0x04 Node 0x24 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x23 0x21 Node 0x25 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Dock Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 1 0x1c Node 0x26 [Vendor Defined Widget] wcaps 0xf00100: Mono Connection: 3 0x14* 0x15 0x1c --endcollapse-- !!ALSA Device nodes crw-rw----+ 1 root audio 116, 7 May 31 18:02 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 2 May 31 18:02 /dev/snd/controlC4 crw-rw----+ 1 root audio 116, 6 May 31 18:02 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 5 May 31 18:07 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 4 May 31 19:13 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 3 May 31 18:02 /dev/snd/pcmC0D2p crw-rw----+ 1 root audio 116, 1 May 31 18:02 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 May 31 18:02 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 80 May 31 18:02 . drwxr-xr-x 3 root root 220 May 31 18:02 .. lrwxrwxrwx 1 root root 12 May 31 18:02 pci-0000:00:1b.0 -> ../controlC0 lrwxrwxrwx 1 root root 12 May 31 18:02 platform-thinkpad_acpi -> ../controlC4 !!Aplay/Arecord output APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD1984 Analog [AD1984 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 2: AD1984 Alt Analog [AD1984 Alt Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD1984 Analog [AD1984 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0xf8220000 irq 46' Mixer name : 'Analog Devices AD1984' Components : 'HDA:11d41984,17aa20d6,00100400' Controls : 31 Simple ctrls : 15 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 39 Mono: Playback 39 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 39 Mono: Front Left: Playback 39 [100%] [0.00dB] [on] Front Right: Playback 39 [100%] [0.00dB] [on] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 39 Mono: Front Left: Playback 0 [0%] [-58.50dB] [off] Front Right: Playback 0 [0%] [-58.50dB] [off] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Beep',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 15 Mono: Playback 15 [100%] [0.00dB] [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 54 Front Left: Capture 40 [74%] [1.50dB] [on] Front Right: Capture 40 [74%] [1.50dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Enabled' Simple mixer control 'Digital',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 60 [50%] [0.00dB] Front Right: Capture 60 [50%] [0.00dB] Simple mixer control 'Dock Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Dock Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Independent HP',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' Simple mixer control 'Internal Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [off] Front Right: Playback 31 [100%] [12.00dB] [off] Simple mixer control 'Internal Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] !!-------Mixer controls for card 4 [ThinkPadEC] Card hw:4 'ThinkPadEC'/'ThinkPad Console Audio Control at EC reg 0x30, fw 7MHT25WW-1.03' Mixer name : 'ThinkPad EC 7MHT25WW-1.03' Components : '' Controls : 1 Simple ctrls : 1 Simple mixer control 'Console',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] !!Alsactl output --startcollapse-- state.Intel { control.1 { iface MIXER name 'Headphone Playback Volume' value.0 39 value.1 39 comment { access 'read write' type INTEGER count 2 range '0 - 39' dbmin -5850 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.2 { iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.3 { iface MIXER name 'Speaker Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 39' dbmin -5850 dbmax 0 dbvalue.0 -5850 dbvalue.1 -5850 } } control.4 { iface MIXER name 'Speaker Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.5 { iface MIXER name 'Independent HP' value Disabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.6 { iface MIXER name 'Internal Mic Playback Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 1200 dbvalue.1 1200 } } control.7 { iface MIXER name 'Internal Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.8 { iface MIXER name 'Dock Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.9 { iface MIXER name 'Dock Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.10 { iface MIXER name 'Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.11 { iface MIXER name 'Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.12 { iface MIXER name 'Auto-Mute Mode' value Enabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.13 { iface MIXER name 'Capture Volume' value.0 40 value.1 40 comment { access 'read write' type INTEGER count 2 range '0 - 54' dbmin -5850 dbmax 2250 dbvalue.0 150 dbvalue.1 150 } } control.14 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.15 { iface MIXER name 'Internal Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.16 { iface MIXER name 'Dock Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.17 { iface MIXER name 'Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.18 { iface MIXER name 'Master Playback Volume' value 39 comment { access 'read write' type INTEGER count 1 range '0 - 39' dbmin -5850 dbmax 0 dbvalue.0 0 } } control.19 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.20 { iface CARD name 'Internal Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.21 { iface CARD name 'Dock Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.22 { iface CARD name 'Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.23 { iface CARD name 'Headphone Jack' value true comment { access read type BOOLEAN count 1 } } control.24 { iface CARD name 'Speaker Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.25 { iface MIXER name 'Beep Playback Volume' value 15 comment { access 'read write' type INTEGER count 1 range '0 - 15' dbmin -4500 dbmax 0 dbvalue.0 0 } } control.26 { iface MIXER name 'Beep Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.27 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.28 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.29 { iface PCM device 2 name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.30 { iface MIXER name 'Digital Capture Volume' value.0 60 value.1 60 comment { access 'read write user' type INTEGER count 2 range '0 - 120' tlv '0000000100000008fffff44800000032' dbmin -3000 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.31 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } } state.ThinkPadEC { control.1 { iface MIXER name 'Console Playback Switch' value true comment { access read type BOOLEAN count 1 } } } --endcollapse-- !!All Loaded Modules Module btrfs raid6_pq zlib_deflate xor ufs qnx4 hfsplus hfs minix ntfs msdos jfs xfs libcrc32c nls_iso8859_1 ipt_MASQUERADE iptable_nat nf_nat_ipv4 nf_nat nf_conntrack_ipv4 nf_defrag_ipv4 xt_conntrack nf_conntrack ipt_REJECT xt_CHECKSUM iptable_mangle xt_tcpudp bridge stp llc ip6table_filter ip6_tables iptable_filter ip_tables ebtable_nat ebtables x_tables bnep rfcomm bluetooth binfmt_misc snd_hda_codec_analog dm_multipath snd_hda_intel arc4 snd_hda_codec scsi_dh snd_hwdep snd_pcm iwl4965 thinkpad_acpi iwlegacy mac80211 nvram snd_page_alloc snd_seq_midi snd_seq_midi_event cfg80211 coretemp snd_rawmidi kvm_intel snd_seq snd_seq_device pcmcia kvm snd_timer serio_raw snd parport_pc soundcore ppdev yenta_socket pcmcia_rsrc pcmcia_core lpc_ich lp mac_hid parport xts gf128mul dm_crypt mmc_block psmouse sdhci_pci sdhci ahci libahci i915 i2c_algo_bit drm_kms_helper drm e1000e video ptp pps_core !!Sysfs Files /sys/class/sound/hwC0D0/init_pin_configs: 0x11 0x0421401f 0x12 0x90170110 0x13 0x511301f0 0x14 0x04a15021 0x15 0x90a7012e 0x16 0x593301f0 0x17 0x55a601f0 0x18 0x55a601f0 0x1a 0x91f311f0 0x1b 0x614411f0 0x1c 0x21a15020 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: !!ALSA/HDA dmesg [ 35.045988] ieee80211 phy0: Selected rate control algorithm 'iwl-4965-rs' [ 35.076727] hda_intel: probe_mask set to 0x1 for device 17aa:20ac [ 35.076782] snd_hda_intel 0000:00:1b.0: irq 46 for MSI/MSI-X [ 35.084674] device-mapper: multipath: version 1.5.1 loaded -- [ 35.496579] Mic=0x14 [ 35.504968] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input7 [ 35.505383] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 [ 35.506228] input: HDA Intel Dock Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 [ 35.569962] pcmcia_socket pcmcia_socket0: cs: memory probe 0x0c0000-0x0fffff: ---- ALSA info for 5ccc618fee67f0f0b2122dd4b32a02fd2b6a1569 END ---- I had do do diff between 5ccc618f and aa95d61b4 to fit 100k email limit. ---- ALSA info for aa95d61b43e0fcb0b2ce68e5efa37174fd9e5cd3 BEGIN ---- 6c6 < !!Script ran on: Sun Jun 1 02:56:49 UTC 2014 --- > !!Script ran on: Sun Jun 1 05:51:12 UTC 2014 26c26 < Kernel release: 3.10-rc7-359-g5ccc618-generic --- > Kernel release: 3.10-rc7-358-gaa95d61-generic 36c36 < Driver version: k3.10-rc7-359-g5ccc618-generic --- > Driver version: k3.10-rc7-358-gaa95d61-generic 60c60 < HDA Intel at 0xf8220000 irq 46 --- > HDA Intel at 0xf8220000 irq 45 156c156 < IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 --- > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=1, unsol=0 160c160,167 < Converter: stream=0, channel=0 --- > Control: name="IEC958 Playback Con Mask", index=0, device=0 > Control: name="IEC958 Playback Pro Mask", index=0, device=0 > Control: name="IEC958 Playback Default", index=0, device=0 > Control: name="IEC958 Playback Switch", index=0, device=0 > Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 > Control: name="IEC958 Playback Source", index=0, device=0 > Device: name="AD198x Digital", type="SPDIF", device=1 > Converter: stream=8, channel=0 172,175d178 < Control: name="Headphone Playback Volume", index=0, device=0 < ControlAmp: chs=3, dir=Out, idx=0, ofs=0 < Device: name="AD1984 Analog", type="Audio", device=0 < Device: name="AD1984 Alt Analog", type="Audio", device=2 177c180 < Amp-Out vals: [0x27 0x27] --- > Amp-Out vals: [0x00 0x00] 182c185 < Control: name="Speaker Playback Volume", index=0, device=0 --- > Control: name="PCM Playback Volume", index=0, device=0 183a187 > Device: name="AD198x Analog", type="Audio", device=0 185,186c189,190 < Amp-Out vals: [0x00 0x00] < Converter: stream=0, channel=0 --- > Amp-Out vals: [0x27 0x27] > Converter: stream=8, channel=0 209c213 < Amp-In vals: [0x00 0x00] [0x80 0x80] --- > Amp-In vals: [0x00 0x00] [0x00 0x00] 213c217 < Device: name="AD1984 Analog", type="Audio", device=0 --- > Device: name="AD198x Analog", type="Audio", device=0 229c233 < Amp-In vals: [0x00 0x00] [0x80 0x80] --- > Amp-In vals: [0x00 0x00] [0x00 0x00] 241a246 > Control: name="Input Source", index=0, device=0 246a252,256 > Control: name="Capture Volume", index=1, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Control: name="Capture Switch", index=1, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Control: name="Input Source", index=1, device=0 253c263 < 0x03* 0x04 --- > 0x03 0x04* 258c268 < Control: name="Beep Playback Volume", index=0, device=0 --- > Control: name="Digital Beep Playback Volume", index=0, device=0 260c270 < Control: name="Beep Playback Switch", index=0, device=0 --- > Control: name="Digital Beep Playback Switch", index=0, device=0 263c273 < Amp-Out vals: [0x0f] --- > Amp-Out vals: [0x8f] 267d276 < Control: name="Headphone Jack", index=0, device=0 275c284 < Unsolicited: tag=01, enabled=1 --- > Unsolicited: tag=00, enabled=0 281d289 < Control: name="Speaker Phantom Jack", index=0, device=0 283c291 < Amp-Out vals: [0x80 0x80] --- > Amp-Out vals: [0x00 0x00] 285c293 < EAPD 0x0: --- > EAPD 0x2: EAPD 290c298 < Pin-ctls: 0x40: OUT --- > Pin-ctls: 0xc0: OUT HP 298c306 < Amp-Out vals: [0x9f] --- > Amp-Out vals: [0x80] 313d320 < Control: name="Mic Jack", index=0, device=0 322c329 < Unsolicited: tag=03, enabled=1 --- > Unsolicited: tag=00, enabled=0 326d332 < Control: name="Internal Mic Phantom Jack", index=0, device=0 375a382,383 > Control: name="IEC958 Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 377c385 < Amp-Out vals: [0xa7 0xa7] --- > Amp-Out vals: [0x27 0x27] 387d394 < Control: name="Dock Mic Jack", index=0, device=0 396c403 < Unsolicited: tag=02, enabled=1 --- > Unsolicited: tag=00, enabled=0 404c411 < Amp-In vals: [0x80 0x80] [0x80 0x80] --- > Amp-In vals: [0x00 0x00] [0x00 0x00] 410a418,421 > Control: name="Mic Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Control: name="Mic Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 414a426,429 > Control: name="Beep Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=3, ofs=0 > Control: name="Beep Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=3, ofs=0 419,422d433 < Control: name="Mic Playback Volume", index=0, device=0 < ControlAmp: chs=3, dir=In, idx=0, ofs=0 < Control: name="Mic Playback Switch", index=0, device=0 < ControlAmp: chs=3, dir=In, idx=0, ofs=0 424c435 < Amp-In vals: [0x80 0x80] [0x9f 0x9f] [0x80 0x80] [0x80 0x80] [0x80 0x80] --- > Amp-In vals: [0x80 0x80] [0x9f 0x9f] [0x80 0x80] [0x17 0x17] [0x80 0x80] 429c440 < Amp-Out vals: [0x9f 0x9f] --- > Amp-Out vals: [0x1f 0x1f] 434c445 < 0x03* 0x04 --- > 0x03 0x04* 459,466c470,477 < crw-rw----+ 1 root audio 116, 7 May 31 18:02 /dev/snd/controlC0 < crw-rw----+ 1 root audio 116, 2 May 31 18:02 /dev/snd/controlC4 < crw-rw----+ 1 root audio 116, 6 May 31 18:02 /dev/snd/hwC0D0 < crw-rw----+ 1 root audio 116, 5 May 31 18:07 /dev/snd/pcmC0D0c < crw-rw----+ 1 root audio 116, 4 May 31 19:13 /dev/snd/pcmC0D0p < crw-rw----+ 1 root audio 116, 3 May 31 18:02 /dev/snd/pcmC0D2p < crw-rw----+ 1 root audio 116, 1 May 31 18:02 /dev/snd/seq < crw-rw----+ 1 root audio 116, 33 May 31 18:02 /dev/snd/timer --- > crw-rw----+ 1 root audio 116, 7 May 31 22:39 /dev/snd/controlC0 > crw-rw----+ 1 root audio 116, 2 May 31 22:39 /dev/snd/controlC4 > crw-rw----+ 1 root audio 116, 6 May 31 22:39 /dev/snd/hwC0D0 > crw-rw----+ 1 root audio 116, 5 May 31 22:44 /dev/snd/pcmC0D0c > crw-rw----+ 1 root audio 116, 4 May 31 22:44 /dev/snd/pcmC0D0p > crw-rw----+ 1 root audio 116, 3 May 31 22:44 /dev/snd/pcmC0D1p > crw-rw----+ 1 root audio 116, 1 May 31 22:39 /dev/snd/seq > crw-rw----+ 1 root audio 116, 33 May 31 22:39 /dev/snd/timer 470,473c481,484 < drwxr-xr-x 2 root root 80 May 31 18:02 . < drwxr-xr-x 3 root root 220 May 31 18:02 .. < lrwxrwxrwx 1 root root 12 May 31 18:02 pci-0000:00:1b.0 -> ../controlC0 < lrwxrwxrwx 1 root root 12 May 31 18:02 platform-thinkpad_acpi -> ../controlC4 --- > drwxr-xr-x 2 root root 80 May 31 22:39 . > drwxr-xr-x 3 root root 220 May 31 22:39 .. > lrwxrwxrwx 1 root root 12 May 31 22:39 pci-0000:00:1b.0 -> ../controlC0 > lrwxrwxrwx 1 root root 12 May 31 22:39 platform-thinkpad_acpi -> ../controlC4 482,483c493,494 < card 0: Intel [HDA Intel], device 0: AD1984 Analog [AD1984 Analog] < Subdevices: 1/1 --- > card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] > Subdevices: 0/1 485c496 < card 0: Intel [HDA Intel], device 2: AD1984 Alt Analog [AD1984 Alt Analog] --- > card 0: Intel [HDA Intel], device 1: AD198x Digital [AD198x Digital] 492,493c503,504 < card 0: Intel [HDA Intel], device 0: AD1984 Analog [AD1984 Analog] < Subdevices: 1/1 --- > card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] > Subdevices: 2/2 494a506 > Subdevice #1: subdevice #1 501c513 < Card hw:0 'Intel'/'HDA Intel at 0xf8220000 irq 46' --- > Card hw:0 'Intel'/'HDA Intel at 0xf8220000 irq 45' 504,505c516,517 < Controls : 31 < Simple ctrls : 15 --- > Controls : 36 > Simple ctrls : 20 512c524 < Capabilities: pvolume pswitch --- > Capabilities: pswitch 514d525 < Limits: Playback 0 - 39 516,517c527,528 < Front Left: Playback 39 [100%] [0.00dB] [on] < Front Right: Playback 39 [100%] [0.00dB] [on] --- > Front Left: Playback [on] > Front Right: Playback [on] 519c530 < Capabilities: pvolume pswitch --- > Capabilities: pswitch 521d531 < Limits: Playback 0 - 39 523,524c533,534 < Front Left: Playback 0 [0%] [-58.50dB] [off] < Front Right: Playback 0 [0%] [-58.50dB] [off] --- > Front Left: Playback [on] > Front Right: Playback [on] 528c538 < Limits: Playback 0 - 255 --- > Limits: Playback 0 - 39 530,531c540,541 < Front Left: Playback 255 [100%] [0.00dB] < Front Right: Playback 255 [100%] [0.00dB] --- > Front Left: Playback 39 [100%] [0.00dB] > Front Right: Playback 39 [100%] [0.00dB] 546,547c556,564 < Simple mixer control 'Beep',0 < Capabilities: pvolume pvolume-joined pswitch pswitch-joined --- > Simple mixer control 'IEC958',0 > Capabilities: pvolume pswitch pswitch-joined > Playback channels: Front Left - Front Right > Limits: Playback 0 - 39 > Mono: > Front Left: Playback 39 [100%] [0.00dB] [off] > Front Right: Playback 39 [100%] [0.00dB] [off] > Simple mixer control 'IEC958 Default PCM',0 > Capabilities: pswitch pswitch-joined 549,550c566,577 < Limits: Playback 0 - 15 < Mono: Playback 15 [100%] [0.00dB] [on] --- > Mono: Playback [on] > Simple mixer control 'IEC958 Playback Source',0 > Capabilities: enum > Items: 'PCM' 'ADC' > Item0: 'PCM' > Simple mixer control 'Beep',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 31 > Mono: > Front Left: Playback 23 [74%] [0.00dB] [on] > Front Right: Playback 23 [74%] [0.00dB] [on] 557,560c584,589 < Simple mixer control 'Auto-Mute Mode',0 < Capabilities: enum < Items: 'Disabled' 'Enabled' < Item0: 'Enabled' --- > Simple mixer control 'Capture',1 > Capabilities: cvolume cswitch > Capture channels: Front Left - Front Right > Limits: Capture 0 - 54 > Front Left: Capture 39 [72%] [0.00dB] [off] > Front Right: Capture 39 [72%] [0.00dB] [off] 566a596,600 > Simple mixer control 'Digital Beep',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined > Playback channels: Mono > Limits: Playback 0 - 15 > Mono: Playback 15 [100%] [0.00dB] [off] 581,584c615,622 < Simple mixer control 'Independent HP',0 < Capabilities: enum < Items: 'Disabled' 'Enabled' < Item0: 'Disabled' --- > Simple mixer control 'Input Source',0 > Capabilities: cenum > Items: 'Mic' 'Internal Mic' 'Mix' 'Dock Mic' > Item0: 'Internal Mic' > Simple mixer control 'Input Source',1 > Capabilities: cenum > Items: 'Mic' 'Internal Mic' 'Mix' 'Dock Mic' > Item0: 'Mic' 620c658 < name 'Headphone Playback Volume' --- > name 'PCM Playback Volume' 647,649c685,687 < name 'Speaker Playback Volume' < value.0 0 < value.1 0 --- > name 'Speaker Playback Switch' > value.0 true > value.1 true 652c690 < type INTEGER --- > type BOOLEAN 654,658d691 < range '0 - 39' < dbmin -5850 < dbmax 0 < dbvalue.0 -5850 < dbvalue.1 -5850 663,665c696,698 < name 'Speaker Playback Switch' < value.0 false < value.1 false --- > name 'Mic Playback Volume' > value.0 0 > value.1 0 668c701 < type BOOLEAN --- > type INTEGER 669a703,707 > range '0 - 31' > dbmin -3450 > dbmax 1200 > dbvalue.0 -3450 > dbvalue.1 -3450 674,675c712,714 < name 'Independent HP' < value Disabled --- > name 'Mic Playback Switch' > value.0 false > value.1 false 678,681c717,718 < type ENUMERATED < count 1 < item.0 Disabled < item.1 Enabled --- > type BOOLEAN > count 2 713,715c750,752 < name 'Dock Mic Playback Volume' < value.0 0 < value.1 0 --- > name 'Beep Playback Volume' > value.0 23 > value.1 23 723,724c760,761 < dbvalue.0 -3450 < dbvalue.1 -3450 --- > dbvalue.0 0 > dbvalue.1 0 729,731c766,768 < name 'Dock Mic Playback Switch' < value.0 false < value.1 false --- > name 'Beep Playback Switch' > value.0 true > value.1 true 740c777 < name 'Mic Playback Volume' --- > name 'Dock Mic Playback Volume' 756c793 < name 'Mic Playback Switch' --- > name 'Dock Mic Playback Switch' 767,768c804,806 < name 'Auto-Mute Mode' < value Enabled --- > name 'Mic Boost Volume' > value.0 0 > value.1 0 771,774c809,815 < type ENUMERATED < count 1 < item.0 Disabled < item.1 Enabled --- > type INTEGER > count 2 > range '0 - 3' > dbmin 0 > dbmax 3000 > dbvalue.0 0 > dbvalue.1 0 778a820,851 > name 'Internal Mic Boost Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 3' > dbmin 0 > dbmax 3000 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.14 { > iface MIXER > name 'Dock Mic Boost Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 3' > dbmin 0 > dbmax 3000 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.15 { > iface MIXER 793c866 < control.14 { --- > control.16 { 804c877 < control.15 { --- > control.17 { 806,808c879,882 < name 'Internal Mic Boost Volume' < value.0 0 < value.1 0 --- > name 'Capture Volume' > index 1 > value.0 39 > value.1 39 813,815c887,889 < range '0 - 3' < dbmin 0 < dbmax 3000 --- > range '0 - 54' > dbmin -5850 > dbmax 2250 820c894 < control.16 { --- > control.18 { 822,824c896,899 < name 'Dock Mic Boost Volume' < value.0 0 < value.1 0 --- > name 'Capture Switch' > index 1 > value.0 false > value.1 false 827c902 < type INTEGER --- > type BOOLEAN 829,833d903 < range '0 - 3' < dbmin 0 < dbmax 3000 < dbvalue.0 0 < dbvalue.1 0 836c906 < control.17 { --- > control.19 { 838,840c908,909 < name 'Mic Boost Volume' < value.0 0 < value.1 0 --- > name 'Input Source' > value 'Internal Mic' 843,849c912,917 < type INTEGER < count 2 < range '0 - 3' < dbmin 0 < dbmax 3000 < dbvalue.0 0 < dbvalue.1 0 --- > type ENUMERATED > count 1 > item.0 Mic > item.1 'Internal Mic' > item.2 Mix > item.3 'Dock Mic' 852c920 < control.18 { --- > control.20 { 854,855c922,924 < name 'Master Playback Volume' < value 39 --- > name 'Input Source' > index 1 > value Mic 858c927 < type INTEGER --- > type ENUMERATED 859a929,943 > item.0 Mic > item.1 'Internal Mic' > item.2 Mix > item.3 'Dock Mic' > } > } > control.21 { > iface MIXER > name 'IEC958 Playback Volume' > value.0 39 > value.1 39 > comment { > access 'read write' > type INTEGER > count 2 863a948 > dbvalue.1 0 866c951 < control.19 { --- > control.22 { 868,869c953,954 < name 'Master Playback Switch' < value true --- > name 'IEC958 Playback Source' > value PCM 872c957 < type BOOLEAN --- > type ENUMERATED 873a959,960 > item.0 PCM > item.1 ADC 876,879c963,966 < control.20 { < iface CARD < name 'Internal Mic Phantom Jack' < value true --- > control.23 { > iface MIXER > name 'IEC958 Playback Con Mask' > value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' 882c969 < type BOOLEAN --- > type IEC958 886,889c973,976 < control.21 { < iface CARD < name 'Dock Mic Jack' < value false --- > control.24 { > iface MIXER > name 'IEC958 Playback Pro Mask' > value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' 892c979 < type BOOLEAN --- > type IEC958 896,899c983,986 < control.22 { < iface CARD < name 'Mic Jack' < value false --- > control.25 { > iface MIXER > name 'IEC958 Playback Default' > value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' 901,902c988,989 < access read < type BOOLEAN --- > access 'read write' > type IEC958 906,909c993,996 < control.23 { < iface CARD < name 'Headphone Jack' < value true --- > control.26 { > iface MIXER > name 'IEC958 Playback Switch' > value false 911c998 < access read --- > access 'read write' 916,918c1003,1005 < control.24 { < iface CARD < name 'Speaker Phantom Jack' --- > control.27 { > iface MIXER > name 'IEC958 Default PCM Playback Switch' 921c1008 < access read --- > access 'read write' 926c1013 < control.25 { --- > control.28 { 928c1015 < name 'Beep Playback Volume' --- > name 'Digital Beep Playback Volume' 940c1027 < control.26 { --- > control.29 { 942c1029,1053 < name 'Beep Playback Switch' --- > name 'Digital Beep Playback Switch' > value false > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.30 { > iface MIXER > name 'Master Playback Volume' > value 39 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 39' > dbmin -5850 > dbmax 0 > dbvalue.0 0 > } > } > control.31 { > iface MIXER > name 'Master Playback Switch' 950c1061 < control.27 { --- > control.32 { 952a1064,1075 > value.0 3 > value.1 4 > comment { > access read > type INTEGER > count 2 > range '0 - 36' > } > } > control.33 { > iface PCM > name 'Capture Channel Map' 962c1085 < control.28 { --- > control.34 { 964a1088 > index 1 974c1098 < control.29 { --- > control.35 { 976c1100 < device 2 --- > device 1 987c1111 < control.30 { --- > control.36 { 1004,1020d1127 < control.31 { < iface MIXER < name 'PCM Playback Volume' < value.0 255 < value.1 255 < comment { < access 'read write user' < type INTEGER < count 2 < range '0 - 255' < tlv '0000000100000008ffffec1400000014' < dbmin -5100 < dbmax 0 < dbvalue.0 0 < dbvalue.1 0 < } < } 1041,1054d1147 < btrfs < raid6_pq < zlib_deflate < xor < ufs < qnx4 < hfsplus < hfs < minix < ntfs < msdos < jfs < xfs < libcrc32c 1078d1170 < bnep 1079a1172 > bnep 1082c1175 < snd_hda_codec_analog --- > arc4 1083a1177,1182 > snd_hda_codec_analog > iwl4965 > scsi_dh > iwlegacy > mac80211 > coretemp 1085c1184 < arc4 --- > cfg80211 1087c1186,1187 < scsi_dh --- > kvm_intel > kvm 1089,1090d1188 < snd_pcm < iwl4965 1092,1094c1190 < iwlegacy < mac80211 < nvram --- > snd_pcm 1095a1192,1193 > nvram > pcmcia 1098,1099d1195 < cfg80211 < coretemp 1101c1197 < kvm_intel --- > serio_raw 1102a1199 > yenta_socket 1104,1105c1201 < pcmcia < kvm --- > pcmcia_rsrc 1107c1203,1204 < serio_raw --- > lpc_ich > pcmcia_core 1109d1205 < parport_pc 1110a1207,1208 > parport_pc > mac_hid 1112,1115d1209 < yenta_socket < pcmcia_rsrc < pcmcia_core < lpc_ich 1117d1210 < mac_hid 1123,1127d1215 < psmouse < sdhci_pci < sdhci < ahci < libahci 1128a1217,1219 > e1000e > ahci > psmouse 1130a1222 > libahci 1132,1133c1224,1225 < e1000e < video --- > sdhci_pci > sdhci 1135a1228 > video 1164,1173c1257,1260 < [ 35.045988] ieee80211 phy0: Selected rate control algorithm 'iwl-4965-rs' < [ 35.076727] hda_intel: probe_mask set to 0x1 for device 17aa:20ac < [ 35.076782] snd_hda_intel 0000:00:1b.0: irq 46 for MSI/MSI-X < [ 35.084674] device-mapper: multipath: version 1.5.1 loaded < -- < [ 35.496579] Mic=0x14 < [ 35.504968] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input7 < [ 35.505383] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 < [ 35.506228] input: HDA Intel Dock Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 < [ 35.569962] pcmcia_socket pcmcia_socket0: cs: memory probe 0x0c0000-0x0fffff: --- > [ 38.133419] cfg80211: Calling CRDA to update world regulatory domain > [ 38.179214] hda_intel: probe_mask set to 0x1 for device 17aa:20ac > [ 38.179301] snd_hda_intel 0000:00:1b.0: irq 45 for MSI/MSI-X > [ 38.531005] EXT4-fs (sda3): mounted filesystem with ordered data mode. Opts: (null) ---- ALSA info for aa95d61b43e0fcb0b2ce68e5efa37174fd9e5cd3 END ---- ---- ALSA info for latest daily upstream BEGIN ---- 6c6 < !!Script ran on: Sun Jun 1 02:56:49 UTC 2014 --- > !!Script ran on: Sun Jun 1 06:36:37 UTC 2014 26c26 < Kernel release: 3.10-rc7-359-g5ccc618-generic --- > Kernel release: 3.15.0-999-generic 36c36 < Driver version: k3.10-rc7-359-g5ccc618-generic --- > Driver version: k3.15.0-999-generic 61c61 < 4 [ThinkPadEC ]: ThinkPad EC - ThinkPad Console Audio Control --- > 29 [ThinkPadEC ]: ThinkPad EC - ThinkPad Console Audio Control 109c109 < power_save : 1 --- > power_save : 0 127d126 < hotkey_report_mode : 0 156c155 < IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 --- > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=1, unsol=0 185c184 < Amp-Out vals: [0x00 0x00] --- > Amp-Out vals: [0x27 0x27] 209c208 < Amp-In vals: [0x00 0x00] [0x80 0x80] --- > Amp-In vals: [0x00 0x00] [0x00 0x00] 229c228 < Amp-In vals: [0x00 0x00] [0x80 0x80] --- > Amp-In vals: [0x00 0x00] [0x00 0x00] 245c244 < 0x14 0x15* 0x16 0x20 0x25 --- > 0x14* 0x15 0x16 0x20 0x25 258,261d256 < Control: name="Beep Playback Volume", index=0, device=0 < ControlAmp: chs=1, dir=Out, idx=0, ofs=0 < Control: name="Beep Playback Switch", index=0, device=0 < ControlAmp: chs=1, dir=Out, idx=0, ofs=0 263c258 < Amp-Out vals: [0x0f] --- > Amp-Out vals: [0x8f] 283c278 < Amp-Out vals: [0x80 0x80] --- > Amp-Out vals: [0x00 0x00] 285c280 < EAPD 0x0: --- > EAPD 0x2: EAPD 422a418,421 > Control: name="Beep Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=3, ofs=0 > Control: name="Beep Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=3, ofs=0 424c423 < Amp-In vals: [0x80 0x80] [0x9f 0x9f] [0x80 0x80] [0x80 0x80] [0x80 0x80] --- > Amp-In vals: [0x80 0x80] [0x9f 0x9f] [0x80 0x80] [0x17 0x17] [0x80 0x80] 429c428 < Amp-Out vals: [0x9f 0x9f] --- > Amp-Out vals: [0x1f 0x1f] 459,466c458,465 < crw-rw----+ 1 root audio 116, 7 May 31 18:02 /dev/snd/controlC0 < crw-rw----+ 1 root audio 116, 2 May 31 18:02 /dev/snd/controlC4 < crw-rw----+ 1 root audio 116, 6 May 31 18:02 /dev/snd/hwC0D0 < crw-rw----+ 1 root audio 116, 5 May 31 18:07 /dev/snd/pcmC0D0c < crw-rw----+ 1 root audio 116, 4 May 31 19:13 /dev/snd/pcmC0D0p < crw-rw----+ 1 root audio 116, 3 May 31 18:02 /dev/snd/pcmC0D2p < crw-rw----+ 1 root audio 116, 1 May 31 18:02 /dev/snd/seq < crw-rw----+ 1 root audio 116, 33 May 31 18:02 /dev/snd/timer --- > crw-rw----+ 1 root audio 116, 3 May 31 23:24 /dev/snd/controlC0 > crw-rw----+ 1 root audio 116, 2 May 31 23:24 /dev/snd/controlC29 > crw-rw----+ 1 root audio 116, 7 May 31 23:24 /dev/snd/hwC0D0 > crw-rw----+ 1 root audio 116, 5 May 31 23:32 /dev/snd/pcmC0D0c > crw-rw----+ 1 root audio 116, 4 May 31 23:31 /dev/snd/pcmC0D0p > crw-rw----+ 1 root audio 116, 6 May 31 23:24 /dev/snd/pcmC0D2p > crw-rw----+ 1 root audio 116, 1 May 31 23:24 /dev/snd/seq > crw-rw----+ 1 root audio 116, 33 May 31 23:24 /dev/snd/timer 470,473c469,472 < drwxr-xr-x 2 root root 80 May 31 18:02 . < drwxr-xr-x 3 root root 220 May 31 18:02 .. < lrwxrwxrwx 1 root root 12 May 31 18:02 pci-0000:00:1b.0 -> ../controlC0 < lrwxrwxrwx 1 root root 12 May 31 18:02 platform-thinkpad_acpi -> ../controlC4 --- > drwxr-xr-x 2 root root 80 May 31 23:24 . > drwxr-xr-x 3 root root 220 May 31 23:24 .. > lrwxrwxrwx 1 root root 12 May 31 23:24 pci-0000:00:1b.0 -> ../controlC0 > lrwxrwxrwx 1 root root 13 May 31 23:24 platform-thinkpad_acpi -> ../controlC29 523,524c522,523 < Front Left: Playback 0 [0%] [-58.50dB] [off] < Front Right: Playback 0 [0%] [-58.50dB] [off] --- > Front Left: Playback 39 [100%] [0.00dB] [on] > Front Right: Playback 39 [100%] [0.00dB] [on] 547,550c546,551 < Capabilities: pvolume pvolume-joined pswitch pswitch-joined < Playback channels: Mono < Limits: Playback 0 - 15 < Mono: Playback 15 [100%] [0.00dB] [on] --- > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 31 > Mono: > Front Left: Playback 23 [74%] [0.00dB] [on] > Front Right: Playback 23 [74%] [0.00dB] [on] 600c601 < !!-------Mixer controls for card 4 [ThinkPadEC] --- > !!-------Mixer controls for card 29 [ThinkPadEC] 602c603 < Card hw:4 'ThinkPadEC'/'ThinkPad Console Audio Control at EC reg 0x30, fw 7MHT25WW-1.03' --- > Card hw:29 'ThinkPadEC'/'ThinkPad Console Audio Control at EC reg 0x30, fw 7MHT25WW-1.03' 648,649c649,650 < value.0 0 < value.1 0 --- > value.0 39 > value.1 39 657,658c658,659 < dbvalue.0 -5850 < dbvalue.1 -5850 --- > dbvalue.0 0 > dbvalue.1 0 664,665c665,666 < value.0 false < value.1 false --- > value.0 true > value.1 true 899c900 < value false --- > value true 909c910 < value true --- > value false 929c930,931 < value 15 --- > value.0 23 > value.1 23 933,936c935,938 < count 1 < range '0 - 15' < dbmin -4500 < dbmax 0 --- > count 2 > range '0 - 31' > dbmin -3450 > dbmax 1200 937a940 > dbvalue.1 0 943c946,947 < value true --- > value.0 true > value.1 true 947c951 < count 1 --- > count 2 1041,1054c1045,1046 < btrfs < raid6_pq < zlib_deflate < xor < ufs < qnx4 < hfsplus < hfs < minix < ntfs < msdos < jfs < xfs < libcrc32c --- > ctr > ccm 1078d1069 < bnep 1079a1071 > bnep 1080a1073 > 6lowpan_iphc 1083c1076 < dm_multipath --- > snd_hda_codec_generic 1085c1078 < arc4 --- > snd_hda_controller 1087d1079 < scsi_dh 1089,1090d1080 < snd_pcm < iwl4965 1092,1093c1082,1084 < iwlegacy < mac80211 --- > dm_multipath > coretemp > snd_pcm 1095c1086,1087 < snd_page_alloc --- > kvm_intel > scsi_dh 1096a1089,1090 > arc4 > kvm 1098,1099c1092,1094 < cfg80211 < coretemp --- > iwl4965 > iwlegacy > mac80211 1101d1095 < kvm_intel 1102a1097 > cfg80211 1105d1099 < kvm 1107c1101 < serio_raw --- > microcode 1109,1111c1103 < parport_pc < soundcore < ppdev --- > serio_raw 1112a1105,1106 > soundcore > mac_hid 1114c1108 < pcmcia_core --- > ppdev 1115a1110,1111 > pcmcia_core > parport_pc 1117d1112 < mac_hid 1123,1125d1117 < psmouse < sdhci_pci < sdhci 1126a1119,1120 > sdhci_pci > psmouse 1127a1122 > sdhci 1128a1124 > video 1129a1126 > e1000e 1132,1133d1128 < e1000e < video 1164,1167c1159,1162 < [ 35.045988] ieee80211 phy0: Selected rate control algorithm 'iwl-4965-rs' < [ 35.076727] hda_intel: probe_mask set to 0x1 for device 17aa:20ac < [ 35.076782] snd_hda_intel 0000:00:1b.0: irq 46 for MSI/MSI-X < [ 35.084674] device-mapper: multipath: version 1.5.1 loaded --- > [ 43.432228] input: ThinkPad Extra Buttons as /devices/platform/thinkpad_acpi/input/input7 > [ 43.624371] snd_hda_intel 0000:00:1b.0: probe_mask set to 0x1 for device 17aa:20ac > [ 43.624479] snd_hda_intel 0000:00:1b.0: irq 46 for MSI/MSI-X > [ 43.800655] systemd-udevd[578]: renamed network interface wlan0 to wlan2 1169,1173c1164,1176 < [ 35.496579] Mic=0x14 < [ 35.504968] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input7 < [ 35.505383] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 < [ 35.506228] input: HDA Intel Dock Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 < [ 35.569962] pcmcia_socket pcmcia_socket0: cs: memory probe 0x0c0000-0x0fffff: --- > [ 44.022500] cfg80211: (5735000 KHz - 5835000 KHz @ 40000 KHz), (300 mBi, 2000 mBm), (N/A) > [ 44.101116] sound hdaudioC0D0: autoconfig: line_outs=1 (0x12/0x0/0x0/0x0/0x0) type:speaker > [ 44.101123] sound hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > [ 44.101127] sound hdaudioC0D0: hp_outs=1 (0x11/0x0/0x0/0x0/0x0) > [ 44.101130] sound hdaudioC0D0: mono: mono_out=0x0 > [ 44.101133] sound hdaudioC0D0: inputs: > [ 44.101137] sound hdaudioC0D0: Internal Mic=0x15 > [ 44.101140] sound hdaudioC0D0: Dock Mic=0x1c > [ 44.101144] sound hdaudioC0D0: Mic=0x14 > [ 44.112515] input: HDA Intel Dock Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 > [ 44.112657] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 > [ 44.112794] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 > [ 45.815258] init: failsafe main process (883) killed by TERM signal ---- ALSA info for latest daily upstream END ----

3 10

Re: [alsa-devel] Set eapd to follow stream on or off
by Kailang 04 Jun '14

04 Jun '14

Hi Takashi, I abandon this update. Playing a slient stream will be had pop noise. It is hardware issue. It is not to avoid. But it need to read and write our hidden register to control eapd. Resume back need to fill eapd to keep low state. I think I just send the patch to ubuntu team. Thanks. > > >> At Thu, 22 May 2014 14:26:04 +0800, >> Kailang wrote: >>> >>> >>> >>> > At Wed, 21 May 2014 17:24:16 +0800, >>> > Kailang wrote: >>> >> >>> >> > At Wed, 21 May 2014 14:28:12 +0800, >>> >> > Kailang wrote: >>> >> >> >>> >> >> Hi Takashi, >>> >> >> >>> >> >> Attach is the update patch. >>> >> > >>> >> > Keep the function in lower letters. >>> >> > >>> >> > Also, update the patch description as well (and better to put more >>> >> > words). >>> >> >>> >> Update as attach. >>> >> Thanks. >>> > >>> > Well, do you really want to turn off EAPD at closing the stream? >>> >>> Yes. >>> >>> > As I repeatedly explained, this results in more frequent EAPD on/off, >>> > i.e. more pop noises. >>> > >>> >>> I have sent the patch for HP. >>> I will wait for feedback from customer. >> >> Did you send the test script, too? > No. > >> >>> If it is solve the issue, I will advise you. >>> >>> Our AE says, windows use this method. HP is agree this usage. >> >> Windows does this likely because of possible power saving, not because >> of pop noises. >> >> I'm fine to take the patch, but only when I'm convinced why it really >> fixes the pop noise problem in the scenario below: >> - playing a silent stream >> - frequent open / close a stream >> > > You are right. It is the key points. But I think first stream will show > pop noise. It was hard to remove. > Keep EAPD to high after first stream start maybe a good idea. > Thanks. > I think to request the machine is better. > I will let our AE to request the machine. > >> >> Takashi >> >> >>> >>> > >>> > Takashi >>> > >>> >> >>> >> > >>> >> > >>> >> > thanks, >>> >> > >>> >> > Takashi >>> >> > >>> >> >> Many thanks. >>> >> >> >>> >> >> BR, >>> >> >> Kailang >>> >> >> >>> >> >> >>> >> >> > At Wed, 21 May 2014 13:58:14 +0800, >>> >> >> > Kailang wrote: >>> >> >> >> >>> >> >> >> >>> >> >> >> > At Tue, 20 May 2014 18:05:42 +0800, >>> >> >> >> > Kailang wrote: >>> >> >> >> >> >>> >> >> >> >> > At Tue, 20 May 2014 17:20:44 +0800, >>> >> >> >> >> > Kailang wrote: >>> >> >> >> >> >> >>> >> >> >> >> >> >>> >> >> >> >> >> > At Tue, 20 May 2014 16:52:43 +0800, >>> >> >> >> >> >> > Kailang wrote: >>> >> >> >> >> >> >> >>> >> >> >> >> >> >> >>> >> >> >> >> >> >> > At Tue, 20 May 2014 15:15:49 +0800, >>> >> >> >> >> >> >> > Kailang wrote: >>> >> >> >> >> >> >> >> >>> >> >> >> >> >> >> >> Hi Takashi, >>> >> >> >> >> >> >> >> >>> >> >> >> >> >> >> >> HP machine had pop noise via speaker. >>> >> >> >> >> >> >> >> If EAPD on or off, speaker will have pop noise. >>> >> >> >> >> >> >> >> So, they want to set eapd to follow with stream on >>> >> >> >> >> >> >> >> or >>> >> >> >> >> >> >> >> off. >>> >> >> >> >> >> >> > >>> >> >> >> >> >> >> > I can apply this as is, but still I wonder whether >>> >> >> >> >> >> >> > this >>> >> >> >> >> >> >> > results >>> >> >> >> >> >> >> > in >>> >> >> >> >> >> >> > other problems. Doing EAPD on/off there means that >>> >> >> >> >> >> >> > it >>> >> >> >> >> >> >> > happens >>> >> >> >> >> >> >> > far >>> >> >> >> >> >> >> > more often. For example, what happens if you do >>> >> >> >> >> >> >> > like >>> >> >> >> >> >> >> > below? >>> >> >> >> >> >> >> > >>> >> >> >> >> >> >> > while true; do >>> >> >> >> >> >> >> > echo -n > /dev/snd/pcmC0D0p >>> >> >> >> >> >> >> > done >>> >> >> >> >> >> >> > >>> >> >> >> >> >> >> > And, what if the device is resumed from power-saving >>> >> >> >> >> >> >> > by >>> >> >> >> >> >> >> > an >>> >> >> >> >> >> >> > audio >>> >> >> >> >> >> >> > playback? Immediately after the power up, it'll >>> >> >> >> >> >> >> > start >>> >> >> >> >> >> >> > the >>> >> >> >> >> >> >> > streaming, >>> >> >> >> >> >> >> > so it'll turn on EAPD. >>> >> >> >> >> >> >> > >>> >> >> >> >> >> >> > That said, isn't it just a need for some delay >>> >> >> >> >> >> >> > before >>> >> >> >> >> >> >> > EAPD >>> >> >> >> >> >> >> > on/off? >>> >> >> >> >> >> >> >>> >> >> >> >> >> >> No, eapd on or off will cause pop noise. >>> >> >> >> >> >> > >>> >> >> >> >> >> > Does it do always? >>> >> >> >> >> >> No, pop noise just one pop sound. It can not remove by >>> >> >> >> >> >> software. >>> >> >> >> >> >> But this machine need to do this always. >>> >> >> >> >> > >>> >> >> >> >> > So, doing EAPD on/off often is rather worse. It's what >>> >> >> >> >> > your >>> >> >> >> >> > patch >>> >> >> >> >> > does. >>> >> >> >> >> > >>> >> >> >> >> > Once again: did you try the script above? If the pop >>> >> >> >> >> > noise >>> >> >> >> >> > always >>> >> >> >> >> >>> >> >> >> >> I had try it on my develop machine. I have no HP issue >>> >> >> >> >> machine >>> >> >> >> >> on >>> >> >> >> >> my >>> >> >> >> >> site. >>> >> >> >> >> >>> >> >> >> >> > happens, the operation like above is the worst scenario. >>> >> >> >> >> > >>> >> >> >> >> > Then, let's think again how to *reduce* it. We need one >>> >> >> >> >> > EAPD >>> >> >> >> >> > on >>> >> >> >> >> > at >>> >> >> >> >> > least after each power up. But it's needed only when you >>> >> >> >> >> > actually >>> >> >> >> >> > plays a stream. That is, we need only enable EAPD at >>> >> >> >> >> > playback >>> >> >> >> >> > stream >>> >> >> >> >> > open, but leave EAPD as is at close. >>> >> >> >> >> > >>> >> >> >> >> >>> >> >> >> >> [we need only enable EAPD at playback stream open, but leave >>> >> >> >> >> EAPD >>> >> >> >> >> as >>> >> >> >> >> is >>> >> >> >> >> at >>> >> >> >> >> close.] >>> >> >> >> >> Yes. >>> >> >> >> >> So, I need to create eapd low in alc269_shutup. Right? >>> >> >> >> > >>> >> >> >> > Better to create own shutup callback (you can call >>> >> >> >> > alc269_shutup() >>> >> >> >> > from there, too) and set it to spec->shutup in the fixup. >>> >> >> >> > The >>> >> >> >> > pincfg >>> >> >> >> > cleanup isn't often needed or may have bad effect. You >>> >> >> >> > should >>> >> >> >> > test >>> >> >> >> > with real machines what are needed and what are not. >>> >> >> >> >>> >> >> >> static void alc290_fixup_pop_noise(struct hda_codec *codec, >>> >> >> >> const struct hda_fixup *fix, int action) >>> >> >> >> { >>> >> >> >> struct alc_spec *spec = codec->spec; >>> >> >> >> >>> >> >> >> if (action == HDA_FIXUP_ACT_PRE_PROBE) { >>> >> >> >> spec->gen.pcm_playback_hook = ALC290_playback_pcm_hook; >>> >> >> >> } >>> >> >> >> if (action == HDA_FIXUP_ACT_INIT) { >>> >> >> >> set_eapd(codec, 0x14, false); >>> >> >> >> } >>> >> >> >> } >>> >> >> >> >>> >> >> >> Could I modify as upper code? >>> >> >> >> To set eapd low in HDA_FIXUP_ACT_INIT state. >>> >> >> >> If system resume, it will initial eapd to low. >>> >> >> >> To write eapd to same value, it will have no pop noise. >>> >> >> >> It will keep always low for eapd, only stream open and close >>> >> >> >> have >>> >> >> >> change >>> >> >> >> state for eapd. >>> >> >> > >>> >> >> > Yes, it looks OK. >>> >> >> > >>> >> >> > >>> >> >> > Takashi >>> >> >> > >>> >> >> > >>> >> >> >> >>> >> >> >> > >>> >> >> >> > >>> >> >> >> > Takashi >>> >> >> >> > >>> >> >> >> >> >>> >> >> >> >> > >>> >> >> >> >> > Takashi >>> >> >> >> >> > >>> >> >> >> >> >> >> Our customer just want to do eapd on when stream is >>> >> >> >> >> >> >> open. >>> >> >> >> >> >> >> The pop noise will not clear. Because music gain was >>> >> >> >> >> >> >> biger >>> >> >> >> >> >> >> than >>> >> >> >> >> >> >> pop >>> >> >> >> >> >> >> noise. >>> >> >> >> >> >> > >>> >> >> >> >> >> > Opening a stream doesn't mean that it follows the loud >>> >> >> >> >> >> > sound >>> >> >> >> >> >> > immediately. The script above wouldn't produce any >>> >> >> >> >> >> > sound >>> >> >> >> >> >> > but >>> >> >> >> >> >> > just >>> >> >> >> >> >> > does open/close the stream. Did you try that? >>> >> >> >> >> >> > >>> >> >> >> >> >> > >>> >> >> >> >> >> > Takashi >>> >> >> >> >> >> > >>> >> >> >> >> >> > ------Please consider the environment before printing >>> >> >> >> >> >> > this >>> >> >> >> >> >> > e-mail. >>> >> >> >> >> >> >>> >> >> >> >> >>> >> >> >> >>> >> >> [2 0001-set-eapd-follow-stream-on-off.patch >>> >> >> <application/octet-stream >>> >> >> (quoted-printable)>] >>> >> >> >>> >> > >>> >> > ------Please consider the environment before printing this e-mail. >>> >> [2 0002-set-eapd-follow-stream-on-off.patch <application/octet-stream >>> >> (quoted-printable)>] >>> >> >>> >

1 0

[alsa-devel] [PATCH 1/1] Alsa: seq: correctly detect input buffer overflow
by Adam Goode 04 Jun '14

04 Jun '14

snd_seq_event_dup returns -ENOMEM in some buffer-full conditions, but usually returns -EAGAIN. Make -EAGAIN trigger the overflow condition in snd_seq_fifo_event_in so that the fifo is cleared and -ENOSPC is returned to userspace as stated in the alsa-lib docs. Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index 5599899..53a403e 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -124,7 +124,7 @@ int snd_seq_fifo_event_in(struct snd_seq_fifo *f, snd_use_lock_use(&f->use_lock); err = snd_seq_event_dup(f->pool, event, &cell, 1, NULL); /* always non-blocking */ if (err < 0) { - if (err == -ENOMEM) + if ((err == -ENOMEM) || (err == -EAGAIN)) atomic_inc(&f->overflow); snd_use_lock_free(&f->use_lock); return err; -- 1.9.1.423.g4596e3a

2 1

[alsa-devel] [PATCH - seq 1/1] ALSA: seq: Continue broadcasting events to ports if one of them fails
by Adam Goode 04 Jun '14

04 Jun '14

Sometimes PORT_EXIT messages are lost when a process is exiting. This happens if you subscribe to the announce port with client A, then subscribe to the announce port with client B, then kill client A. Client B will not see the PORT_EXIT message because client A's port is closing and is earlier in the announce port subscription list. The for each loop will try to send the announcement to client A and fail, then will stop trying to broadcast to other ports. Killing B works fine since the announcement will already have gone to A. The CLIENT_EXIT message does not get lost. How to reproduce problem: *** termA $ aseqdump -p 0:1 0:1 Port subscribed 0:1 -> 128:0 *** termB $ aseqdump -p 0:1 *** termA 0:1 Client start client 129 0:1 Port start 129:0 0:1 Port subscribed 0:1 -> 129:0 *** termB 0:1 Port subscribed 0:1 -> 129:0 *** termA ^C *** termB 0:1 Client exit client 128 <--- expected Port exit as well (before client exit) Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 9ca5e64..f9b5315 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -660,7 +660,7 @@ static int deliver_to_subscribers(struct snd_seq_client *client, int atomic, int hop) { struct snd_seq_subscribers *subs; - int err = 0, num_ev = 0; + int err = 0, sticky_err = 0, num_ev = 0; struct snd_seq_event event_saved; struct snd_seq_client_port *src_port; struct snd_seq_port_subs_info *grp; @@ -685,8 +685,12 @@ static int deliver_to_subscribers(struct snd_seq_client *client, subs->info.flags & SNDRV_SEQ_PORT_SUBS_TIME_REAL); err = snd_seq_deliver_single_event(client, event, 0, atomic, hop); - if (err < 0) - break; + if (err < 0) { + /* save first error that occurs and continue */ + if (!sticky_err) + sticky_err = err; + continue; + } num_ev++; /* restore original event record */ *event = event_saved; @@ -697,7 +701,7 @@ static int deliver_to_subscribers(struct snd_seq_client *client, up_read(&grp->list_mutex); *event = event_saved; /* restore */ snd_seq_port_unlock(src_port); - return (err < 0) ? err : num_ev; + return (sticky_err < 0) ? sticky_err : num_ev; } @@ -709,7 +713,7 @@ static int port_broadcast_event(struct snd_seq_client *client, struct snd_seq_event *event, int atomic, int hop) { - int num_ev = 0, err = 0; + int num_ev = 0, err = 0, sticky_err = 0; struct snd_seq_client *dest_client; struct snd_seq_client_port *port; @@ -724,14 +728,18 @@ static int port_broadcast_event(struct snd_seq_client *client, err = snd_seq_deliver_single_event(NULL, event, SNDRV_SEQ_FILTER_BROADCAST, atomic, hop); - if (err < 0) - break; + if (err < 0) { + /* save first error that occurs and continue */ + if (!sticky_err) + sticky_err = err; + continue; + } num_ev++; } read_unlock(&dest_client->ports_lock); snd_seq_client_unlock(dest_client); event->dest.port = SNDRV_SEQ_ADDRESS_BROADCAST; /* restore */ - return (err < 0) ? err : num_ev; + return (sticky_err < 0) ? sticky_err : num_ev; } /* @@ -741,7 +749,7 @@ static int port_broadcast_event(struct snd_seq_client *client, static int broadcast_event(struct snd_seq_client *client, struct snd_seq_event *event, int atomic, int hop) { - int err = 0, num_ev = 0; + int err = 0, sticky_err = 0, num_ev = 0; int dest; struct snd_seq_addr addr; @@ -760,12 +768,16 @@ static int broadcast_event(struct snd_seq_client *client, err = snd_seq_deliver_single_event(NULL, event, SNDRV_SEQ_FILTER_BROADCAST, atomic, hop); - if (err < 0) - break; + if (err < 0) { + /* save first error that occurs and continue */ + if (!sticky_err) + sticky_err = err; + continue; + } num_ev += err; } event->dest = addr; /* restore */ - return (err < 0) ? err : num_ev; + return (sticky_err < 0) ? sticky_err : num_ev; } -- 1.9.1.423.g4596e3a

2 2

[alsa-devel] [GIT PULL] ASoC updates for v3.16
by Mark Brown 03 Jun '14

03 Jun '14

The following changes since commit cee429e5c53f7ffc1dd500001d9dff567833c23d: Merge remote-tracking branches 'asoc/topic/ux500', 'asoc/topic/wm8731', 'asoc/topic/wm8804', 'asoc/topic/wm8955' and 'asoc/topic/wm8985' into asoc-next (2014-05-22 00:24:04 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-v3.16-2 for you to fetch changes up to e1d4d3c854f25cff6c6c139588570e124d5e8fa4: ASoC: free jack GPIOs before the sound card is freed (2014-06-03 10:41:16 +0100) ---------------------------------------------------------------- ASoC: Final updates for v3.16 A few more updates from the last week of development, nothing too exciting. Highlights include: - GPIO descriptor support for jacks - More updates and fixes to the Freescale SSI, Intel and rsnd drivers. - New drivers for Analog Devices ADAU1361, ADAU1381, ADAU1761 and ADAU1781, and Realtek RT5677. ---------------------------------------------------------------- Alexander Shiyan (1): ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320 Andrew Lunn (1): ASoC: simple-card: Support setting mclk via a fixed factor Andy Shevchenko (1): ASoC: Intel: remove duplicate headers Axel Lin (2): ASoC: sirf-audio-codec: Simplify the new bitmask value in regmap_update_bits ASoC: cs42l56: Fix new value argument in snd_soc_update_bits calls Charles Keepax (1): ASoC: wm_adsp: Use adsp_err/warn instead of dev_err/warn Chen Zhen (1): ASoC: max98090: Add NI/MI values for user pclk 19.2 MHz Daniel Matuschek (1): ASoC: wm8804: Allow control of master clock divider in PLL generation Fabio Estevam (2): ASoC: fsl_ssi: Add suspend/resume support ASoC: sgtl5000: Fix the cache handling Imre Deak (1): ASoC: Intel: byt/hsw: Add missing kthread_stop to error/cleanup path Jarkko Nikula (8): ASoC: jack: Basic GPIO descriptor conversion ASoC: jack: Add support for GPIO descriptor defined jack pins ASoC: jack: Clarify GPIO descriptor lookup in struct snd_soc_jack_gpio doc ASoC: jack: Fix multiple definition of `snd_soc_jack_add_gpiods' ASoC: Intel: Add Baytrail byt-max98090 machine driver ASoC: Intel: Clear stored Baytrail DSP DMA pointer before stream start ASoC: Intel: Use devm_snd_soc_register_card ASoC: Intel: byt-rt5640: Use card PM ops from core Jyri Sarha (2): ASoC: simple-card: is_top_level_node parameter to simple_card_dai_link_of() ASoC: omap-pcm: Move omap-pcm under include/sound Kees Cook (1): ASoC: Intel: avoid format string leak to thread name Kuninori Morimoto (6): ASoC: rsnd: DT node clean up by using the of_node_put() ASoC: rsnd: save platform_device instead of device ASoC: rsnd: remove rsnd_src_non_ops ASoC: rsnd: module name is unified ASoC: rsnd: care DMA slave channel name for DT ASoC: rsnd: add rsnd_gen_dma_addr() for DMAC addr Lars-Peter Clausen (7): ASoC: davinci-evm: Replace instances of rtd->codec->card with rtd->card ASoC: Add ADAU1X61 and ADAU1X81 CODECs common code ASoC: Add ADAU1361/ADAU1761 audio CODEC support ASoC: Add ADAU1381/ADAU1781 audio CODEC support ASoC: Blackfin: ADAU1X61 eval board support ASoC: Blackfin: ADAU1X81 eval board support ASoC: Add helper functions to cast from DAPM context to CODEC/platform Mark Brown (19): Merge branch 'topic/davinci' of git://git.kernel.org/.../broonie/sound into asoc-omap Merge branch 'topic/fsl' of git://git.kernel.org/.../broonie/sound into asoc-fsl-ssi Merge branches 'topic/rt5640', 'topic/rt5645' and 'topic/rt5651' of git://git.kernel.org/.../broonie/sound into asoc-rl6231 ASoC: intel: The Baytrail/MAX98090 driver depends on I2C ASoC: cache: Fix error code when not using ASoC level cache Merge remote-tracking branches 'asoc/fix/pxa' and 'asoc/fix/tlv320aic3x' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branches 'asoc/topic/adau' and 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/alc5623' into asoc-next Merge remote-tracking branch 'asoc/topic/cs42l56' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl-ssi' into asoc-next Merge remote-tracking branches 'asoc/topic/gpio' and 'asoc/topic/intel' into asoc-next Merge remote-tracking branches 'asoc/topic/max98090' and 'asoc/topic/max98095' into asoc-next Merge remote-tracking branches 'asoc/topic/omap' and 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/rl6231' and 'asoc/topic/rt5677' into asoc-next Merge remote-tracking branches 'asoc/topic/samsung', 'asoc/topic/sgtl5000', 'asoc/topic/simple' and 'asoc/topic/sirf' into asoc-next Merge remote-tracking branch 'asoc/topic/tegra' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8804' and 'asoc/topic/wm9713' into asoc-next Markus Pargmann (4): ASoC: fsl-ssi: make fsl,mode property optional ASoC: fsl-ssi: Fix baudclock handling ASoC: fsl-ssi: reorder and document fsl_ssi_private ASoC: fsl-ssi: Use regmap Matt Reimer (1): ASoC: wm9713: correct mono out PGA sources Oder Chiou (4): ASoC: rt5640: Add RL6231 class device shared support for RT5640, RT5645 and RT5651 ASoC: rt5640: Add the function of the PLL clock calculation to RL6231 shared support ASoC: rt5640: Add the function "get_clk_info" to RL6231 shared support ASoC: add RT5677 CODEC driver Peter Ujfalusi (1): ASoC: tlv320aci3x: Fix custom snd_soc_dapm_put_volsw_aic3x() function Sascha Hauer (4): ASoC: fsl-ssi: introduce SoC specific data ASoC: fsl-ssi: set bitclock in master mode from hw_params ASoC: fsl-ssi: remove unnecessary spinlock ASoC: fsl-ssi: Set framerate divider correctly for i2s master mode Stephen Boyd (1): ASoC: pxa: pxa-ssp: Terminate of match table Stephen Warren (2): ASoC: tegra: free jack GPIOs before the sound card is freed ASoC: free jack GPIOs before the sound card is freed Sylwester Nawrocki (1): ASoC: samsung: Handle errors when getting the op_clk clock Takashi Iwai (1): ASoC: alc5623: Fix Kconfig dependency Tushar Behera (3): ASoC: max98090: Add master clock handling ASoC: max98095: Add master clock handling ASoC: samsung: Use params_width() Xiubo Li (1): ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error. .../devicetree/bindings/sound/max98090.txt | 6 + .../devicetree/bindings/sound/max98095.txt | 6 + .../devicetree/bindings/sound/renesas,rsnd.txt | 1 + .../devicetree/bindings/sound/simple-card.txt | 3 + include/linux/platform_data/adau17x1.h | 109 + {sound/soc/omap => include/sound}/omap-pcm.h | 0 include/sound/rt5677.h | 21 + include/sound/soc.h | 49 +- sound/soc/blackfin/Kconfig | 26 + sound/soc/blackfin/Makefile | 4 + sound/soc/blackfin/bfin-eval-adau1x61.c | 142 + sound/soc/blackfin/bfin-eval-adau1x81.c | 130 + sound/soc/codecs/Kconfig | 52 +- sound/soc/codecs/Makefile | 22 +- sound/soc/codecs/adau1761-i2c.c | 60 + sound/soc/codecs/adau1761-spi.c | 77 + sound/soc/codecs/adau1761.c | 803 +++++ sound/soc/codecs/adau1761.h | 23 + sound/soc/codecs/adau1781-i2c.c | 58 + sound/soc/codecs/adau1781-spi.c | 75 + sound/soc/codecs/adau1781.c | 511 +++ sound/soc/codecs/adau1781.h | 23 + sound/soc/codecs/adau17x1.c | 866 +++++ sound/soc/codecs/adau17x1.h | 124 + sound/soc/codecs/cs42l56.c | 70 +- sound/soc/codecs/cs42l56.h | 10 +- sound/soc/codecs/max98090.c | 31 +- sound/soc/codecs/max98090.h | 1 + sound/soc/codecs/max98095.c | 24 + sound/soc/codecs/rl6231.c | 152 + sound/soc/codecs/rl6231.h | 34 + sound/soc/codecs/rt5640.c | 108 +- sound/soc/codecs/rt5640.h | 8 - sound/soc/codecs/rt5645.c | 111 +- sound/soc/codecs/rt5645.h | 7 - sound/soc/codecs/rt5651.c | 107 +- sound/soc/codecs/rt5651.h | 1 - sound/soc/codecs/rt5677.c | 3498 ++++++++++++++++++++ sound/soc/codecs/rt5677.h | 1451 ++++++++ sound/soc/codecs/sgtl5000.c | 75 +- sound/soc/codecs/sirf-audio-codec.c | 8 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/wm8804.c | 17 +- sound/soc/codecs/wm8804.h | 4 + sound/soc/codecs/wm9713.c | 3 +- sound/soc/codecs/wm_adsp.c | 37 +- sound/soc/davinci/davinci-evm.c | 9 +- sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/fsl/Kconfig | 8 +- sound/soc/fsl/fsl_ssi.c | 576 ++-- sound/soc/fsl/fsl_ssi.h | 50 +- sound/soc/generic/simple-card.c | 39 +- sound/soc/intel/Kconfig | 9 + sound/soc/intel/Makefile | 2 + sound/soc/intel/byt-max98090.c | 203 ++ sound/soc/intel/byt-rt5640.c | 27 +- sound/soc/intel/haswell.c | 9 +- sound/soc/intel/sst-acpi.c | 1 + sound/soc/intel/sst-baytrail-ipc.c | 8 +- sound/soc/intel/sst-baytrail-pcm.c | 1 + sound/soc/intel/sst-haswell-ipc.c | 15 +- sound/soc/intel/sst-haswell-pcm.c | 1 - sound/soc/omap/ams-delta.c | 14 +- sound/soc/omap/omap-dmic.c | 2 +- sound/soc/omap/omap-hdmi.c | 2 +- sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/omap/omap-mcpdm.c | 2 +- sound/soc/omap/omap-twl4030.c | 28 +- sound/soc/omap/rx51.c | 18 +- sound/soc/pxa/hx4700.c | 9 +- sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/samsung/h1940_uda1380.c | 11 +- sound/soc/samsung/i2s.c | 14 +- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/rx1950_uda1380.c | 12 +- sound/soc/samsung/s3c-i2s-v2.c | 8 +- sound/soc/samsung/s3c2412-i2s.c | 6 +- sound/soc/samsung/s3c24xx-i2s.c | 6 +- sound/soc/samsung/smartq_wm8987.c | 11 +- sound/soc/samsung/smdk_wm8580.c | 8 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/spdif.c | 4 +- sound/soc/sh/rcar/core.c | 87 +- sound/soc/sh/rcar/dvc.c | 20 +- sound/soc/sh/rcar/gen.c | 95 + sound/soc/sh/rcar/rsnd.h | 10 +- sound/soc/sh/rcar/src.c | 50 +- sound/soc/sh/rcar/ssi.c | 28 +- sound/soc/soc-cache.c | 3 + sound/soc/soc-jack.c | 88 +- sound/soc/tegra/tegra_alc5632.c | 16 +- sound/soc/tegra/tegra_max98090.c | 16 +- sound/soc/tegra/tegra_rt5640.c | 16 +- sound/soc/tegra/tegra_wm8903.c | 11 +- 94 files changed, 9610 insertions(+), 904 deletions(-) create mode 100644 include/linux/platform_data/adau17x1.h rename {sound/soc/omap => include/sound}/omap-pcm.h (100%) create mode 100644 include/sound/rt5677.h create mode 100644 sound/soc/blackfin/bfin-eval-adau1x61.c create mode 100644 sound/soc/blackfin/bfin-eval-adau1x81.c create mode 100644 sound/soc/codecs/adau1761-i2c.c create mode 100644 sound/soc/codecs/adau1761-spi.c create mode 100644 sound/soc/codecs/adau1761.c create mode 100644 sound/soc/codecs/adau1761.h create mode 100644 sound/soc/codecs/adau1781-i2c.c create mode 100644 sound/soc/codecs/adau1781-spi.c create mode 100644 sound/soc/codecs/adau1781.c create mode 100644 sound/soc/codecs/adau1781.h create mode 100644 sound/soc/codecs/adau17x1.c create mode 100644 sound/soc/codecs/adau17x1.h create mode 100644 sound/soc/codecs/rl6231.c create mode 100644 sound/soc/codecs/rl6231.h create mode 100644 sound/soc/codecs/rt5677.c create mode 100644 sound/soc/codecs/rt5677.h create mode 100644 sound/soc/intel/byt-max98090.c

3 7

[alsa-devel] gitcompile fails
by Mark Brand 03 Jun '14

03 Jun '14

Fresh git clones of alsa-driver and alsa-kmirror produce a ./gitcompile failure for me: ln -s ../../alsa-kernel/include/pcm-indirect.h cp ../../alsa-kernel/include/pcm.h . patch -p0 -i pcm.patch pcm.h patching file pcm.h Hunk #1 FAILED at 280. Hunk #2 succeeded at 291 (offset -3 lines). [snip] Hunk #11 succeeded at 566 (offset -4 lines). 1 out of 11 hunks FAILED -- saving rejects to file pcm.h.rej make[3]: *** [pcm.h] Error 1 make[3]: Leaving directory `/home/mark/workspace/alsa/alsa-driver/alsa/include/sound' Since I've seen this before (http://mailman.alsa-project.org/pipermail/alsa-devel/2012-July/053419.html), would I be forgiven for assuming that this is once again as a result of a mismatch between alsa-driver and alsa-kmirror? Any suggestion will be much appreciated, thank you. Mark Brand.

1 0

[alsa-devel] [PATCH] ASoC: free jack GPIOs before the sound card is freed
by Stephen Warren 03 Jun '14

03 Jun '14

From: Stephen Warren <swarren(a)nvidia.com> This is the same change as commit fb6b8e71448a "ASoC: tegra: free jack GPIOs before the sound card is freed", but applied to all other ASoC machine drivers where code inspection indicates the same problem exists. That commit's description is: ========== snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to generate an initial jack status report. If sound card initialization fails, that work item needs to be cancelled, so it doesn't run after the card has been freed. Specifically, freeing the card calls snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which is called from the work queue item. snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine drivers do call this function in the platform driver remove() callback. However, this happens after the sound card is freed, at least when the card is freed due to errors late during snd_soc_instantiate_card(). This leaves a window where the work item can execute after the card is freed. In next-20140522, sound card initialization does fail for unrelated reasons, and hits the problem described above. To solve this, fix the Tegra ASoC machine drivers to clean up the Jack GPIOs during the snd_soc_card's .remove() callback, which is executed before the overall card object is freed. also, guard the cleanup call based on whether we actually setup up the GPIOs in the first place. Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove function to match where the GPIOs get set up. However, there is no such callback. ========== Note that I have not even compile-tested this in most cases, since most of the drivers rely on specific mach-* support I don't have enabled, and don't support COMPILE_TEST. Testing by the relevant board maintainers would be useful. Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> (maintainer:OMAP AUDIO SUPPORT) Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> (maintainer:OMAP AUDIO SUPPORT) Cc: Philipp Zabel <philipp.zabel(a)gmail.com> (maintainer:ARM/H4700 (HP IPA...) Cc: Paul Parsons <lost.distance(a)yahoo.com> (maintainer:ARM/H4700 (HP IPA...) Cc: Eric Miao <eric.y.miao(a)gmail.com> (maintainer:PXA2xx/PXA3xx SUP...) Cc: Russell King <linux(a)arm.linux.org.uk> (maintainer:PXA2xx/PXA3xx SUP...) Cc: Haojian Zhuang <haojian.zhuang(a)gmail.com> (maintainer:PXA2xx/PXA3xx SUP...) Cc: Ben Dooks <ben-linux(a)fluff.org> (maintainer:ARM/SAMSUNG ARM A...) Cc: Kukjin Kim <kgene.kim(a)samsung.com> (maintainer:ARM/SAMSUNG ARM A...) Cc: Sangbeom Kim <sbkim73(a)samsung.com> (supporter:SAMSUNG AUDIO (AS...) Cc: linux-omap(a)vger.kernel.org (open list:OMAP AUDIO SUPPORT) Cc: linux-samsung-soc(a)vger.kernel.org (moderated list:ARM/SAMSUNG ARM A...) Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Mark, do you need this patch split up by architecture for topic branches? I assume not if you've already sent your pull requests. This applies fine on top of tag asoc-v3.16. --- sound/soc/omap/ams-delta.c | 14 ++++++++++---- sound/soc/omap/omap-twl4030.c | 28 ++++++++++++++-------------- sound/soc/omap/rx51.c | 18 +++++++++--------- sound/soc/pxa/hx4700.c | 9 ++++++++- sound/soc/samsung/h1940_uda1380.c | 11 +++++++++-- sound/soc/samsung/rx1950_uda1380.c | 12 ++++++++++-- sound/soc/samsung/smartq_wm8987.c | 11 +++++++++-- 7 files changed, 69 insertions(+), 34 deletions(-) diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index bb243c663e6b..1f41951d8b7f 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -527,6 +527,15 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int ams_delta_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_gpios), + ams_delta_hook_switch_gpios); + + return 0; +} + /* DAI glue - connects codec <--> CPU */ static struct snd_soc_dai_link ams_delta_dai_link = { .name = "CX20442", @@ -543,6 +552,7 @@ static struct snd_soc_dai_link ams_delta_dai_link = { static struct snd_soc_card ams_delta_audio_card = { .name = "AMS_DELTA", .owner = THIS_MODULE, + .remove = ams_delta_card_remove, .dai_link = &ams_delta_dai_link, .num_links = 1, @@ -579,10 +589,6 @@ static int ams_delta_remove(struct platform_device *pdev) dev_warn(&pdev->dev, "failed to unregister V253 line discipline\n"); - snd_soc_jack_free_gpios(&ams_delta_hook_switch, - ARRAY_SIZE(ams_delta_hook_switch_gpios), - ams_delta_hook_switch_gpios); - snd_soc_unregister_card(card); card->dev = NULL; return 0; diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index 64141db311b2..b4e282871658 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -231,6 +231,19 @@ static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd) return ret; } +static int omap_twl4030_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card); + + if (priv->jack_detect > 0) + snd_soc_jack_free_gpios(&priv->hs_jack, + ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + + return 0; +} + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link omap_twl4030_dai_links[] = { { @@ -258,6 +271,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = { /* Audio machine driver */ static struct snd_soc_card omap_twl4030_card = { .owner = THIS_MODULE, + .remove = omap_twl4030_card_remove, .dai_link = omap_twl4030_dai_links, .num_links = ARRAY_SIZE(omap_twl4030_dai_links), @@ -353,19 +367,6 @@ static int omap_twl4030_probe(struct platform_device *pdev) return 0; } -static int omap_twl4030_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card); - - if (priv->jack_detect > 0) - snd_soc_jack_free_gpios(&priv->hs_jack, - ARRAY_SIZE(hs_jack_gpios), - hs_jack_gpios); - - return 0; -} - static const struct of_device_id omap_twl4030_of_match[] = { {.compatible = "ti,omap-twl4030", }, { }, @@ -380,7 +381,6 @@ static struct platform_driver omap_twl4030_driver = { .of_match_table = omap_twl4030_of_match, }, .probe = omap_twl4030_probe, - .remove = omap_twl4030_remove, }; module_platform_driver(omap_twl4030_driver); diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 6951dc812055..47a10290535b 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -334,6 +334,14 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) return err; } +static int rx51_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios), + rx51_av_jack_gpios); + + return 0; +} + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link rx51_dai[] = { { @@ -368,6 +376,7 @@ static struct snd_soc_codec_conf rx51_codec_conf[] = { static struct snd_soc_card rx51_sound_card = { .name = "RX-51", .owner = THIS_MODULE, + .remove = rx51_card_remove, .dai_link = rx51_dai, .num_links = ARRAY_SIZE(rx51_dai), .aux_dev = rx51_aux_dev, @@ -499,14 +508,6 @@ static int rx51_soc_probe(struct platform_device *pdev) return 0; } -static int rx51_soc_remove(struct platform_device *pdev) -{ - snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios), - rx51_av_jack_gpios); - - return 0; -} - #if defined(CONFIG_OF) static const struct of_device_id rx51_audio_of_match[] = { { .compatible = "nokia,n900-audio", }, @@ -522,7 +523,6 @@ static struct platform_driver rx51_soc_driver = { .of_match_table = of_match_ptr(rx51_audio_of_match), }, .probe = rx51_soc_probe, - .remove = rx51_soc_remove, }; module_platform_driver(rx51_soc_driver); diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index dcc9b04bd92c..6b81acaffddd 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -152,6 +152,13 @@ static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd) return err; } +static int hx4700_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&hs_jack, 1, &hs_jack_gpio); + + return 0; +} + /* hx4700 digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link hx4700_dai = { .name = "ak4641", @@ -170,6 +177,7 @@ static struct snd_soc_dai_link hx4700_dai = { static struct snd_soc_card snd_soc_card_hx4700 = { .name = "iPAQ hx4700", .owner = THIS_MODULE, + .remove = hx4700_card_remove, .dai_link = &hx4700_dai, .num_links = 1, .dapm_widgets = hx4700_dapm_widgets, @@ -206,7 +214,6 @@ static int hx4700_audio_probe(struct platform_device *pdev) static int hx4700_audio_remove(struct platform_device *pdev) { - snd_soc_jack_free_gpios(&hs_jack, 1, &hs_jack_gpio); snd_soc_unregister_card(&snd_soc_card_hx4700); gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0); diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 9f2fb69dbaae..720357f11a7f 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -189,6 +189,14 @@ static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int h1940_uda1380_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + + return 0; +} + /* s3c24xx digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link h1940_uda1380_dai[] = { { @@ -206,6 +214,7 @@ static struct snd_soc_dai_link h1940_uda1380_dai[] = { static struct snd_soc_card h1940_asoc = { .name = "h1940", .owner = THIS_MODULE, + .remove = h1940_uda1380_card_remove, .dai_link = h1940_uda1380_dai, .num_links = ARRAY_SIZE(h1940_uda1380_dai), @@ -257,8 +266,6 @@ err_out: static void __exit h1940_exit(void) { platform_device_unregister(s3c24xx_snd_device); - snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), - hp_jack_gpios); gpio_free(S3C_GPIO_END + 9); } diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 5b3e504d3a32..192aa9fc102f 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -31,6 +31,7 @@ #include "s3c24xx-i2s.h" static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd); +static int rx1950_uda1380_card_remove(struct snd_soc_pcm_runtime *rtd); static int rx1950_startup(struct snd_pcm_substream *substream); static int rx1950_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); @@ -116,6 +117,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static struct snd_soc_card rx1950_asoc = { .name = "rx1950", .owner = THIS_MODULE, + .remove = rx1950_uda1380_card_remove, .dai_link = rx1950_uda1380_dai, .num_links = ARRAY_SIZE(rx1950_uda1380_dai), @@ -234,6 +236,14 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int rx1950_uda1380_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + + return 0; +} + static int __init rx1950_init(void) { int ret; @@ -278,8 +288,6 @@ err_gpio: static void __exit rx1950_exit(void) { platform_device_unregister(s3c24xx_snd_device); - snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), - hp_jack_gpios); gpio_free(S3C2410_GPA(1)); } diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index df55db5d3554..271a904277a1 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -182,6 +182,14 @@ static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd) return err; } +static int smartq_wm8987_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&smartq_jack, ARRAY_SIZE(smartq_jack_gpios), + smartq_jack_gpios); + + return 0; +} + static struct snd_soc_dai_link smartq_dai[] = { { .name = "wm8987", @@ -198,6 +206,7 @@ static struct snd_soc_dai_link smartq_dai[] = { static struct snd_soc_card snd_soc_smartq = { .name = "SmartQ", .owner = THIS_MODULE, + .remove = smartq_wm8987_card_remove, .dai_link = smartq_dai, .num_links = ARRAY_SIZE(smartq_dai), @@ -259,8 +268,6 @@ err_unregister_device: static void __exit smartq_exit(void) { gpio_free(S3C64XX_GPK(12)); - snd_soc_jack_free_gpios(&smartq_jack, ARRAY_SIZE(smartq_jack_gpios), - smartq_jack_gpios); platform_device_unregister(smartq_snd_device); } -- 1.8.1.5

2 1

[alsa-devel] [RFC] set up an sync channel between audio and display driver (i.e. ALSA and DRM)
by Lin, Mengdong 03 Jun '14

03 Jun '14

This RFC is based on previous discussion to set up a generic communication channel between display and audio driver and an internal design of Intel MCG/VPG HDMI audio driver. It's still an initial draft and your advice would be appreciated to improve the design. The basic idea is to create a new avsink module and let both drm and alsa depend on it. This new module provides a framework and APIs for synchronization between the display and audio driver. 1. Display/Audio Client The avsink core provides APIs to create, register and lookup a display/audio client. A specific display driver (eg. i915) or audio driver (eg. HD-Audio driver) can create a client, add some resources objects (shared power wells, display outputs, and audio inputs, register ops) to the client, and then register this client to avisink core. The peer driver can look up a registered client by a name or type, or both. If a client gives a valid peer client name on registration, avsink core will bind the two clients as peer for each other. And we expect a display client and an audio client to be peers for each other in a system. int avsink_new_client ( const char *name, int type, /* client type, display or audio */ struct module *module, void *context, const char *peer_name, struct avsink_client **client_ret); int avsink_free_client (struct avsink_client *client); int avsink_register_client(struct avsink_client *client); int avisink_unregister_client(int client_handle); struct avsink_client *avsink_lookup_client(const char *name, int type); struct avsink_client { const char *name; /* client name */ int type; /* client type*/ void *context; struct module *module; /* top-level module for locking */ struct avsink_client *peer; /* peer client */ /* shared power wells */ struct avsink_power_well *power_well; int num_power_wells; /* endpoints, display outputs or audio inputs */ struct avsink_endpoint * endpoint; int num_endpints; struct avsink_registers_ops *reg_ops; /* ops to access registers of a client */ void *private_data; ... }; On system boots, the avsink module is loaded before the display and audio driver module. And the display and audio driver may be loaded on parallel. * If a specific display driver (eg. i915) supports avsink, it can create a display client, add power wells and display outputs to the client, and then register the display client to the avsink core. Then it may look up if there is any audio client registered, by name or type, and may find an audio client registered by some audio driver. * If an audio driver supports avsink, it usually should look up a registered display client by name or type at first, because it may need the shared power well in GPU and check the display outputs' name to bind the audio inputs. If the display client is not registered yet, the audio driver can choose to wait (maybe in a work queue) or return -EAGAIN for a deferred probe. After the display client is found, the audio driver can register an audio client with the display client's name as the peer name, the avsink core will bind the display and audio clients to each other. Open question: If the display or audio driver is disabled by the black list, shall we introduce a time out to avoid waiting for the other client registered endlessly? 2. Shared power wells (optional) The audio and display devices, maybe only part of them, may share a common power well (e.g. for Intel Haswell and Broadwell). If so, the driver that controls the power well should define a power well object, implement the get/put ops, and add it to its avsink client before registering the client to avsink core. Then the peer client can look up this power well by its name, and get/put this power well as a user. A client can have multiple power well objects. struct avsink_power_well { const char *name; /* name of the power well */ void *context; /* parameter of get/put ops, maybe device pointer for this power well */ struct avsink_power_well_ops *ops }; struct avsink_power_well_ops { int (*get)(void *context); int (*put)(void *context); }; API: int avsink_new_power(struct avsink_client *client, const char *power_name, void * power_context, struct avsink_power_well_ops *ops, struct avsink_power_well **power_ret); struct avsink_power_well *avisnk_lookup_power(const char *name); int avsink_get_power(struct avsink_power_well *power); /* Reqesut the power */ int avsink_put_power(struct avsink_power_well *power); /* Release the power */ For example, the i915 display driver can create a device for the shared power well in Haswell GPU, implement its PM functions, and use the device pointer as the context when creating the power well object, like below struct avsink_power_well_ops i915_power_well_ops = { .get = pm_runtime_get_sync; .put = pm_runtime_put_sync; }; ... avsink_new_power ( display_client, "i915_display_power_well", pdev, /* pointer of the power well device */ &i915_power_well_ops, ...) Power domain is not used here since a single device seems enough to represent a power well. 3. Display output and audio input endpoints A display client should register the display output endpoints and its audio peer client should register the audio input endpoints. A client can have multiple endpoints. The avsink core will bind an audio input and a display output as peer to each other. This is to allow the audio and display driver to synchronize with each other for each display pipeline. All endpoints should be added to a client before the client is registered to avsink core. Dynamic endpoints are not supported now. A display out here represents a physical HDMI/DP output port. And as long as it's usable in the system (i.e. physically connected to the HDMP/DP port on the machine board), the display output should be registered not matter the port is connected to an external display device or not. And if HW and display driver can support DP1.2 daisy chain (multiple DP display devices can be connected to a single port), multiple static displays outputs should be defined for the DP port according to the HW capability. The port & display device number can be indicated by the name (e.g. "i915_DDI_B", "i915_DDI_B_DEV0", "i915_DDI_B_DEV1", or "i915_DDI_B_DEV2"), defined by the display driver. The audio driver can check the endpoints of its peer display client and use an display endpoint's name, or a presumed display endpoint name, as peer name when registering an audio endpoint, thus the avsink core will bind the two display and audio endpoints as peers. struct avsink_endpoint { const char *name; /*name of the endpoint */ int type; /* DISPLAY_OUTPUT or AUDIO_INPUT */ void *context; /* private data, used as parameter of the ops */ struct avsink_endpoint_ops *ops; struct avsink_endpoint *peer; /* peer endpoint */ }; struct avsink_endpoint_ops { int (*get_caps) (enum had_caps_list query_element, void *capabilities, void *context); int (*set_caps) (enum had_caps_list set_element, void *capabilities, void *context); int (*event_handler) (enum avsink_event_type event_type, void *context); }; API: int avsink_new_endpoint (struct avsink_client *client, const char *name, int type, /* DISPLAY_OUTPUT or AUDIO_INPUT*/ void *context, const char *peer_name, /* can be NULL if no clue */ avsink_endpoint_ops *ops, struct avsink_endpoint **endpoint_ret); int avsink_endpoint_get_caps(struct avsink_endpoint *endpoint, enum avsink_caps_list t get_element, void *capabilities); int avsink_endpoint_set_caps(struct avsink_endpoint *endpoint, enum had_caps_list set_element, void *capabilities); int avsink_endpoint_post_event(struct avsink_endpoint *endpoint, enum avsink_event_type event_type); 4. Get/Set caps on an endpoint The display or audio driver can get or set capabilities on an endpoint. Depending on the capability ID, the avsink core will call get_caps/set_caps ops of this endpoint, or call get_caps/set_caps ops of its peer endpoint and return the result to the caller. enum avsink_caps_list { /* capabilities for display output endpoints */ AVSINK_GET_DISPLAY_ELD = 1, AVSINK_GET_DISPLAY_TYPE, /* HDMI or DisplayPort */ AVSINK_GET_DISPLAY_NAME, /* Hope to use display device name under /sys/class/drm, like "card0-DP-1", for user * space to figure out which HDMI/DP output on the drm side corresponds to which audio * stream device on the alsa side */ AVSINK_GET_DISPLAY_SAMPLING_FREQ, /* HDMI TMDS clock or DP link symbol clock, for audio driver to * program N value */ AVSINK_GET_DISPLAY_HDCP_STATUS, AVSINK_GET_DISPLAY_AUDIO_STATUS, /* Whether audio is enabled */ AVSINK_SET_DISPLAY_ENABLE_AUDIO, /* Enable audio */ AVSINK_SET_DISPLAY_DISABLE_AUDIO, /* Disable audio */ AVSINK_SET_DISPLAY_ENABLE_AUDIO_INT, /* Enable audio interrupt */ AVSINK_SET_DISPLAY_DISABLE_AUDIO_INT, /* Disable audio interrupt */ /* capabilities for audio input endpoints */ AVSINK_GET_AUDIO_IS_BUSY, /* Whether there is an active audio streaming */ OTHERS_TBD, }; For example, the audio driver can query ELD info on an audio input endpoint by using caps AVSINK_GET_DISPLAY_ELD, and avsink core will call get_caps() on the peer display output endpoint and return the ELD info to the audio driver. Some audio driver may only use part of these caps. E.g. HD-Audio driver can use bus commands instead of the ops to control the audio on gfx side, so it doesn't use caps like ENABLE/DISABLE_AUDIO or ENABLE/DISABLE_AUDIO. When the display driver want to disable a display pipeline for hot-plug, mode change or power saving, it can use caps AVSINK_GET_AUDIO_IS_BUSY to check if the audio input is busy (active streaming) on this display pipeline. And if audio is busy, the display driver can choose to wait or go ahead to disable display pipeline anyway. For the latter case, the audio input endpoint will be notified by an event and should abort audio streaming. 5. Event handling of endpoints A driver can post events on an endpoint. Depending on the event type, the avsink core will call the endpoint's event handler or pass the event to its peer endpoint and trigger the peer's event handler function if defined. int avsink_endpoint_post_event(struct avsink_endpoint *endpoint, enum avsink_event_type event_type); Now we only defined event types which should be handled by the audio input endpoints. The event types can be extended in the future. enum avsink_event_type { AVSINK_EVENT_DISPLAY_DISABLE = 1, /* The display pipeline is disabled for hot-plug, mode change or * suspend. Audio driver should stop any active streaming. */ AVSINK_EVENT_DISPLAY_ENABLE, /* The display pipeline is enabled after hot-plug, mode change or * resume. Audio driver can restore previously interrupted streaming */ AVSINK_EVENT_DISPLAY_MODE_CHANGE, /* Display mode change event. At this time, the new display mode is * configured but the display pipeline is not enabled yet. Audio driver * can do some configurations such as programing N value */ AVSINK_EVENT_DISPLAY_AUDIO_BUFFER_DONE, /* Audio Buffer done interrupts. Only for audio drivers if DMA and * interrupt are handled by GPU */ AVSINK_EVENT_DISPLAY_AUDIO_BUFFER_UNDERRUN, /* Audio Buffer under run interrupts. Only for audio drivers if * DMA and interrupt are handled by GPU */ }; So for a display driver, it can post an event on a display output endpoint and get processed by the peer audio input endpoint. Or it can also directly post an event on a peer audio input endpoint, by using the 'peer' pointer on a display output endpoint. 6. Display register operation (optional) Some audio driver needs to access GPU audio registers. The register ops are provided by the peer display client. struct avsink_registers_ops { int (*read_register) (uint32_t reg_addr, uint32_t *data, void *context); int (*write_register) (uint32_t reg_addr, uint32_t data, void *context); int (*read_modify_register) (uint32_t reg_addr, uint32_t data, uint32_t mask, void *context); int avsink_define_reg_ops (struct avsink_client *client, struct avsink_registers_ops *ops); And avsink core provides API for the audio driver to access the display registers: int avsink_read_display_register(struct avsink_client *client , uint32_t offset, uint32_t *data); int avsink_write_display_register(struct avsink_client *client , uint32_t offset, uint32_t data); int avsink_read_modify_display_register(struct avsink_client *client, uint32_t offset, uint32_t data, uint32_t mask); If the client is an audio client, the avsink core will find it peer display client and call its register ops; and if the client is a display client, the avsink core will just call its own register ops. Thanks Mengdong

10 23

Re: [alsa-devel] [PATCH] ARM: i.MX: Compile FIQ code for AVIC & TZIC only
by Lothar Waßmann 03 Jun '14

03 Jun '14

Hi, Alexander Shiyan wrote: > There are no reason to compile FIQ specific code for GIC IRQ > controller (i.MX6). This patch restrict compilation for FIQ > specific code for AVIC and TZIC only. > > Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> > --- > arch/arm/mach-imx/Makefile | 6 +++--- > 1 file changed, 3 insertions(+), 3 deletions(-) > > diff --git a/arch/arm/mach-imx/Makefile b/arch/arm/mach-imx/Makefile > index bbe93bb..620a760 100644 > --- a/arch/arm/mach-imx/Makefile > +++ b/arch/arm/mach-imx/Makefile > @@ -1,4 +1,4 @@ > -obj-y := time.o cpu.o system.o irq-common.o > +obj-y := time.o cpu.o system.o > > obj-$(CONFIG_SOC_IMX1) += clk-imx1.o mm-imx1.o > obj-$(CONFIG_SOC_IMX21) += clk-imx21.o mm-imx21.o > @@ -21,8 +21,8 @@ obj-$(CONFIG_COMMON_CLK) += clk-pllv1.o clk-pllv2.o clk-pllv3.o clk-gate2.o \ > obj-$(CONFIG_IMX_HAVE_IOMUX_V1) += iomux-v1.o > obj-$(CONFIG_ARCH_MXC_IOMUX_V3) += iomux-v3.o > > -obj-$(CONFIG_MXC_TZIC) += tzic.o > -obj-$(CONFIG_MXC_AVIC) += avic.o > +obj-$(CONFIG_MXC_TZIC) += tzic.o irq-common.o > +obj-$(CONFIG_MXC_AVIC) += avic.o irq-common.o > > obj-$(CONFIG_MXC_USE_EPIT) += epit.o > obj-$(CONFIG_MXC_DEBUG_BOARD) += 3ds_debugboard.o > This patch introduces a build error in sound/soc/fsl/imx-pcm-fiq.c: sound/built-in.o: In function `imx_pcm_fiq_free': last.c:(.text+0x2aca8): undefined reference to `mxc_set_irq_fiq' sound/built-in.o: In function `imx_pcm_fiq_init': last.c:(.text+0x2b1a4): undefined reference to `mxc_set_irq_fiq' last.c:(.text+0x2b200): undefined reference to `mxc_set_irq_fiq' make[1]: *** [vmlinux] Error 1 make: *** [sub-make] Error 2 This could be fixed with the following patch: --- mxc-fiq.orig/sound/soc/fsl/Kconfig 2014-06-02 11:25:38.315960451 +0200 +++ mxc-fiq/sound/soc/fsl/Kconfig 2014-06-02 11:25:38.411960595 +0200 @@ -15,7 +15,7 @@ config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC + select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_AVIC || MXC_TZIC) help Say Y if you want to add Synchronous Serial Interface (SSI) support for the Freescale CPUs. Lothar Waßmann -- ___________________________________________________________ Ka-Ro electronics GmbH | Pascalstraße 22 | D - 52076 Aachen Phone: +49 2408 1402-0 | Fax: +49 2408 1402-10 Geschäftsführer: Matthias Kaussen Handelsregistereintrag: Amtsgericht Aachen, HRB 4996 www.karo-electronics.de | info(a)karo-electronics.de ___________________________________________________________

3 3

[alsa-devel] [PATCH] firewire-lib: Remove a comment about restriction of asynchronous operation
by Takashi Sakamoto 03 Jun '14

03 Jun '14

The comment for fcp_avc_transaction() describes it doesn't support this type of operation. But it was already supported by this commit. 00a7bb81c20f3e81711e28e0f6c08cee8fd18514 ALSA: firewire-lib: Add support for deferred transaction Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fcp.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/firewire/fcp.c b/sound/firewire/fcp.c index d48e326..0619597 100644 --- a/sound/firewire/fcp.c +++ b/sound/firewire/fcp.c @@ -218,8 +218,6 @@ struct fcp_transaction { * * @command and @response can point to the same buffer. * - * Asynchronous operation (INTERIM, NOTIFY) is not supported at the moment. - * * Returns the actual size of the response frame, or a negative error code. */ int fcp_avc_transaction(struct fw_unit *unit, -- 1.8.3.2

2 1

[alsa-devel] [PATCH 0/6] ASoC: rsnd: tidyup for DMA support for DT
by Kuninori Morimoto 03 Jun '14

03 Jun '14

Hi Mark These patches are tidyup code and fixup for DMA support of DT #1 - #4 are cleanup code patches #5 - #6 are fixup for DMA for DT Kuninori Morimoto (6): ASoC: rsnd: DT node clean up by using the of_node_put() ASoC: rsnd: save platform_device instead of device ASoC: rsnd: remove rsnd_src_non_ops ASoC: rsnd: module name is unified ASoC: rsnd: care DMA slave channel name for DT ASoC: rsnd: add rsnd_gen_dma_addr() for DMAC addr .../devicetree/bindings/sound/renesas,rsnd.txt | 1 + sound/soc/sh/rcar/core.c | 87 ++++++++++++++++-- sound/soc/sh/rcar/dvc.c | 20 ++++- sound/soc/sh/rcar/gen.c | 95 ++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 10 ++- sound/soc/sh/rcar/src.c | 50 +++++++---- sound/soc/sh/rcar/ssi.c | 28 ++++-- 7 files changed, 260 insertions(+), 31 deletions(-)

4 9

[alsa-devel] [PATCH] ASoC: rsnd: fixup SSI interrupts style of DeviceTree
by Kuninori Morimoto 03 Jun '14

03 Jun '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current SSI interrupts has been set to each port. But it is too ugly style. SSI interrupts will be listed under rcar_sound,ssi node by this patch Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- >> Mark There is still no DT user for this driver at this point. No conflict happens on platform. .../devicetree/bindings/sound/renesas,rsnd.txt | 50 ++++++++------------ sound/soc/sh/rcar/ssi.c | 2 +- 2 files changed, 21 insertions(+), 31 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 8346cab..1fba56b 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -53,36 +53,26 @@ rcar_sound: rcar_sound@0xffd90000 { }; rcar_sound,ssi { - ssi0: ssi@0 { - interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi1: ssi@1 { - interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi2: ssi@2 { - interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi3: ssi@3 { - interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi4: ssi@4 { - interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi5: ssi@5 { - interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi6: ssi@6 { - interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi7: ssi@7 { - interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi8: ssi@8 { - interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi9: ssi@9 { - interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; - }; + interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>, + <0 371 IRQ_TYPE_LEVEL_HIGH>, + <0 372 IRQ_TYPE_LEVEL_HIGH>, + <0 373 IRQ_TYPE_LEVEL_HIGH>, + <0 374 IRQ_TYPE_LEVEL_HIGH>, + <0 375 IRQ_TYPE_LEVEL_HIGH>, + <0 376 IRQ_TYPE_LEVEL_HIGH>, + <0 377 IRQ_TYPE_LEVEL_HIGH>, + <0 378 IRQ_TYPE_LEVEL_HIGH>, + <0 379 IRQ_TYPE_LEVEL_HIGH>; + ssi0: ssi@0 { }; + ssi1: ssi@1 { }; + ssi2: ssi@2 { }; + ssi3: ssi@3 { }; + ssi4: ssi@4 { }; + ssi5: ssi@5 { }; + ssi6: ssi@6 { }; + ssi7: ssi@7 { }; + ssi8: ssi@8 { }; + ssi9: ssi@9 { }; }; rcar_sound,dai { diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 2df723d..b311bc6 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -588,7 +588,7 @@ static void rsnd_of_parse_ssi(struct platform_device *pdev, /* * irq */ - ssi_info->pio_irq = irq_of_parse_and_map(np, 0); + ssi_info->pio_irq = irq_of_parse_and_map(node, i); /* * DMA -- 1.7.9.5

4 6

[alsa-devel] [PATCH] ASoC: soc-cache: Fix the return value of snd_soc_cache_init when reg_size is 0
by Fabio Estevam 02 Jun '14

02 Jun '14

Commit b59dce53ef6 (ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error.) caused the following probe error: wm8962 0-001a: customer id 0 revision D wm8962 0-001a: ASoC: Failed to set cache compression type: -22 imx-wm8962 sound: snd_soc_register_card failed (-22) imx-wm8962: probe of sound failed with error -22 If reg_size just return success instead. Suggested-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/soc-cache.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 8fff5b6..00e70b6 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -73,7 +73,7 @@ int snd_soc_cache_init(struct snd_soc_codec *codec) reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; if (!reg_size) - return -EINVAL; + return 0; mutex_init(&codec->cache_rw_mutex); -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: cache: Fix error code when not using ASoC level cache
by Mark Brown 02 Jun '14

02 Jun '14

From: Mark Brown <broonie(a)linaro.org> It is not an error to have no cache so we shouldn't return an error code and cause our callers to fail, just silently do nothing instead. Thanks to Jarkko for identify the problematic commit. Reported-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Reported-by: Fabio Estevam <festevam(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-cache.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 8fff5b6..00e70b6 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -73,7 +73,7 @@ int snd_soc_cache_init(struct snd_soc_codec *codec) reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; if (!reg_size) - return -EINVAL; + return 0; mutex_init(&codec->cache_rw_mutex); -- 2.0.0.rc2

1 0

[alsa-devel] Failed to set cache compression type
by Fabio Estevam 02 Jun '14

02 Jun '14

Hi, I am booting linux-next 20140602 on a mx6qsabresd and I get the following probe error: wm8962 0-001a: customer id 0 revision D wm8962 0-001a: ASoC: Failed to set cache compression type: -22 imx-wm8962 sound: snd_soc_register_card failed (-22) imx-wm8962: probe of sound failed with error -22 What is the proper way to fix it? Thanks, Fabio Estevam

4 5

Re: [alsa-devel] [PATCH 1/1] ALSA: hda/realtek - Correction of fixup codes for PB V7900 laptop
by Ronan 02 Jun '14

02 Jun '14

> Message du 02/06/14 08:46 > De : "Takashi Iwai" > A : ronan.marquet(a)orange.fr > Copie à : alsa-devel(a)alsa-project.org > Objet : Re: [PATCH 1/1] ALSA: hda/realtek - Correction of fixup codes for PB V7900 laptop > > At Sun, 1 Jun 2014 18:38:53 +0200, > ronan.marquet(a)orange.fr wrote: > > > > From: Ronan Marquet > > > > Correcion of wrong fixup entries add in commit ca8f0424 to replace > > static model quirk for PB V7900 laptop (will model). > > > > Signed-off-by: Ronan Marquet > > > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > > index 49e884f..0e3a7f6 100644 > > --- a/sound/pci/hda/patch_realtek.c > > +++ b/sound/pci/hda/patch_realtek.c > > @@ -1647,12 +1647,10 @@ static const struct hda_fixup alc260_fixups[] = { > > [ALC260_FIXUP_COEF] = { > > .type = HDA_FIXUP_VERBS, > > .v.verbs = (const struct hda_verb[]) { > > - { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, > > - { 0x20, AC_VERB_SET_PROC_COEF, 0x3040 }, > > + { 0x1a, AC_VERB_SET_COEF_INDEX, 0x07 }, > > + { 0x1a, AC_VERB_SET_PROC_COEF, 0x3040 }, > > That part looks correct, however: > > > { } > > }, > > - .chained = true, > > - .chain_id = ALC260_FIXUP_HP_PIN_0F, > > Is this removal mandatory? In other words, is keeping this harmful? > The original quirk had this, so it may be safer to keep it for other > machines. > Yes, if i keep this two lines, i have a crackling sound and the internal output is blinking into Gnome's sound setting (Jack detection problem ?).Whitout this lines, all is working find on my laptop but, perhpas, it could be fixe differently ? thanks, Ronan > > thanks, > > Takashi > > > }, > > [ALC260_FIXUP_GPIO1] = { > > .type = HDA_FIXUP_VERBS, > > -- > > 2.0.0.rc2 > > >

2 2

[alsa-devel] [PATCH v2] ASoC: soc-cache: Fix the return value of snd_soc_cache_init when reg_size is 0
by Fabio Estevam 02 Jun '14

02 Jun '14

Commit b59dce53ef6 (ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error.) caused the following probe error: wm8962 0-001a: customer id 0 revision D wm8962 0-001a: ASoC: Failed to set cache compression type: -22 imx-wm8962 sound: snd_soc_register_card failed (-22) imx-wm8962: probe of sound failed with error -22 If reg_size is 0 just return success instead. Suggested-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Fix typo "If reg_size is 0 just return success instead." sound/soc/soc-cache.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 8fff5b6..00e70b6 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -73,7 +73,7 @@ int snd_soc_cache_init(struct snd_soc_codec *codec) reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; if (!reg_size) - return -EINVAL; + return 0; mutex_init(&codec->cache_rw_mutex); -- 1.8.3.2

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix COEF widget NID for ALC260 replacer fixup
by Takashi Iwai 02 Jun '14

02 Jun '14

The conversion to a fixup table for Replacer model with ALC260 in commit 20f7d928 took the wrong widget NID for COEF setups. Namely, NID 0x1a should have been used instead of NID 0x20, which is the common node for all Realtek codecs but ALC260. Fixes: 20f7d928fa6e ('ALSA: hda/realtek - Replace ALC260 model=replacer with the auto-parser') Cc: <stable(a)vger.kernel.org> [v3.4+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a00cd69bc3aa..3d611c37f25a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1669,8 +1669,8 @@ static const struct hda_fixup alc260_fixups[] = { [ALC260_FIXUP_REPLACER] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { - { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, - { 0x20, AC_VERB_SET_PROC_COEF, 0x3050 }, + { 0x1a, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x1a, AC_VERB_SET_PROC_COEF, 0x3050 }, { } }, .chained = true, -- 1.9.3

1 0

[alsa-devel] [PATCH] ASoC: intel: The Baytrail/MAX98090 driver depends on I2C
by Mark Brown 02 Jun '14

02 Jun '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index cd3498736e91..c30fedb3e149 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -52,7 +52,7 @@ config SND_SOC_INTEL_BYT_RT5640_MACH config SND_SOC_INTEL_BYT_MAX98090_MACH tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C select SND_SOC_INTEL_BAYTRAIL select SND_SOC_MAX98090 help -- 2.0.0.rc4

2 2

[alsa-devel] On patch_realtek.c can' sent command to the codec via
by asd 02 Jun '14

02 Jun '14

Hi On patch_realtek.c can' sent command to the codec via command snd_hda_codec_write. Suppose to be entire module useless and hda-intel module to be a active.

1 0

[alsa-devel] Realtek Alc 883 not have eapd
by asd 02 Jun '14

02 Jun '14

Below is patch wich disable eapd.

2 1

[alsa-devel] [PATCH] firewire-lib: Use IEC 61883-6 compliant labels for Raw Audio data
by Takashi Sakamoto 02 Jun '14

02 Jun '14

According to AM824 in IEC 61883-6:2002, 2 bits in LSB of label for Raw Audio data means Valid Length Code (VBL). Ths value is: - b00 for 24 bits sample (label is 0x40) - b01 for 20 bits sample (label is 0x41) - b10 for 16 bits sample (label is 0x42) But current firewire-lib apply 24 bits label for both of 16/24 bits samples. As long as developers investigate BeBoB/Fireworks/OXFW/Dice, all of them have a behaviour to ignore the label. They can generate correct sound even if firewire-lib gives wrong label (i.e. 0xff). On BeBoB, this is not only for Raw Audio data channel, but also for IEC 60958 Conformant data channel. So there is little possibility of regression. Acked-by: Clemens Ladisch <clemens(a)ladisch.de> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 31dd1cf..f96bf4c 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -418,7 +418,7 @@ static void amdtp_write_s16(struct amdtp_stream *s, for (i = 0; i < frames; ++i) { for (c = 0; c < channels; ++c) { buffer[s->pcm_positions[c]] = - cpu_to_be32((*src << 8) | 0x40000000); + cpu_to_be32((*src << 8) | 0x42000000); src++; } buffer += s->data_block_quadlets; -- 1.8.3.2

2 1

[alsa-devel] [PATCH 1/1] ALSA: hda/realtek - Correction of fixup codes for PB V7900 laptop
by ronan.marquet＠orange.fr 02 Jun '14

02 Jun '14

From: Ronan Marquet <ronan.marquet(a)orange.fr> Correcion of wrong fixup entries add in commit ca8f0424 to replace static model quirk for PB V7900 laptop (will model). Signed-off-by: Ronan Marquet <ronan.marquet(a)orange.fr> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 49e884f..0e3a7f6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1647,12 +1647,10 @@ static const struct hda_fixup alc260_fixups[] = { [ALC260_FIXUP_COEF] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { - { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, - { 0x20, AC_VERB_SET_PROC_COEF, 0x3040 }, + { 0x1a, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x1a, AC_VERB_SET_PROC_COEF, 0x3040 }, { } }, - .chained = true, - .chain_id = ALC260_FIXUP_HP_PIN_0F, }, [ALC260_FIXUP_GPIO1] = { .type = HDA_FIXUP_VERBS, -- 2.0.0.rc2

2 1

[alsa-devel] no sound with intel audio
by Petr Maivald 01 Jun '14

01 Jun '14

[1.] One line summary of the problem: no sound [2.] Full description of the problem/report: no sound [3.] Keywords (i.e., modules, networking, kernel): [4.] Kernel version (from /proc/version): Linux version 3.15.0-031500rc7-generic (apw@gomeisa) (gcc version 4.6.3 (Ubuntu/Linaro 4.6.3-1ubuntu5) ) #201405251935 SMP Sun May 25 23:36:23 UTC 2014 [5.] Output of Oops.. message (if applicable) with symbolic information resolved (see Documentation/oops-tracing.txt) [6.] A small shell script or example program which triggers the problem (if possible) [7.] Environment Description: Ubuntu 14.04 LTS Release: 14.04 [7.1.] Software (add the output of the ver_linux script here) If some fields are empty or look unusual you may have an old version. Compare to the current minimal requirements in Documentation/Changes. Linux BuildOut 3.15.0-031500rc7-generic #201405251935 SMP Sun May 25 23:36:23 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux Gnu C 4.8 Gnu make 3.81 binutils 2.24 util-linux 2.20.1 mount support module-init-tools 15 e2fsprogs 1.42.9 PPP 2.4.5 Linux C Library 2.19 Dynamic linker (ldd) 2.19 Procps 3.3.9 Net-tools 1.60 Kbd 1.15.5 Sh-utils 8.21 wireless-tools 30 Modules Loaded nls_iso8859_1 dm_crypt ctr ccm rfcomm bnep intel_rapl binfmt_misc arc4 x86_pkg_temp_thermal intel_powerclamp snd_hda_codec_realtek snd_hda_codec_hdmi snd_hda_codec_generic iwlmvm coretemp kvm_intel kvm mac80211 crct10dif_pclmul crc32_pclmul ghash_clmulni_intel aesni_intel aes_x86_64 uvcvideo lrw gf128mul glue_helper ablk_helper cryptd snd_hda_intel videobuf2_vmalloc snd_hda_controller videobuf2_memops microcode snd_hda_codec i915 videobuf2_core snd_hwdep snd_pcm videodev rts5139 snd_seq_midi snd_seq_midi_event btusb snd_rawmidi snd_seq bluetooth iwlwifi drm_kms_helper dell_wmi sparse_keymap 6lowpan_iphc cfg80211 drm parport_pc dell_laptop ppdev mei_me lp snd_seq_device dcdbas snd_timer i2c_algo_bit mei parport snd soundcore lpc_ich joydev wmi serio_raw video soc_button_array mac_hid psmouse ahci libahci r8169 mii [7.2.] Processor information (from /proc/cpuinfo): processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 69 model name : Intel(R) Core(TM) i7-4500U CPU @ 1.80GHz stepping : 1 microcode : 0x17 cpu MHz : 2049.000 cache size : 4096 KB physical id : 0 siblings : 4 core id : 0 cpu cores : 2 apicid : 0 initial apicid : 0 fpu : yes fpu_exception : yes cpuid level : 13 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe syscall nx pdpe1gb rdtscp lm constant_tsc arch_perfmon pebs bts rep_good nopl xtopology nonstop_tsc aperfmperf eagerfpu pni pclmulqdq dtes64 monitor ds_cpl vmx est tm2 ssse3 fma cx16 xtpr pdcm pcid sse4_1 sse4_2 movbe popcnt tsc_deadline_timer aes xsave avx f16c rdrand lahf_lm abm ida arat epb xsaveopt pln pts dtherm tpr_shadow vnmi flexpriority ept vpid fsgsbase tsc_adjust bmi1 avx2 smep bmi2 erms invpcid bogomips : 4788.80 clflush size : 64 cache_alignment : 64 address sizes : 39 bits physical, 48 bits virtual power management: processor : 1 vendor_id : GenuineIntel cpu family : 6 model : 69 model name : Intel(R) Core(TM) i7-4500U CPU @ 1.80GHz stepping : 1 microcode : 0x17 cpu MHz : 1899.000 cache size : 4096 KB physical id : 0 siblings : 4 core id : 0 cpu cores : 2 apicid : 1 initial apicid : 1 fpu : yes fpu_exception : yes cpuid level : 13 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe syscall nx pdpe1gb rdtscp lm constant_tsc arch_perfmon pebs bts rep_good nopl xtopology nonstop_tsc aperfmperf eagerfpu pni pclmulqdq dtes64 monitor ds_cpl vmx est tm2 ssse3 fma cx16 xtpr pdcm pcid sse4_1 sse4_2 movbe popcnt tsc_deadline_timer aes xsave avx f16c rdrand lahf_lm abm ida arat epb xsaveopt pln pts dtherm tpr_shadow vnmi flexpriority ept vpid fsgsbase tsc_adjust bmi1 avx2 smep bmi2 erms invpcid bogomips : 4788.80 clflush size : 64 cache_alignment : 64 address sizes : 39 bits physical, 48 bits virtual power management: processor : 2 vendor_id : GenuineIntel cpu family : 6 model : 69 model name : Intel(R) Core(TM) i7-4500U CPU @ 1.80GHz stepping : 1 microcode : 0x17 cpu MHz : 1786.500 cache size : 4096 KB physical id : 0 siblings : 4 core id : 1 cpu cores : 2 apicid : 2 initial apicid : 2 fpu : yes fpu_exception : yes cpuid level : 13 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe syscall nx pdpe1gb rdtscp lm constant_tsc arch_perfmon pebs bts rep_good nopl xtopology nonstop_tsc aperfmperf eagerfpu pni pclmulqdq dtes64 monitor ds_cpl vmx est tm2 ssse3 fma cx16 xtpr pdcm pcid sse4_1 sse4_2 movbe popcnt tsc_deadline_timer aes xsave avx f16c rdrand lahf_lm abm ida arat epb xsaveopt pln pts dtherm tpr_shadow vnmi flexpriority ept vpid fsgsbase tsc_adjust bmi1 avx2 smep bmi2 erms invpcid bogomips : 4788.80 clflush size : 64 cache_alignment : 64 address sizes : 39 bits physical, 48 bits virtual power management: processor : 3 vendor_id : GenuineIntel cpu family : 6 model : 69 model name : Intel(R) Core(TM) i7-4500U CPU @ 1.80GHz stepping : 1 microcode : 0x17 cpu MHz : 1974.000 cache size : 4096 KB physical id : 0 siblings : 4 core id : 1 cpu cores : 2 apicid : 3 initial apicid : 3 fpu : yes fpu_exception : yes cpuid level : 13 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe syscall nx pdpe1gb rdtscp lm constant_tsc arch_perfmon pebs bts rep_good nopl xtopology nonstop_tsc aperfmperf eagerfpu pni pclmulqdq dtes64 monitor ds_cpl vmx est tm2 ssse3 fma cx16 xtpr pdcm pcid sse4_1 sse4_2 movbe popcnt tsc_deadline_timer aes xsave avx f16c rdrand lahf_lm abm ida arat epb xsaveopt pln pts dtherm tpr_shadow vnmi flexpriority ept vpid fsgsbase tsc_adjust bmi1 avx2 smep bmi2 erms invpcid bogomips : 4788.80 clflush size : 64 cache_alignment : 64 address sizes : 39 bits physical, 48 bits virtual power management: [7.3.] Module information (from /proc/modules): nls_iso8859_1 12713 0 - Live 0x0000000000000000 dm_crypt 23456 0 - Live 0x0000000000000000 ctr 13193 3 - Live 0x0000000000000000 ccm 17856 3 - Live 0x0000000000000000 rfcomm 75078 8 - Live 0x0000000000000000 bnep 19884 2 - Live 0x0000000000000000 intel_rapl 19238 0 - Live 0x0000000000000000 binfmt_misc 17508 1 - Live 0x0000000000000000 arc4 12573 2 - Live 0x0000000000000000 x86_pkg_temp_thermal 14312 0 - Live 0x0000000000000000 intel_powerclamp 19099 0 - Live 0x0000000000000000 snd_hda_codec_realtek 72529 1 - Live 0x0000000000000000 snd_hda_codec_hdmi 48229 1 - Live 0x0000000000000000 snd_hda_codec_generic 70087 1 snd_hda_codec_realtek, Live 0x0000000000000000 iwlmvm 220688 0 - Live 0x0000000000000000 coretemp 13638 0 - Live 0x0000000000000000 kvm_intel 148919 0 - Live 0x0000000000000000 kvm 463808 1 kvm_intel, Live 0x0000000000000000 mac80211 663883 1 iwlmvm, Live 0x0000000000000000 crct10dif_pclmul 14268 0 - Live 0x0000000000000000 crc32_pclmul 13180 0 - Live 0x0000000000000000 ghash_clmulni_intel 13230 0 - Live 0x0000000000000000 aesni_intel 152648 6 - Live 0x0000000000000000 aes_x86_64 17131 1 aesni_intel, Live 0x0000000000000000 uvcvideo 82299 0 - Live 0x0000000000000000 lrw 13323 1 aesni_intel, Live 0x0000000000000000 gf128mul 14951 1 lrw, Live 0x0000000000000000 glue_helper 14095 1 aesni_intel, Live 0x0000000000000000 ablk_helper 13597 1 aesni_intel, Live 0x0000000000000000 cryptd 20531 3 ghash_clmulni_intel,aesni_intel,ablk_helper, Live 0x0000000000000000 snd_hda_intel 30608 1 - Live 0x0000000000000000 videobuf2_vmalloc 13216 1 uvcvideo, Live 0x0000000000000000 snd_hda_controller 35518 1 snd_hda_intel, Live 0x0000000000000000 videobuf2_memops 13362 1 videobuf2_vmalloc, Live 0x0000000000000000 microcode 24391 0 - Live 0x0000000000000000 snd_hda_codec 144671 5 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_codec_generic,snd_hda_intel,snd_hda_controller, Live 0x0000000000000000 i915 873516 4 - Live 0x0000000000000000 videobuf2_core 45435 1 uvcvideo, Live 0x0000000000000000 snd_hwdep 13613 1 snd_hda_codec, Live 0x0000000000000000 snd_pcm 113863 4 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_controller,snd_hda_codec, Live 0x0000000000000000 videodev 149504 2 uvcvideo,videobuf2_core, Live 0x0000000000000000 rts5139 318308 0 - Live 0x0000000000000000 (C) snd_seq_midi 13564 0 - Live 0x0000000000000000 snd_seq_midi_event 14899 1 snd_seq_midi, Live 0x0000000000000000 btusb 32555 0 - Live 0x0000000000000000 snd_rawmidi 30865 1 snd_seq_midi, Live 0x0000000000000000 snd_seq 67636 2 snd_seq_midi,snd_seq_midi_event, Live 0x0000000000000000 bluetooth 461775 22 rfcomm,bnep,btusb, Live 0x0000000000000000 iwlwifi 185872 1 iwlmvm, Live 0x0000000000000000 drm_kms_helper 59729 1 i915, Live 0x0000000000000000 dell_wmi 12681 0 - Live 0x0000000000000000 sparse_keymap 13890 1 dell_wmi, Live 0x0000000000000000 6lowpan_iphc 18968 1 bluetooth, Live 0x0000000000000000 cfg80211 515506 3 iwlmvm,mac80211,iwlwifi, Live 0x0000000000000000 drm 310655 4 i915,drm_kms_helper, Live 0x0000000000000000 parport_pc 32906 0 - Live 0x0000000000000000 dell_laptop 18315 0 - Live 0x0000000000000000 ppdev 17711 0 - Live 0x0000000000000000 mei_me 18562 0 - Live 0x0000000000000000 lp 17799 0 - Live 0x0000000000000000 snd_seq_device 14497 3 snd_seq_midi,snd_rawmidi,snd_seq, Live 0x0000000000000000 dcdbas 15017 1 dell_laptop, Live 0x0000000000000000 snd_timer 30118 2 snd_pcm,snd_seq, Live 0x0000000000000000 i2c_algo_bit 13564 1 i915, Live 0x0000000000000000 mei 87522 1 mei_me, Live 0x0000000000000000 parport 42481 3 parport_pc,ppdev,lp, Live 0x0000000000000000 snd 74195 13 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_codec_generic,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_rawmidi,snd_seq,snd_seq_device,snd_timer, Live 0x0000000000000000 soundcore 12680 2 snd_hda_codec,snd, Live 0x0000000000000000 lpc_ich 21176 0 - Live 0x0000000000000000 joydev 17587 0 - Live 0x0000000000000000 wmi 19379 1 dell_wmi, Live 0x0000000000000000 serio_raw 13483 0 - Live 0x0000000000000000 video 19932 1 i915, Live 0x0000000000000000 soc_button_array 12769 0 - Live 0x0000000000000000 mac_hid 13275 0 - Live 0x0000000000000000 psmouse 113320 0 - Live 0x0000000000000000 ahci 30167 1 - Live 0x0000000000000000 libahci 32191 1 ahci, Live 0x0000000000000000 r8169 73316 0 - Live 0x0000000000000000 mii 13981 1 r8169, Live 0x0000000000000000 [7.4.] Loaded driver and hardware information (/proc/ioports, /proc/iomem) 0000-0cf7 : PCI Bus 0000:00 0000-001f : dma1 0020-0021 : pic1 0040-0043 : timer0 0050-0053 : timer1 0060-0060 : keyboard 0062-0062 : EC data 0064-0064 : keyboard 0066-0066 : EC cmd 0070-0077 : rtc0 0080-008f : dma page reg 00a0-00a1 : pic2 00c0-00df : dma2 00f0-00ff : fpu 03c0-03df : vesafb 0680-069f : pnp 00:03 0cf8-0cff : PCI conf1 0d00-ffff : PCI Bus 0000:00 164e-164f : pnp 00:03 1800-1803 : ACPI PM1a_EVT_BLK 1804-1805 : ACPI PM1a_CNT_BLK 1808-180b : ACPI PM_TMR 1810-1815 : ACPI CPU throttle 1830-1833 : iTCO_wdt 1850-1850 : ACPI PM2_CNT_BLK 1854-1857 : pnp 00:05 1860-187f : iTCO_wdt 1880-189f : ACPI GPE0_BLK 3000-3fff : PCI Bus 0000:01 3000-30ff : 0000:01:00.0 3000-30ff : r8169 4000-403f : 0000:00:02.0 4040-405f : 0000:00:1f.3 4060-407f : 0000:00:1f.2 4060-407f : ahci 4080-4087 : 0000:00:1f.2 4080-4087 : ahci 4088-408f : 0000:00:1f.2 4088-408f : ahci 4090-4093 : 0000:00:1f.2 4090-4093 : ahci 4094-4097 : 0000:00:1f.2 4094-4097 : ahci fd60-fd63 : pnp 00:03 ffff-ffff : pnp 00:03 ffff-ffff : pnp 00:03 ffff-ffff : pnp 00:03 00000000-00000fff : reserved 00001000-0009d3ff : System RAM 0009d400-0009ffff : reserved 000a0000-000bffff : PCI Bus 0000:00 000c0000-000cebff : Video ROM 000cf000-000cffff : Adapter ROM 000e0000-000fffff : reserved 000f0000-000fffff : System ROM 00100000-94d5ffff : System RAM 01000000-0177cd98 : Kernel code 0177cd99-01d1ef7f : Kernel data 01e7b000-01fe0fff : Kernel bss 94d60000-95d5ffff : reserved 95d60000-9aabefff : System RAM 9aabf000-9aebefff : reserved 9aebf000-9afbefff : ACPI Non-volatile Storage 9afbf000-9affefff : ACPI Tables 9afff000-9affffff : System RAM 9b000000-9f9fffff : reserved 9ba00000-9f9fffff : Graphics Stolen Memory 9fa00000-feafffff : PCI Bus 0000:00 9fa10000-9fa1ffff : pnp 00:08 9fa22000-9fa22fff : pnp 00:08 a0000000-afffffff : 0000:00:02.0 b0000000-b03fffff : 0000:00:02.0 b0400000-b04fffff : PCI Bus 0000:01 b0400000-b0403fff : 0000:01:00.0 b0400000-b0403fff : r8169 b0500000-b05fffff : PCI Bus 0000:02 b0500000-b0501fff : 0000:02:00.0 b0500000-b0501fff : iwlwifi b0600000-b06fffff : PCI Bus 0000:01 b0600000-b0600fff : 0000:01:00.0 b0600000-b0600fff : r8169 b0700000-b070ffff : 0000:00:14.0 b0700000-b070ffff : xhci_hcd b0710000-b0713fff : 0000:00:03.0 b0710000-b0713fff : ICH HD audio b0714000-b0717fff : 0000:00:1b.0 b0714000-b0717fff : ICH HD audio b0718000-b071801f : 0000:00:16.0 b0718000-b071801f : mei_me b0719000-b07190ff : 0000:00:1f.3 b071b000-b071b7ff : 0000:00:1f.2 b071b000-b071b7ff : ahci b071c000-b071c3ff : 0000:00:1d.0 b071c000-b071c3ff : ehci_hcd e0000000-efffffff : PCI MMCONFIG 0000 [bus 00-ff] e0000000-efffffff : reserved e0000000-efffffff : pnp 00:08 fe101000-fe112fff : reserved feb00000-feb0ffff : reserved fec00000-fec00fff : reserved fec00000-fec003ff : IOAPIC 0 fed00000-fee00fff : reserved fed00000-fed003ff : HPET 0 fed10000-fed17fff : pnp 00:08 fed18000-fed18fff : pnp 00:08 fed19000-fed19fff : pnp 00:08 fed1c000-fed1ffff : pnp 00:08 fed1f410-fed1f414 : iTCO_wdt fed20000-fed3ffff : pnp 00:08 fed90000-fed93fff : pnp 00:08 fee00000-fee00fff : Local APIC ff000000-ff000fff : pnp 00:08 ffc00000-ffffffff : reserved 100000000-25f5fffff : System RAM 25f600000-25fffffff : RAM buffer [7.5.] PCI information ('lspci -vvv' as root) 00:00.0 Host bridge: Intel Corporation Haswell-ULT DRAM Controller (rev 09) Subsystem: Dell Device 05ec Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort+ >SERR- <PERR- INTx- Latency: 0 Capabilities: [e0] Vendor Specific Information: Len=0c <?> 00:02.0 VGA compatible controller: Intel Corporation Haswell-ULT Integrated Graphics Controller (rev 09) (prog-if 00 [VGA controller]) Subsystem: Dell Device 05ec Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 63 Region 0: Memory at b0000000 (64-bit, non-prefetchable) [size=4M] Region 2: Memory at a0000000 (64-bit, prefetchable) [size=256M] Region 4: I/O ports at 4000 [size=64] Expansion ROM at <unassigned> [disabled] Capabilities: [90] MSI: Enable+ Count=1/1 Maskable- 64bit- Address: fee0f00c Data: 4122 Capabilities: [d0] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [a4] PCI Advanced Features AFCap: TP+ FLR+ AFCtrl: FLR- AFStatus: TP- Kernel driver in use: i915 00:03.0 Audio device: Intel Corporation Haswell-ULT HD Audio Controller (rev 09) Subsystem: Dell Device 05ec Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 65 Region 0: Memory at b0710000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit- Address: fee0f00c Data: 4162 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- Kernel driver in use: snd_hda_intel 00:14.0 USB controller: Intel Corporation Lynx Point-LP USB xHCI HC (rev 04) (prog-if 30 [XHCI]) Subsystem: Dell Device 05ec Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 58 Region 0: Memory at b0700000 (64-bit, non-prefetchable) [size=64K] Capabilities: [70] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA PME(D0-,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst+ PME-Enable- DSel=0 DScale=0 PME- Capabilities: [80] MSI: Enable+ Count=1/8 Maskable- 64bit+ Address: 00000000fee0f00c Data: 41a1 Kernel driver in use: xhci_hcd 00:16.0 Communication controller: Intel Corporation Lynx Point-LP HECI #0 (rev 04) Subsystem: Dell Device 05ec Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 61 Region 0: Memory at b0718000 (64-bit, non-prefetchable) [size=32] Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst+ PME-Enable- DSel=0 DScale=0 PME- Capabilities: [8c] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0f00c Data: 41d1 Kernel driver in use: mei_me 00:1b.0 Audio device: Intel Corporation Lynx Point-LP HD Audio Controller (rev 04) Subsystem: Dell Device 05ec Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 64 Region 0: Memory at b0714000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0f00c Data: 4152 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- Capabilities: [100 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed- WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=ff Status: NegoPending- InProgress- VC1: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable- ID=0 ArbSelect=Fixed TC/VC=00 Status: NegoPending- InProgress- Kernel driver in use: snd_hda_intel 00:1c.0 PCI bridge: Intel Corporation Lynx Point-LP PCI Express Root Port 3 (rev e4) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 I/O behind bridge: 00003000-00003fff Memory behind bridge: b0600000-b06fffff Prefetchable memory behind bridge: 00000000b0400000-00000000b04fffff Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA- VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [40] Express (v2) Root Port (Slot+), MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #3, Speed 5GT/s, Width x1, ASPM L0s L1, Exit Latency L0s <512ns, L1 <16us ClockPM- Surprise- LLActRep+ BwNot+ LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive+ BWMgmt+ ABWMgmt- SltCap: AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug- Surprise- Slot #2, PowerLimit 10.000W; Interlock- NoCompl+ SltCtl: Enable: AttnBtn- PwrFlt- MRL- PresDet- CmdCplt- HPIrq- LinkChg- Control: AttnInd Unknown, PwrInd Unknown, Power- Interlock- SltSta: Status: AttnBtn- PowerFlt- MRL- CmdCplt- PresDet+ Interlock- Changed: MRL- PresDet- LinkState+ RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- PMEIntEna+ CRSVisible- RootCap: CRSVisible- RootSta: PME ReqID 0000, PMEStatus- PMEPending- DevCap2: Completion Timeout: Range ABC, TimeoutDis+, LTR+, OBFF Not Supported ARIFwd- DevCtl2: Completion Timeout: 50us to 50ms, TimeoutDis-, LTR+, OBFF Disabled ARIFwd- LnkCtl2: Target Link Speed: 5GT/s, EnterCompliance- SpeedDis- Transmit Margin: Normal Operating Range, EnterModifiedCompliance- ComplianceSOS- Compliance De-emphasis: -6dB LnkSta2: Current De-emphasis Level: -3.5dB, EqualizationComplete-, EqualizationPhase1- EqualizationPhase2-, EqualizationPhase3-, LinkEqualizationRequest- Capabilities: [80] MSI: Enable+ Count=1/1 Maskable- 64bit- Address: fee0f00c Data: 4171 Capabilities: [90] Subsystem: Dell Device 05ec Capabilities: [a0] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [100 v0] #00 Capabilities: [200 v1] L1 PM Substates L1SubCap: PCI-PM_L1.2+ PCI-PM_L1.1+ ASPM_L1.2+ ASPM_L1.1+ L1_PM_Substates+ PortCommonModeRestoreTime=40us PortTPowerOnTime=10us Kernel driver in use: pcieport 00:1c.3 PCI bridge: Intel Corporation Lynx Point-LP PCI Express Root Port 4 (rev e4) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 I/O behind bridge: 0000f000-00000fff Memory behind bridge: b0500000-b05fffff Prefetchable memory behind bridge: 00000000fff00000-00000000000fffff Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA- VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [40] Express (v2) Root Port (Slot+), MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #4, Speed 5GT/s, Width x1, ASPM L0s L1, Exit Latency L0s <512ns, L1 <16us ClockPM- Surprise- LLActRep+ BwNot+ LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive+ BWMgmt+ ABWMgmt- SltCap: AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug- Surprise- Slot #3, PowerLimit 10.000W; Interlock- NoCompl+ SltCtl: Enable: AttnBtn- PwrFlt- MRL- PresDet- CmdCplt- HPIrq- LinkChg- Control: AttnInd Unknown, PwrInd Unknown, Power- Interlock- SltSta: Status: AttnBtn- PowerFlt- MRL- CmdCplt- PresDet+ Interlock- Changed: MRL- PresDet- LinkState+ RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- PMEIntEna+ CRSVisible- RootCap: CRSVisible- RootSta: PME ReqID 0000, PMEStatus- PMEPending- DevCap2: Completion Timeout: Range ABC, TimeoutDis+, LTR+, OBFF Not Supported ARIFwd- DevCtl2: Completion Timeout: 50us to 50ms, TimeoutDis-, LTR+, OBFF Disabled ARIFwd- LnkCtl2: Target Link Speed: 5GT/s, EnterCompliance- SpeedDis- Transmit Margin: Normal Operating Range, EnterModifiedCompliance- ComplianceSOS- Compliance De-emphasis: -6dB LnkSta2: Current De-emphasis Level: -3.5dB, EqualizationComplete-, EqualizationPhase1- EqualizationPhase2-, EqualizationPhase3-, LinkEqualizationRequest- Capabilities: [80] MSI: Enable+ Count=1/1 Maskable- 64bit- Address: fee0f00c Data: 4191 Capabilities: [90] Subsystem: Dell Device 05ec Capabilities: [a0] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [100 v0] #00 Capabilities: [200 v1] L1 PM Substates L1SubCap: PCI-PM_L1.2+ PCI-PM_L1.1+ ASPM_L1.2+ ASPM_L1.1+ L1_PM_Substates+ PortCommonModeRestoreTime=40us PortTPowerOnTime=10us Kernel driver in use: pcieport 00:1d.0 USB controller: Intel Corporation Lynx Point-LP USB EHCI #1 (rev 04) (prog-if 20 [EHCI]) Subsystem: Dell Device 05ec Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 23 Region 0: Memory at b071c000 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst+ PME-Enable- DSel=0 DScale=0 PME- Capabilities: [58] Debug port: BAR=1 offset=00a0 Capabilities: [98] PCI Advanced Features AFCap: TP+ FLR+ AFCtrl: FLR- AFStatus: TP- Kernel driver in use: ehci-pci 00:1f.0 ISA bridge: Intel Corporation Lynx Point-LP LPC Controller (rev 04) Subsystem: Dell Device 05ec Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Capabilities: [e0] Vendor Specific Information: Len=0c <?> Kernel driver in use: lpc_ich 00:1f.2 SATA controller: Intel Corporation Lynx Point-LP SATA Controller 1 [AHCI mode] (rev 04) (prog-if 01 [AHCI 1.0]) Subsystem: Dell Device 05ec Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 60 Region 0: I/O ports at 4088 [size=8] Region 1: I/O ports at 4094 [size=4] Region 2: I/O ports at 4080 [size=8] Region 3: I/O ports at 4090 [size=4] Region 4: I/O ports at 4060 [size=32] Region 5: Memory at b071b000 (32-bit, non-prefetchable) [size=2K] Capabilities: [80] MSI: Enable+ Count=1/1 Maskable- 64bit- Address: fee0c00c Data: 41c1 Capabilities: [70] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold-) Status: D0 NoSoftRst+ PME-Enable- DSel=0 DScale=0 PME- Capabilities: [a8] SATA HBA v1.0 BAR4 Offset=00000004 Kernel driver in use: ahci 00:1f.3 SMBus: Intel Corporation Lynx Point-LP SMBus Controller (rev 04) Subsystem: Dell Device 05ec Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin C routed to IRQ 11 Region 0: Memory at b0719000 (64-bit, non-prefetchable) [size=256] Region 4: I/O ports at 4040 [size=32] 01:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8101E/RTL8102E PCI Express Fast Ethernet controller (rev 07) Subsystem: Dell Device 05ec Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 59 Region 0: I/O ports at 3000 [size=256] Region 2: Memory at b0600000 (64-bit, non-prefetchable) [size=4K] Region 4: Memory at b0400000 (64-bit, prefetchable) [size=16K] Capabilities: [40] Power Management version 3 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=375mA PME(D0+,D1+,D2+,D3hot+,D3cold+) Status: D0 NoSoftRst+ PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0800c Data: 41b1 Capabilities: [70] Express (v2) Endpoint, MSI 01 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s unlimited, L1 <64us ExtTag- AttnBtn- AttnInd- PwrInd- RBE+ FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd+ ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 512 bytes DevSta: CorrErr+ UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x1, ASPM L0s L1, Exit Latency L0s unlimited, L1 <64us ClockPM+ Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- DevCap2: Completion Timeout: Range ABCD, TimeoutDis+, LTR+, OBFF Via message/WAKE# DevCtl2: Completion Timeout: 50us to 50ms, TimeoutDis-, LTR+, OBFF Disabled LnkCtl2: Target Link Speed: 2.5GT/s, EnterCompliance- SpeedDis- Transmit Margin: Normal Operating Range, EnterModifiedCompliance- ComplianceSOS- Compliance De-emphasis: -6dB LnkSta2: Current De-emphasis Level: -6dB, EqualizationComplete-, EqualizationPhase1- EqualizationPhase2-, EqualizationPhase3-, LinkEqualizationRequest- Capabilities: [b0] MSI-X: Enable- Count=4 Masked- Vector table: BAR=4 offset=00000000 PBA: BAR=4 offset=00000800 Capabilities: [d0] Vital Product Data No end tag found Capabilities: [100 v1] Advanced Error Reporting UESta: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UEMsk: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UESvrt: DLP+ SDES+ TLP- FCP+ CmpltTO- CmpltAbrt- UnxCmplt- RxOF+ MalfTLP+ ECRC- UnsupReq- ACSViol- CESta: RxErr+ BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr- CEMsk: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr+ AERCap: First Error Pointer: 00, GenCap+ CGenEn- ChkCap+ ChkEn- Capabilities: [140 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed- WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=ff Status: NegoPending- InProgress- Capabilities: [160 v1] Device Serial Number 01-00-00-00-36-4c-e0-00 Capabilities: [170 v1] Latency Tolerance Reporting Max snoop latency: 0ns Max no snoop latency: 0ns Kernel driver in use: r8169 02:00.0 Network controller: Intel Corporation Wireless 7260 (rev 73) Subsystem: Intel Corporation Wireless-N 7260 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 62 Region 0: Memory at b0500000 (64-bit, non-prefetchable) [size=8K] Capabilities: [c8] Power Management version 3 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [d0] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0400c Data: 41e1 Capabilities: [40] Express (v2) Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <512ns, L1 unlimited ExtTag- AttnBtn- AttnInd- PwrInd- RBE+ FLReset+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd+ ExtTag- PhantFunc- AuxPwr+ NoSnoop+ FLReset- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x1, ASPM L0s L1, Exit Latency L0s <4us, L1 <32us ClockPM+ Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- DevCap2: Completion Timeout: Range B, TimeoutDis+, LTR+, OBFF Via WAKE# DevCtl2: Completion Timeout: 16ms to 55ms, TimeoutDis-, LTR+, OBFF Disabled LnkCtl2: Target Link Speed: 2.5GT/s, EnterCompliance- SpeedDis- Transmit Margin: Normal Operating Range, EnterModifiedCompliance- ComplianceSOS- Compliance De-emphasis: -6dB LnkSta2: Current De-emphasis Level: -3.5dB, EqualizationComplete-, EqualizationPhase1- EqualizationPhase2-, EqualizationPhase3-, LinkEqualizationRequest- Capabilities: [100 v1] Advanced Error Reporting UESta: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UEMsk: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UESvrt: DLP+ SDES+ TLP- FCP+ CmpltTO- CmpltAbrt- UnxCmplt- RxOF+ MalfTLP+ ECRC- UnsupReq- ACSViol- CESta: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr- CEMsk: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr+ AERCap: First Error Pointer: 00, GenCap- CGenEn- ChkCap- ChkEn- Capabilities: [140 v1] Device Serial Number fc-f8-ae-ff-ff-a8-4f-9e Capabilities: [14c v1] Latency Tolerance Reporting Max snoop latency: 0ns Max no snoop latency: 0ns Capabilities: [154 v1] Vendor Specific Information: ID=cafe Rev=1 Len=014 <?> Kernel driver in use: iwlwifi [7.6.] SCSI information (from /proc/scsi/scsi) Attached devices: Host: scsi0 Channel: 00 Id: 00 Lun: 00 Vendor: ATA Model: WDC WD10JPVX-75J Rev: 01.0 Type: Direct-Access ANSI SCSI revision: 05 Host: scsi1 Channel: 00 Id: 00 Lun: 00 Vendor: HL-DT-ST Model: DVD+-RW GU90N Rev: A100 Type: CD-ROM ANSI SCSI revision: 05 Host: scsi3 Channel: 00 Id: 00 Lun: 00 Vendor: Generic- Model: xD/SD/M.S. Rev: 1.00 Type: Direct-Access ANSI SCSI revision: 02 [7.7.] Other information that might be relevant to the problem (please look in /proc and include all information that you think to be relevant): 1 1136 14 149 1523 1631 22 2596 3 39 480 52 63 78 9 cpuinfo ioports meminfo slabinfo vmallocinfo 10 1144 1444 1494 1528 1640 23 26 301 40 489 53 642 787 907 crypto irq misc softirqs vmstat 100 1193 1450 1497 153 17 2323 2675 31 41 49 54 7 79 919 devices kallsyms modules stat zoneinfo 101 12 1460 15 1538 175 2357 2681 32 42 490 55 707 791 936 diskstats kcore mounts swaps 102 1209 1461 150 1541 176 2358 2682 33 43 5 56 741 8 979 dma key-users mtrr sys 103 1223 1463 1505 1562 18 2365 2683 339 436 50 564 743 81 acpi driver kmsg net sysrq-trigger 1091 13 1464 1508 1591 19 24 27 34 44 504 57 75 819 asound execdomains kpagecount pagetypeinfo sysvipc 11 1300 147 151 1597 1908 2400 28 35 45 505 572 755 848 buddyinfo fb kpageflags partitions timer_list 1109 1338 1470 1511 16 1980 2403 29 36 46 506 576 76 852 bus filesystems latency_stats sched_debug timer_stats 1113 1340 1474 1515 1601 2 2486 293 37 47 507 59 765 857 cgroups fs loadavg schedstat tty 1127 1380 1478 152 1607 20 25 2954 371 478 51 61 766 858 cmdline interrupts locks scsi uptime 1132 1395 148 1520 1625 21 2575 2973 38 48 511 62 77 861 consoles iomem mdstat self version [X.] Other notes, patches, fixes, workarounds: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1324330

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[alsa-devel] [PATCH 1/3] ASoC: rt5640: Add RL6231 class device shared support for RT5640, RT5645 and RT5651
by Oder Chiou 01 Jun '14

01 Jun '14

The patch adds the RL6231 class device shared support for RT5640, RT5645 and RT5651. The function of the DMIC clock calculation can be shared by RL6231 shared support. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 9 +++++++ sound/soc/codecs/Makefile | 2 ++ sound/soc/codecs/rl6231.c | 67 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rl6231.h | 18 +++++++++++++ sound/soc/codecs/rt5640.c | 23 ++++------------ sound/soc/codecs/rt5645.c | 22 +++------------- sound/soc/codecs/rt5651.c | 22 ++++------------ 7 files changed, 110 insertions(+), 53 deletions(-) create mode 100644 sound/soc/codecs/rl6231.c create mode 100644 sound/soc/codecs/rl6231.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4f96b97..6951381 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -402,6 +402,15 @@ config SND_SOC_PCM512x_SPI select SND_SOC_PCM512x select REGMAP_SPI +config SND_SOC_RL6231 + tristate + default y if SND_SOC_RT5640=y + default y if SND_SOC_RT5645=y + default y if SND_SOC_RT5651=y + default m if SND_SOC_RT5640=m + default m if SND_SOC_RT5645=m + default m if SND_SOC_RT5651=m + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1ccdaf0..d1e913a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -60,6 +60,7 @@ snd-soc-pcm3008-objs := pcm3008.o snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o +snd-soc-rl6231-objs := rl6231.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o @@ -216,6 +217,7 @@ obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o +obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c new file mode 100644 index 0000000..13ac551 --- /dev/null +++ b/sound/soc/codecs/rl6231.c @@ -0,0 +1,67 @@ +/* + * rl6231.c - RL6231 class device shared support + * + * Copyright 2014 Realtek Semiconductor Corp. + * + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rl6231.h" + +/** + * rl6231_calc_dmic_clk - Calculate the parameter of dmic. + * + * @rate: base clock rate. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +int rl6231_calc_dmic_clk(int rate) +{ + int div[] = {2, 3, 4, 6, 8, 12}, idx = -EINVAL; + int i, red, bound, temp; + + red = 3000000 * 12; + for (i = 0; i < ARRAY_SIZE(div); i++) { + bound = div[i] * 3000000; + if (rate > bound) + continue; + temp = bound - rate; + if (temp < red) { + red = temp; + idx = i; + } + } + + return idx; +} +EXPORT_SYMBOL_GPL(rl6231_calc_dmic_clk); + +MODULE_DESCRIPTION("RL6231 class device shared support"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rl6231.h b/sound/soc/codecs/rl6231.h new file mode 100644 index 0000000..00032db --- /dev/null +++ b/sound/soc/codecs/rl6231.h @@ -0,0 +1,18 @@ +/* + * rl6231.h - RL6231 class device shared support + * + * Copyright 2014 Realtek Semiconductor Corp. + * + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RL6231_H__ +#define __RL6231_H__ + +int rl6231_calc_dmic_clk(int rate); + +#endif /* __RL6231_H__ */ diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 9e0d48f..0cefc16 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -31,6 +31,7 @@ #include <sound/initval.h> #include <sound/tlv.h> +#include "rl6231.h" #include "rt5640.h" #define RT5640_DEVICE_ID 0x6231 @@ -453,30 +454,16 @@ static const struct snd_kcontrol_new rt5640_specific_snd_controls[] = { * @kcontrol: The kcontrol of this widget. * @event: Event id. * - * Choose dmic clock between 1MHz and 3MHz. - * It is better for clock to approximate 3MHz. */ static int set_dmic_clk(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - int div[] = {2, 3, 4, 6, 8, 12}; - int idx = -EINVAL, i; - int rate, red, bound, temp; - - rate = rt5640->sysclk; - red = 3000000 * 12; - for (i = 0; i < ARRAY_SIZE(div); i++) { - bound = div[i] * 3000000; - if (rate > bound) - continue; - temp = bound - rate; - if (temp < red) { - red = temp; - idx = i; - } - } + int idx = -EINVAL; + + idx = rl6231_calc_dmic_clk(rt5640->sysclk); + if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); else diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index ab97d72..caa5519 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -26,6 +26,7 @@ #include <sound/initval.h> #include <sound/tlv.h> +#include "rl6231.h" #include "rt5645.h" #define RT5645_DEVICE_ID 0x6308 @@ -519,30 +520,15 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { * @kcontrol: The kcontrol of this widget. * @event: Event id. * - * Choose dmic clock between 1MHz and 3MHz. - * It is better for clock to approximate 3MHz. */ static int set_dmic_clk(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - int div[] = {2, 3, 4, 6, 8, 12}; - int idx = -EINVAL, i; - int rate, red, bound, temp; - - rate = rt5645->sysclk; - red = 3000000 * 12; - for (i = 0; i < ARRAY_SIZE(div); i++) { - bound = div[i] * 3000000; - if (rate > bound) - continue; - temp = bound - rate; - if (temp < red) { - red = temp; - idx = i; - } - } + int idx = -EINVAL; + + idx = rl6231_calc_dmic_clk(rt5645->sysclk); if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 9c88d89..7a7bec6 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -26,6 +26,7 @@ #include <sound/initval.h> #include <sound/tlv.h> +#include "rl6231.h" #include "rt5651.h" #define RT5651_DEVICE_ID_VALUE 0x6281 @@ -371,29 +372,16 @@ static const struct snd_kcontrol_new rt5651_snd_controls[] = { * @kcontrol: The kcontrol of this widget. * @event: Event id. * - * Choose dmic clock between 1MHz and 3MHz. - * It is better for clock to approximate 3MHz. */ static int set_dmic_clk(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); - int div[] = {2, 3, 4, 6, 8, 12}, idx = -EINVAL; - int i, rate, red, bound, temp; - - rate = rt5651->sysclk; - red = 3000000 * 12; - for (i = 0; i < ARRAY_SIZE(div); i++) { - bound = div[i] * 3000000; - if (rate > bound) - continue; - temp = bound - rate; - if (temp < red) { - red = temp; - idx = i; - } - } + int idx = -EINVAL; + + idx = rl6231_calc_dmic_clk(rt5651->sysclk); + if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); else -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH] ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error.
by Xiubo Li 01 Jun '14

01 Jun '14

Since we cannot make sure the 'reg_size' will always be none zero here, and then if 'reg_size' equals to zero, the kzalloc() will return ZERO_SIZE_PTR, which equals to ((void *)16). So this patch fix this with just doing the 'reg_size' zero check before calling kzalloc(). Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/soc-cache.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 3fa77d5..8fff5b6 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -72,6 +72,9 @@ int snd_soc_cache_init(struct snd_soc_codec *codec) reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; + if (!reg_size) + return -EINVAL; + mutex_init(&codec->cache_rw_mutex); dev_dbg(codec->dev, "ASoC: Initializing cache for %s codec\n", -- 1.8.4

2 1

[alsa-devel] [PATCH] sound: pci: ice1712: wm8766.c: Cleaning up uninitialized variables
by Rickard Strandqvist 01 Jun '14

01 Jun '14

There is a risk that the variable will be used without being initialized. This was largely found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/ice1712/wm8766.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ice1712/wm8766.c b/sound/pci/ice1712/wm8766.c index 21b373b..788a31c 100644 --- a/sound/pci/ice1712/wm8766.c +++ b/sound/pci/ice1712/wm8766.c @@ -237,7 +237,7 @@ static int snd_wm8766_ctl_get(struct snd_kcontrol *kcontrol, { struct snd_wm8766 *wm = snd_kcontrol_chip(kcontrol); int n = kcontrol->private_value; - u16 val1, val2; + u16 val1 = 0, val2 = 0; if (wm->ctl[n].get) wm->ctl[n].get(wm, &val1, &val2); -- 1.7.10.4

2 1

[alsa-devel] [PATCH] bebob: Use ARRAY_SIZE() instead of sizeof() for return condition
by Takashi Sakamoto 01 Jun '14

01 Jun '14

The parser for stream format information should return when detecting wrong value for sampling rate. But it doesn't. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 514c7c9..bc4f827 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -749,7 +749,7 @@ parse_stream_formation(u8 *buf, unsigned int len, if (buf[2] == bridgeco_freq_table[i]) break; } - if (i == sizeof(bridgeco_freq_table)) + if (i == ARRAY_SIZE(bridgeco_freq_table)) return -ENOSYS; /* Avoid double count by different entries for the same rate. */ -- 1.8.3.2

2 3

[alsa-devel] [RFC] [PATCH] firewire-lib: Use IEC 61883-6 compliant labels for Raw Audio/
by Takashi Sakamoto 01 Jun '14

01 Jun '14

Hi Clemens, Would I request you to comment this patch? I don't touch this label issue with my patchset because there is a bit possibility to cause a regression. But now, I have a confidence that this modification is reasonable. In detail, please see an attached patch. Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

3 4

[alsa-devel] [PATCH] sound: pci: ice1712: wm8776.c: Cleaning up uninitialized variables
by Rickard Strandqvist 01 Jun '14

01 Jun '14

There is a risk that the variable will be used without being initialized. This was largely found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/ice1712/wm8776.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ice1712/wm8776.c b/sound/pci/ice1712/wm8776.c index e66c0da..52da0ff 100644 --- a/sound/pci/ice1712/wm8776.c +++ b/sound/pci/ice1712/wm8776.c @@ -510,7 +510,7 @@ static int snd_wm8776_ctl_get(struct snd_kcontrol *kcontrol, { struct snd_wm8776 *wm = snd_kcontrol_chip(kcontrol); int n = kcontrol->private_value; - u16 val1, val2; + u16 val1 = 0, val2 = 0; if (wm->ctl[n].get) wm->ctl[n].get(wm, &val1, &val2); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ALSA: ASoC: wm9713: correct mono out PGA sources
by Matt Reimer 01 Jun '14

01 Jun '14

The mono output PGA input only has four possible sources, so omit the rest. Signed-off-by: Matt Reimer <mreimer(a)sdgsystems.com> --- sound/soc/codecs/wm9713.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 5f69b52..caf0269 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -74,8 +74,7 @@ static const char *wm9713_rec_src[] = "Mono Out", "Zh"}; static const char *wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; static const char *wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"}; -static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv", - "Mono Vmid", "Inv Vmid"}; +static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv"}; static const char *wm9713_spk_pga[] = {"Vmid", "Zh", "Headphone", "Speaker", "Inv", "Headphone Vmid", "Speaker Vmid", "Inv Vmid"}; -- 1.7.5.4

3 4

[alsa-devel] [PATCH] sound: synth: emux: soundfont.c: Cleaning up memory leak
by Rickard Strandqvist 01 Jun '14

01 Jun '14

There is a risk for memory leak in when something unexpected happens and the function returns. This was largely found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/synth/emux/soundfont.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 1137b85..09c5eeb 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -1021,6 +1021,7 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, data, count); if (rc < 0) { sf_sample_delete(sflist, sf, smp); + free(zone); return rc; } /* memory offset is updated after */ -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320
by Alexander Shiyan 01 Jun '14

01 Jun '14

Eukrea-i.MX51 board was converted to use DT, ie we no longer have a MACH_EUKREA_MBIMXSD51_BASEBOARD symbol. Transformation of other boards planned for the near future, so this patch removes all these dependencies and restricts build of this driver to ARCH_MXC. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/Kconfig | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index d262ec0..2989319 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -207,12 +207,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on MACH_EUKREA_MBIMX27_BASEBOARD \ - || MACH_EUKREA_MBIMXSD25_BASEBOARD \ - || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD \ - || (OF && ARM) - depends on I2C + depends on ARCH_MXC && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI -- 1.8.3.2

2 2

[alsa-devel] [PATCH v5 0/8] ASoC: fsl-ssi: Driver cleanup
by Markus Pargmann 01 Jun '14

01 Jun '14

Hi, These are the remaining patches of this fsl-ssi cleanup series. v5 adds the possibility to set the bitclock frequency using .set_dai_sysclk as suggested. If it is not explicitly set, the bitclock is calculated using the hw_params. I also added the missing signed-off-by tags to the patches. Best Regards, Markus Markus Pargmann (4): ASoC: fsl-ssi: make fsl,mode property optional ASoC: fsl-ssi: Fix baudclock handling ASoC: fsl-ssi: reorder and document fsl_ssi_private ASoC: fsl-ssi: Use regmap Sascha Hauer (4): ASoC: fsl-ssi: introduce SoC specific data ASoC: fsl-ssi: set bitclock in master mode from hw_params ASoC: fsl-ssi: remove unnecessary spinlock ASoC: fsl-ssi: Set framerate divider correctly for i2s master mode sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_ssi.c | 574 ++++++++++++++++++++++++++++-------------------- sound/soc/fsl/fsl_ssi.h | 50 ++--- 3 files changed, 360 insertions(+), 265 deletions(-) -- 2.0.0.rc2

2 9

[alsa-devel] [PATCH 1/4] ASoC: Intel: Clear stored Baytrail DSP DMA pointer before stream start
by Jarkko Nikula 01 Jun '14

01 Jun '14

Stored DSP DMA pointer must be cleared before starting the stream since PCM pointer callback sst_byt_pcm_pointer() can be called before pointer is updated. In that case last position of previous stream was wronly returned. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 3af38576e91e..8eab97368ea7 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -180,6 +180,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: + pcm_data->hw_ptr = 0; sst_byt_stream_start(byt, pcm_data->stream, 0); break; case SNDRV_PCM_TRIGGER_RESUME: -- 2.0.0.rc2

2 5

[alsa-devel] [PATCH v3] ASoC: cs42l56: Fix new value argument in snd_soc_update_bits calls
by Axel Lin 01 Jun '14

01 Jun '14

The new value argument needs proper shift to match the mask bit fields. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- v3: Define CS42L56_MUTE_ALL as 0xff, so all the mask bits will be set. v2: Address Brian's comment in v1. sound/soc/codecs/cs42l56.c | 70 ++++++++++++++++++++-------------------------- sound/soc/codecs/cs42l56.h | 10 ++++--- 2 files changed, 37 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 5bb134b..fdc4bd2 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -763,14 +763,14 @@ static int cs42l56_set_sysclk(struct snd_soc_dai *codec_dai, case CS42L56_MCLK_11P2896MHZ: case CS42L56_MCLK_12MHZ: case CS42L56_MCLK_12P288MHZ: - cs42l56->mclk_div2 = 1; + cs42l56->mclk_div2 = CS42L56_MCLK_DIV2; cs42l56->mclk_prediv = 0; break; case CS42L56_MCLK_22P5792MHZ: case CS42L56_MCLK_24MHZ: case CS42L56_MCLK_24P576MHZ: - cs42l56->mclk_div2 = 1; - cs42l56->mclk_prediv = 1; + cs42l56->mclk_div2 = CS42L56_MCLK_DIV2; + cs42l56->mclk_prediv = CS42L56_MCLK_PREDIV; break; default: return -EINVAL; @@ -844,57 +844,49 @@ static int cs42l56_digital_mute(struct snd_soc_dai *dai, int mute) /* Hit the DSP Mixer first */ snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_MUTE); + CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | + CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | + CS42L56_MSTA_MUTE_MASK, + CS42L56_MUTE_ALL); /* Mute ADC's */ snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_MUTE); + CS42L56_ADCA_MUTE_MASK | + CS42L56_ADCB_MUTE_MASK, + CS42L56_MUTE_ALL); /* HP And LO */ snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_MUTE_ALL); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_MUTE_ALL); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); + CS42L56_LO_MUTE_MASK, CS42L56_MUTE_ALL); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); - - + CS42L56_LO_MUTE_MASK, CS42L56_MUTE_ALL); } else { snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | + CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | + CS42L56_MSTA_MUTE_MASK, + CS42L56_UNMUTE); + snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_ADCA_MUTE_MASK | + CS42L56_ADCB_MUTE_MASK, + CS42L56_UNMUTE); + snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, CS42L56_UNMUTE); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, CS42L56_UNMUTE); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, CS42L56_UNMUTE); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, CS42L56_UNMUTE); } return 0; } diff --git a/sound/soc/codecs/cs42l56.h b/sound/soc/codecs/cs42l56.h index ad2b50a..5025ec9 100644 --- a/sound/soc/codecs/cs42l56.h +++ b/sound/soc/codecs/cs42l56.h @@ -80,19 +80,21 @@ #define CS42L56_PDN_HPB_MASK 0xc0 /* serial port and clk masks */ -#define CS42L56_MASTER_MODE 1 +#define CS42L56_MASTER_MODE 0x40 #define CS42L56_SLAVE_MODE 0 #define CS42L56_MS_MODE_MASK 0x40 -#define CS42L56_SCLK_INV 1 +#define CS42L56_SCLK_INV 0x20 #define CS42L56_SCLK_INV_MASK 0x20 #define CS42L56_SCLK_MCLK_MASK 0x18 +#define CS42L56_MCLK_PREDIV 0x04 #define CS42L56_MCLK_PREDIV_MASK 0x04 +#define CS42L56_MCLK_DIV2 0x02 #define CS42L56_MCLK_DIV2_MASK 0x02 #define CS42L56_MCLK_DIS_MASK 0x01 #define CS42L56_CLK_AUTO_MASK 0x20 #define CS42L56_CLK_RATIO_MASK 0x1f #define CS42L56_DIG_FMT_I2S 0 -#define CS42L56_DIG_FMT_LEFT_J 1 +#define CS42L56_DIG_FMT_LEFT_J 0x08 #define CS42L56_DIG_FMT_MASK 0x08 /* Class H and misc ctl masks */ @@ -116,7 +118,7 @@ #define CS42L56_DEEMPH_MASK 0x40 #define CS42L56_PLYBCK_GANG_MASK 0x10 #define CS42L56_PCM_INV_MASK 0x0c -#define CS42L56_MUTE 1 +#define CS42L56_MUTE_ALL 0xff #define CS42L56_UNMUTE 0 #define CS42L56_ADCAMIX_MUTE_MASK 0x40 #define CS42L56_ADCBMIX_MUTE_MASK 0x80 -- 1.8.3.2

3 2

[alsa-devel] [PATCHv2] ASoC: Intel: Add Baytrail byt-max98090 machine driver
by Jarkko Nikula 01 Jun '14

01 Jun '14

Add machine driver and ACPI probing for Baytrail SST with MAX98090 codec. Jack detect code from Kevin Strasser <kevin.strasser(a)intel.com>, GPIO resolving from Mika Westerberg <mika.westerberg(a)linux.intel.com> and fixes and cleanups from Liam Girdwood <liam.r.girdwood(a)linux.intel.com>. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- This is for for-next branch since depends on commit f025d3b9c64e("ASoC: jack: Add support for GPIO descriptor defined jack pins"). Or merge topic/gpio or for-next to topic/intel before applying. v2: - Needless DAPM calls removed from init - Using snd_soc_pm_ops for suspend/resume - Using new GPIO descriptors support from ASoC jack - devm_snd_soc_register_card - Bogus Kconfig symbol removed from Makefile - Various cleanups --- sound/soc/intel/Kconfig | 9 ++ sound/soc/intel/Makefile | 2 + sound/soc/intel/byt-max98090.c | 203 +++++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-acpi.c | 1 + 4 files changed, 215 insertions(+) create mode 100644 sound/soc/intel/byt-max98090.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 3c81b3891209..cd3498736e91 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -49,3 +49,12 @@ config SND_SOC_INTEL_BYT_RT5640_MACH help This adds audio driver for Intel Baytrail platform based boards with the RT5640 audio codec. + +config SND_SOC_INTEL_BYT_MAX98090_MACH + tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + select SND_SOC_INTEL_BAYTRAIL + select SND_SOC_MAX98090 + help + This adds audio driver for Intel Baytrail platform based boards + with the MAX98090 audio codec. diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 0db4e2f336dc..4bfca79a42ba 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -23,6 +23,8 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o # Machine support snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o +snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o +obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c new file mode 100644 index 000000000000..5fc98c64a3f4 --- /dev/null +++ b/sound/soc/intel/byt-max98090.c @@ -0,0 +1,203 @@ +/* + * Intel Baytrail SST MAX98090 machine driver + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/acpi.h> +#include <linux/device.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../codecs/max98090.h" + +struct byt_max98090_private { + struct snd_soc_jack jack; +}; + +static const struct snd_soc_dapm_widget byt_max98090_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route byt_max98090_audio_map[] = { + {"IN34", NULL, "Headset Mic"}, + {"IN34", NULL, "MICBIAS"}, + {"MICBIAS", NULL, "Headset Mic"}, + {"DMICL", NULL, "Int Mic"}, + {"Headphone", NULL, "HPL"}, + {"Headphone", NULL, "HPR"}, + {"Ext Spk", NULL, "SPKL"}, + {"Ext Spk", NULL, "SPKR"}, +}; + +static const struct snd_kcontrol_new byt_max98090_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Ext Spk", + .mask = SND_JACK_LINEOUT, + }, + { + .pin = "Int Mic", + .mask = SND_JACK_LINEIN, + }, +}; + +static struct snd_soc_jack_gpio hs_jack_gpios[] = { + { + .name = "hp-gpio", + .idx = 0, + .report = SND_JACK_HEADPHONE | SND_JACK_LINEOUT, + .debounce_time = 200, + }, + { + .name = "mic-gpio", + .idx = 1, + .report = SND_JACK_MICROPHONE | SND_JACK_LINEIN, + .debounce_time = 200, + }, +}; + +static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_card *card = runtime->card; + struct byt_max98090_private *drv = snd_soc_card_get_drvdata(card); + struct snd_soc_jack *jack = &drv->jack; + + card->dapm.idle_bias_off = true; + + ret = snd_soc_dai_set_sysclk(runtime->codec_dai, + M98090_REG_SYSTEM_CLOCK, + 25000000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "Can't set codec clock %d\n", ret); + return ret; + } + + /* Enable jack detection */ + ret = snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpiods(card->dev->parent, jack, + ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + if (ret) + return ret; + + return max98090_mic_detect(codec, jack); +} + +static struct snd_soc_dai_link byt_max98090_dais[] = { + { + .name = "Baytrail Audio", + .stream_name = "Audio", + .cpu_dai_name = "baytrail-pcm-audio", + .codec_dai_name = "HiFi", + .codec_name = "i2c-193C9890:00", + .platform_name = "baytrail-pcm-audio", + .init = byt_max98090_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card byt_max98090_card = { + .name = "byt-max98090", + .dai_link = byt_max98090_dais, + .num_links = ARRAY_SIZE(byt_max98090_dais), + .dapm_widgets = byt_max98090_widgets, + .num_dapm_widgets = ARRAY_SIZE(byt_max98090_widgets), + .dapm_routes = byt_max98090_audio_map, + .num_dapm_routes = ARRAY_SIZE(byt_max98090_audio_map), + .controls = byt_max98090_controls, + .num_controls = ARRAY_SIZE(byt_max98090_controls), +}; + +static int byt_max98090_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct byt_max98090_private *priv; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC); + if (!priv) { + dev_err(&pdev->dev, "allocation failed\n"); + return -ENOMEM; + } + + byt_max98090_card.dev = &pdev->dev; + snd_soc_card_set_drvdata(&byt_max98090_card, priv); + ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_max98090_card); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + + return ret_val; +} + +static int byt_max98090_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct byt_max98090_private *priv = snd_soc_card_get_drvdata(card); + + snd_soc_jack_free_gpios(&priv->jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + + return 0; +} + +static struct platform_driver byt_max98090_driver = { + .probe = byt_max98090_probe, + .remove = byt_max98090_remove, + .driver = { + .name = "byt-max98090", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, +}; +module_platform_driver(byt_max98090_driver) + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver"); +MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:byt-max98090"); diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index 18aee77f8d4a..42edc6f4fc4a 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -247,6 +247,7 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { static struct sst_acpi_mach baytrail_machines[] = { { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" }, + { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-i2s_master" }, {} }; -- 2.0.0.rc2

2 1

[alsa-devel] [PATCH] ASoC: Intel: byt/hsw: Add missing kthread_stop to error/cleanup path
by Jarkko Nikula 01 Jun '14

01 Jun '14

From: Imre Deak <imre.deak(a)intel.com> Baytrail and Haswell SST IPC don't stop the kernel thread in error and cleanup path thus leaving orphan kernel thread behind in such a case. Also while at it, fix one error path in sst-haswell-ipc.c that doesn't free hsw->msg. [Jarkko: I edited the commit log a little] Signed-off-by: Imre Deak <imre.deak(a)intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-ipc.c | 5 ++++- sound/soc/intel/sst-haswell-ipc.c | 12 ++++++++---- 2 files changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index 0d31dbbf4806..d4b8c827c8b8 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -824,7 +824,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) byt->dsp = sst_dsp_new(dev, &byt_dev, pdata); if (byt->dsp == NULL) { err = -ENODEV; - goto err_free_msg; + goto dsp_err; } /* keep the DSP in reset state for base FW loading */ @@ -856,6 +856,8 @@ boot_err: sst_fw_free(byt_sst_fw); fw_err: sst_dsp_free(byt->dsp); +dsp_err: + kthread_stop(byt->tx_thread); err_free_msg: kfree(byt->msg); @@ -870,6 +872,7 @@ void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata) sst_dsp_reset(byt->dsp); sst_fw_free_all(byt->dsp); sst_dsp_free(byt->dsp); + kthread_stop(byt->tx_thread); kfree(byt->msg); } EXPORT_SYMBOL_GPL(sst_byt_dsp_free); diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index e7996b39a484..bee192d42032 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1730,7 +1730,7 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) ret = msg_empty_list_init(hsw); if (ret < 0) - goto list_err; + return -ENOMEM; /* start the IPC message thread */ init_kthread_worker(&hsw->kworker); @@ -1740,7 +1740,7 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) if (IS_ERR(hsw->tx_thread)) { ret = PTR_ERR(hsw->tx_thread); dev_err(hsw->dev, "error: failed to create message TX task\n"); - goto list_err; + goto err_free_msg; } init_kthread_work(&hsw->kwork, ipc_tx_msgs); @@ -1750,7 +1750,7 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) hsw->dsp = sst_dsp_new(dev, &hsw_dev, pdata); if (hsw->dsp == NULL) { ret = -ENODEV; - goto list_err; + goto dsp_err; } /* keep the DSP in reset state for base FW loading */ @@ -1794,8 +1794,11 @@ boot_err: sst_fw_free(hsw_sst_fw); fw_err: sst_dsp_free(hsw->dsp); +dsp_err: + kthread_stop(hsw->tx_thread); +err_free_msg: kfree(hsw->msg); -list_err: + return ret; } EXPORT_SYMBOL_GPL(sst_hsw_dsp_init); @@ -1808,6 +1811,7 @@ void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata) sst_fw_free_all(hsw->dsp); sst_dsp_free(hsw->dsp); kfree(hsw->scratch); + kthread_stop(hsw->tx_thread); kfree(hsw->msg); } EXPORT_SYMBOL_GPL(sst_hsw_dsp_free); -- 2.0.0.rc2

2 1

[alsa-devel] [PATCH] ASoC: tlv320aci3x: Fix custom snd_soc_dapm_put_volsw_aic3x() function
by Peter Ujfalusi 01 Jun '14

01 Jun '14

For some unknown reason the parameters for snd_soc_test_bits() were in wrong order: It was: snd_soc_test_bits(codec, val, mask, reg); /* WRONG!!! */ while it should be: snd_soc_test_bits(codec, reg, mask, val); Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, so the failures I was seeing was indeed caused by something one never expects: wrong parameter order for snd_soc_test_bits() in the tlv320aic3x driver, so regmap failed time to time when we asked it to read a register out of the valid range, since the value was sent as register number :o Regards, Peter sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d7349bc89ad3..e12fafbb1e09 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -169,7 +169,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, mask <<= shift; val <<= shift; - change = snd_soc_test_bits(codec, val, mask, reg); + change = snd_soc_test_bits(codec, reg, mask, val); if (change) { update.kcontrol = kcontrol; update.reg = reg; -- 1.9.3

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs4265: bindings: sound: Add bindings for CS4265 CODEC
by Paul Handrigan 01 Jun '14

01 Jun '14

This patch adds binding documentation for the Cirrus Logic CS4265 CODEC. Signed-off-by: Paul Handrigan <paul.handrigan(a)cirrus.com> --- Documentation/devicetree/bindings/sound/cs4265.txt | 29 ++++++++++++++++++++++ 1 file changed, 29 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4265.txt diff --git a/Documentation/devicetree/bindings/sound/cs4265.txt b/Documentation/devicetree/bindings/sound/cs4265.txt new file mode 100644 index 0000000..03d0dde --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4265.txt @@ -0,0 +1,29 @@ +CS4265 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "cirrus,cs4265" + + - reg : the I2C address of the device for I2C. The I2C address depends on + the state of the AD0 pin. If AD0 is high, the i2c address is 0x4f. + If it is low, the i2c address is 0x4e. + +Optional properties: + + - reset-gpio : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + +Examples: + +codec_ad0_high: cs4265@4f { /* AD0 Pin is high */ + compatible = "cirrus,cs4265"; + reg = <0x4f>; +}; + + +codec_ad0_low: cs4265@4e { /* AD0 Pin is low */ + compatible = "cirrus,cs4265"; + reg = <0x4e>; +}; -- 1.9.1

4 7

[alsa-devel] Bug report: [alsa-sink-USB Audio] alsa-sink.c: Most likely this is a bug in the ALSA driver '(null)'. Please report this issue to the ALSA
by Justin Piszcz 31 May '14

31 May '14

Hi, Bug report: ==> /var/log/user.log <== May 25 06:38:06 atom pulseaudio[6754]: [alsa-sink-USB Audio] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! May 25 06:38:06 atom pulseaudio[6754]: [alsa-sink-USB Audio] alsa-sink.c: Most likely this is a bug in the ALSA driver '(null)'. Please report this issue to the ALSA developers. May 25 06:38:06 atom pulseaudio[6754]: [alsa-sink-USB Audio] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. $ uname -a Linux atom 3.14.0 #2 SMP Wed Apr 2 06:30:21 EDT 2014 x86_64 GNU/Linux $ lsusb Bus 002 Device 004: ID 0764:0501 Cyber Power System, Inc. CP1500 AVR UPS Bus 002 Device 008: ID 0baf:0303 U.S. Robotics Bus 002 Device 007: ID 067b:2303 Prolific Technology, Inc. PL2303 Serial Port Bus 002 Device 010: ID 0461:4d15 Primax Electronics, Ltd Dell Optical Mouse Bus 002 Device 009: ID 413c:2002 Dell Computer Corp. SK-8125 Keyboard Bus 002 Device 006: ID 413c:1002 Dell Computer Corp. Keyboard Hub Bus 002 Device 005: ID 0d8c:0008 C-Media Electronics, Inc. Bus 002 Device 003: ID 2001:f103 D-Link Corp. DUB-H7 7-port USB 2.0 hub Bus 002 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 001 Device 003: ID 0557:2221 ATEN International Co., Ltd Winbond Hermon Bus 001 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Justin.

2 5

[alsa-devel] [PATCH] ALSA: Docs: Fix typo in tegra_hda doc.
by Dylan Reid 31 May '14

31 May '14

The correct name of the third clock is hda2codec_2x. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt index c26f8ca..b4730c2 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt @@ -20,9 +20,9 @@ hda@0,70030000 { clocks = <&tegra_car TEGRA124_CLK_HDA>, <&tegra_car TEGRA124_CLK_HDA2HDMI>, <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>; - clock-names = "hda", "hda2hdmi", "hdacodec_2x"; + clock-names = "hda", "hda2hdmi", "hda2codec_2x"; resets = <&tegra_car 125>, /* hda */ <&tegra_car 128>; /* hda2hdmi */ <&tegra_car 111>, /* hda2codec_2x */ - reset-names = "hda", "hda2hdmi", "hdacodec_2x"; + reset-names = "hda", "hda2hdmi", "hda2codec_2x"; }; -- 1.8.1.3.605.g02339dd

2 1

[alsa-devel] [PATCH 00/49 v4] Enhancement for support of Firewire devices
by Takashi Sakamoto 31 May '14

31 May '14

This 49 patchset is to update previous series: [alsa-devel] [PATCH 00/44 v3] Enhancement of support for Firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074624.html 4 patches are newly added: [PATCH 09/48] dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE - Make firewire-lib simpler. [PATCH 10/48] firewire-lib/dice/speakers: Add common PCM constraints for AMDTP streams - To share codes for drivers. Currently no improvement. [PATCH 25/48] fireworks/firewire-lib: Add a quirk to reset data block counter at bus reset [PATCH 46/48] bebob: Add a quirk of data blocks for MIDI messages for some M-Audio devices - New quirks are identified. For reviewers, you can see previous series in 'request_v3' branch of my remote repository. I hope this will help your work. ---------------------------------------------------------------- The following changes since commit 38137a064199a02630f136ce7bb1580d5d3fedeb: ALSA: lx_core: Translate comments from french to english (2014-04-18 09:57:03 +0200) are available in the git repository at: https://github.com/takaswie/sound.git request_v4 for you to fetch changes up to e851b7740f218d5d5a8e957851b383bde3e28654: firewire/bebob: Add a workaround for M-Audio special Firewire series (2014-04-25 22:29:32 +0900) ---------------------------------------------------------------- Takashi Sakamoto (49): firewire-lib: Rename functions, structure, member for AMDTP firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse for both streams firewire-lib: Add support for AMDTP in-stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: Give syt value as parameter to handle_out_packet() firewire-lib: Add support for duplex streams synchronization in blocking mode firewire-lib: Add support for channel mapping dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE firewire-lib/dice/speakers: Add common PCM constraints for AMDTP streams firewire-lib: Restrict calling flush_context_completion() when context exists firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add support for deferred transaction firewire-lib: Add some AV/C general commands fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks/firewire-lib: Add a quirk for empty packet with TAG0 fireworks/firewire-lib: Add a quirk for the meaning of dbc fireworks/firewire-lib: Add a quirk for wrong dbs in tx packets fireworks/firewire-lib: Add a quirk for fixed interval of reported dbc fireworks/firewire-lib: Add a quirk to reset data block counter at bus reset fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface fireworks/firewire-lib: Add a quirk of data blocks for MIDI in out-stream fireworks: Add PCM interface fireworks: Add hwdep interface fireworks: Add command/response functionality into hwdep interface bebob: Add skelton for BeBoB based devices bebob: Add commands and connections/streams management bebob/firewire-lib: Add a quirk for discontinuity at bus reset bebob: Add proc interface for debugging purpose bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface bebob: Prepare for device specific operations bebob: Add support for Terratec PHASE, EWS series and Aureon bebob: Add support for Yamaha GO series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for M-Audio usual Firewire series bebob: Add support for M-Audio special Firewire series bebob/firewire-lib: Add a quirk of wrong dbc in empty packet for M-Audio special Firewire series bebob: Add a quirk of data blocks for MIDI messages for some M-Audio devices bebob: Send a cue to load firmware for M-Audio Firewire series firewire/bebob: Add a workaround for M-Audio special Firewire series drivers/firewire/core.h | 1 - include/linux/firewire.h | 3 + include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 23 +- sound/firewire/Kconfig | 59 ++ sound/firewire/Makefile | 2 + sound/firewire/amdtp.c | 797 +++++++++++------ sound/firewire/amdtp.h | 200 ++++- sound/firewire/bebob/Makefile | 4 + sound/firewire/bebob/bebob.c | 472 ++++++++++ sound/firewire/bebob/bebob.h | 257 ++++++ sound/firewire/bebob/bebob_command.c | 282 ++++++ sound/firewire/bebob/bebob_focusrite.c | 279 ++++++ sound/firewire/bebob/bebob_hwdep.c | 199 +++++ sound/firewire/bebob/bebob_maudio.c | 792 +++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 168 ++++ sound/firewire/bebob/bebob_pcm.c | 378 ++++++++ sound/firewire/bebob/bebob_proc.c | 196 +++++ sound/firewire/bebob/bebob_stream.c | 1020 ++++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 68 ++ sound/firewire/bebob/bebob_yamaha.c | 50 ++ sound/firewire/cmp.c | 203 +++-- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 88 +- sound/firewire/fcp.c | 189 +++- sound/firewire/fcp.h | 21 + sound/firewire/fireworks/Makefile | 4 + sound/firewire/fireworks/fireworks.c | 357 ++++++++ sound/firewire/fireworks/fireworks.h | 232 +++++ sound/firewire/fireworks/fireworks_command.c | 372 ++++++++ sound/firewire/fireworks/fireworks_hwdep.c | 298 +++++++ sound/firewire/fireworks/fireworks_midi.c | 168 ++++ sound/firewire/fireworks/fireworks_pcm.c | 403 +++++++++ sound/firewire/fireworks/fireworks_proc.c | 232 +++++ sound/firewire/fireworks/fireworks_stream.c | 372 ++++++++ sound/firewire/fireworks/fireworks_transaction.c | 326 +++++++ sound/firewire/speakers.c | 100 +-- 37 files changed, 8173 insertions(+), 460 deletions(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c create mode 100644 sound/firewire/fireworks/fireworks_transaction.c -- 1.8.3.2

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Re: [alsa-devel] Donation of ARM CubieTruck to anyone interested in developing
by Alexey Eromenko 30 May '14

30 May '14

> I might be interested in helping. > > 1) What is your expected timeline to get this working? I would prefer for it to be ready for Debian 8.0 (not a requirement) > 2) Is there enough support in mainline to at least boot the board to a linux console? Mainline has some support for it, yeah, (but not for sound) Linux mainlining effort http://linux-sunxi.org/Linux_mainlining_effort > 3) If so what kernel? The system officially supports Linux-Android kernel 3.4, which has all the drivers built-in and open-sourced (but sound driver is crappy there) Lubuntu Desktop/Server SD Card Installation for Cubietruck: http://docs.cubieboard.org/tutorials/ct1/installation/install_lubuntu_deskt… -- -Alexey Eromenko "Technologov"

1 0

[alsa-devel] Donation of ARM CubieTruck to anyone interested in developing
by Alexey Eromenko 30 May '14

30 May '14

Hello, My name is Alexey, and I'm interested in having ARM CubieTruck hardware well supported by desktop (KDE) use. CubieTruck is an ARM mini-PC aka "Liliputer", based on AllWinner A20 SoC + 2 GB RAM + 8 GB flash + VGA (!) port + SATA + Ethernet + WiFi, making it good enough for a real Debian desktop. CubieTruck http://cubietruck.com/collections/frontpage/products/cubietruck-cubieboard3… My goal: It should have a working 2D, HD Video and 3D graphics as well as audio and networking. Some open-source drivers exist: *2D - xf86-video-fbturbo - link: https://github.com/ssvb/xf86-video-fbturbo *Hardware Video Decoder - "CedarX" link: http://linux-sunxi.org/Reverse_Engineering/Cedar_Status (should play H.264 and WebM/VP8 Full HD 1080p video) *3D - open-source Lima or closed-source Mali drivers. WiFi + Bluetooth - status unknown. (WiFi firmware is probably closed source) If you can - please help stabilize those experimental drivers, especially sound driver is semi-broken. I can donate one ARM CubieTruck sample to interested people. You'll need a proof of past patches to any open-source project or community involvement. -- -Alexey Eromenko "Technologov". Open-source community member. Israel.

1 0

Re: [alsa-devel] [PATCH] ALSA: firewire: Fix dependency on PCM and rawmidi
by Takashi Iwai 30 May '14

30 May '14

At Fri, 30 May 2014 21:15:58 +0900, Takashi Sakamoto wrote: > > (May 30 2014 17:22), Takashi Iwai wrote: > > Now snd-firewire-lib supports rawmidi in addition to PCM, thus we need > > to give a proper dependency. For fixing and simplification, move the > > selections of SND_PCM and SND_RAWMIDI into SND_FIREWIRE_LIB section. > > Then each driver doesn't have to select them but only > > SND_FIREWIRE_LIB. > > > > Reported-by: Jim Davis <jim.epost(a)gmail.com> > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > Reviewed-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> > Tested-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> > > Thanks for this patch. I've received this bug from kbuild test backend > but I'm not so good at Kbuild system itself and have been investigating > how the Kconfig should be. > > If possible, the dependency of firewire-lib should be switchable because > some of firewire devices don't support MIDI interfaces. For example, > iSight doesn't. So there may be needs not to build MIDI modules. > > But for this idea, MIDI functionalities of each drivers also need to be > disable. As a result, this idea makes the codes complicated a bit. So I > agree with this patch. In that case, we'd need either splitting to yet another helper module only for rawmidi (e.g. snd-firewire-rawmidi), or introduce a Kconfig and select it from the card driver, then build conditionally into snd-firewire-libs. OTOH, the current form has a clear merit of simplicity, indeed. So, I applied the patch now. thanks, Takashi > > > Thanks > > Takashi Sakamoto > o-takashi(a)sakamocchi.jp > > > --- > > sound/firewire/Kconfig | 12 ++---------- > > 1 file changed, 2 insertions(+), 10 deletions(-) > > > > diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig > > index 9f363fac7688..775ef2efc296 100644 > > --- a/sound/firewire/Kconfig > > +++ b/sound/firewire/Kconfig > > @@ -9,12 +9,12 @@ if SND_FIREWIRE && FIREWIRE > > > > config SND_FIREWIRE_LIB > > tristate > > - depends on SND_PCM > > + select SND_PCM > > + select SND_RAWMIDI > > > > config SND_DICE > > tristate "DICE-based DACs (EXPERIMENTAL)" > > select SND_HWDEP > > - select SND_PCM > > select SND_FIREWIRE_LIB > > help > > Say Y here to include support for many DACs based on the DICE > > @@ -28,7 +28,6 @@ config SND_DICE > > > > config SND_FIREWIRE_SPEAKERS > > tristate "FireWire speakers" > > - select SND_PCM > > select SND_FIREWIRE_LIB > > help > > Say Y here to include support for the Griffin FireWave Surround > > @@ -39,7 +38,6 @@ config SND_FIREWIRE_SPEAKERS > > > > config SND_ISIGHT > > tristate "Apple iSight microphone" > > - select SND_PCM > > select SND_FIREWIRE_LIB > > help > > Say Y here to include support for the front and rear microphones > > @@ -50,8 +48,6 @@ config SND_ISIGHT > > > > config SND_SCS1X > > tristate "Stanton Control System 1 MIDI" > > - select SND_PCM > > - select SND_RAWMIDI > > select SND_FIREWIRE_LIB > > help > > Say Y here to include support for the MIDI ports of the Stanton > > @@ -64,8 +60,6 @@ config SND_SCS1X > > config SND_FIREWORKS > > tristate "Echo Fireworks board module support" > > select SND_FIREWIRE_LIB > > - select SND_RAWMIDI > > - select SND_PCM > > select SND_HWDEP > > help > > Say Y here to include support for FireWire devices based > > @@ -82,8 +76,6 @@ config SND_FIREWORKS > > config SND_BEBOB > > tristate "BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware" > > select SND_FIREWIRE_LIB > > - select SND_RAWMIDI > > - select SND_PCM > > select SND_HWDEP > > help > > Say Y here to include support for FireWire devices based >

1 0

[alsa-devel] [PATCH] ALSA: hda/analog - Fix silent output on ASUS A8JN
by Takashi Iwai 30 May '14

30 May '14

ASUS A8JN with AD1986A codec seems following the normal EAPD in the normal order (0 = off, 1 = on) unlike other machines with AD1986A. Apply the workaround used for Toshiba laptop that showed the same problem. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=75041 Cc: <stable(a)vger.kernel.org> [3.11+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 40ba06eb44af..06275f8807a8 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -332,6 +332,7 @@ static const struct hda_fixup ad1986a_fixups[] = { static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC), + SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8JN", AD1986A_FIXUP_EAPD), SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK), SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK), SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_FIXUP_3STACK), -- 1.9.3

1 0

[alsa-devel] [PATCH] sound: remove checks for CONFIG_SND_DEBUG_ROM
by Paul Bolle 30 May '14

30 May '14

Checks for CONFIG_SND_DEBUG_ROM were added in v2.5.5 but a Kconfig symbol SND_DEBUG_ROM was never added. These checks have always evaluated to false. Remove them and the printk()s they hide. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Untested. Perhaps one is expected to define this debugging macro by hand, somehow. In that case it would be nice to drop the CONFIG_ prefix. sound/isa/gus/interwave.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 5abbbe477d16..ad55e5cb8e94 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -442,17 +442,11 @@ static void snd_interwave_detect_memory(struct snd_gus_card *gus) for (bank_pos = 0; bank_pos < 16L * 1024L * 1024L; bank_pos += 4L * 1024L * 1024L) { for (i = 0; i < 8; ++i) iwave[i] = snd_gf1_peek(gus, bank_pos + i); -#ifdef CONFIG_SND_DEBUG_ROM - printk(KERN_DEBUG "ROM at 0x%06x = %8phC\n", bank_pos, iwave); -#endif if (strncmp(iwave, "INTRWAVE", 8)) continue; /* first check */ csum = 0; for (i = 0; i < sizeof(struct rom_hdr); i++) csum += snd_gf1_peek(gus, bank_pos + i); -#ifdef CONFIG_SND_DEBUG_ROM - printk(KERN_DEBUG "ROM checksum = 0x%x (computed)\n", csum); -#endif if (csum != 0) continue; /* not valid rom */ gus->gf1.rom_banks++; -- 1.9.3

2 1

[alsa-devel] [PATCH] sound: remove checks for CONFIG_BCM_CS4297A_CSWARM
by Paul Bolle 30 May '14

30 May '14

Checks for CONFIG_BCM_CS4297A_CSWARM were added in v2.6.11. The related Kconfig symbol was never added so these checks always evaluated to true. Remove them. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Untested. sound/oss/swarm_cs4297a.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index f851fd0e199c..a33e8ce8085b 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -2625,15 +2625,12 @@ static int __init cs4297a_init(void) u32 pwr, id; mm_segment_t fs; int rval; -#ifndef CONFIG_BCM_CS4297A_CSWARM u64 cfg; int mdio_val; -#endif CS_DBGOUT(CS_INIT | CS_FUNCTION, 2, printk(KERN_INFO "cs4297a: cs4297a_init_module()+ \n")); -#ifndef CONFIG_BCM_CS4297A_CSWARM mdio_val = __raw_readq(KSEG1 + A_MAC_REGISTER(2, R_MAC_MDIO)) & (M_MAC_MDIO_DIR|M_MAC_MDIO_OUT); @@ -2659,7 +2656,6 @@ static int __init cs4297a_init(void) __raw_writeq(mdio_val | M_MAC_GENC, KSEG1+A_MAC_REGISTER(2, R_MAC_MDIO)); /* Give the codec some time to finish resetting (start the bit clock) */ udelay(100); -#endif if (!(s = kzalloc(sizeof(struct cs4297a_state), GFP_KERNEL))) { CS_DBGOUT(CS_ERROR, 1, printk(KERN_ERR -- 1.9.3

2 1

[alsa-devel] [PATCH] ALSA: firewire: Fix dependency on PCM and rawmidi
by Takashi Iwai 30 May '14

30 May '14

Now snd-firewire-lib supports rawmidi in addition to PCM, thus we need to give a proper dependency. For fixing and simplification, move the selections of SND_PCM and SND_RAWMIDI into SND_FIREWIRE_LIB section. Then each driver doesn't have to select them but only SND_FIREWIRE_LIB. Reported-by: Jim Davis <jim.epost(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/firewire/Kconfig | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 9f363fac7688..775ef2efc296 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -9,12 +9,12 @@ if SND_FIREWIRE && FIREWIRE config SND_FIREWIRE_LIB tristate - depends on SND_PCM + select SND_PCM + select SND_RAWMIDI config SND_DICE tristate "DICE-based DACs (EXPERIMENTAL)" select SND_HWDEP - select SND_PCM select SND_FIREWIRE_LIB help Say Y here to include support for many DACs based on the DICE @@ -28,7 +28,6 @@ config SND_DICE config SND_FIREWIRE_SPEAKERS tristate "FireWire speakers" - select SND_PCM select SND_FIREWIRE_LIB help Say Y here to include support for the Griffin FireWave Surround @@ -39,7 +38,6 @@ config SND_FIREWIRE_SPEAKERS config SND_ISIGHT tristate "Apple iSight microphone" - select SND_PCM select SND_FIREWIRE_LIB help Say Y here to include support for the front and rear microphones @@ -50,8 +48,6 @@ config SND_ISIGHT config SND_SCS1X tristate "Stanton Control System 1 MIDI" - select SND_PCM - select SND_RAWMIDI select SND_FIREWIRE_LIB help Say Y here to include support for the MIDI ports of the Stanton @@ -64,8 +60,6 @@ config SND_SCS1X config SND_FIREWORKS tristate "Echo Fireworks board module support" select SND_FIREWIRE_LIB - select SND_RAWMIDI - select SND_PCM select SND_HWDEP help Say Y here to include support for FireWire devices based @@ -82,8 +76,6 @@ config SND_FIREWORKS config SND_BEBOB tristate "BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware" select SND_FIREWIRE_LIB - select SND_RAWMIDI - select SND_PCM select SND_HWDEP help Say Y here to include support for FireWire devices based -- 1.9.3

1 0

[alsa-devel] Testing rewindability of a pcm
by Alexander E. Patrakov 30 May '14

30 May '14

Hello. [David: if I misrepresent the result of our discussion, please correct me.] On the pulseaudio-discuss list, there was a thread started by me: http://lists.freedesktop.org/archives/pulseaudio-discuss/2014-April/020449.… The initial patch attempted to work around known brokenness of rewind handling in certain ALSA plugins by keeping a whitelist of ALSA plugin types known to implement full or almost-full rewinds absolutely correctly, and falling back to the low-latency no-rewinds mode on others (such as extplug, which only pretends to support rewinds). While David accepted that it would be a good idea to use a not-so-high latency (he proposed ~150-200 ms) if rewinds are not available, keeping a plugin whitelist inside PulseAudio code was shot down as a workaround for an ALSA bug. The alternative proposal is to fix ALSA so that it correctly reports rewindability, and use the reported data, which looks like a good idea. However, there remain some disagreements on the details how ALSA needs to be fixed, that's why this e-mail. David suggests that PulseAudio should use the snd_pcm_rewindable() function when all plugins are fixed to either implement rewinds correctly or to return 0 if they don't actually support rewinds. And for ioplug and extplug, supporting rewinds is not really an option, because the backend library (which can be an AC3 or DTS encoder) does not necessarily have a rewindable API. <offtopic><troll>And we don't even have a rewindable resampler implementation!</troll></offtopic> While fixing snd_pcm_rewindable() is indeed a valid task that I take for some plugins, I have a question. """ Should PulseAudio indeed use a complex dance with snd_pcm_rewindable() to get the essentially-static bit of information whether a given PCM supports full rewinds? Isn't there a better solution? """ The problem is that the information should be used to set the maximum allowed latency of the sink: a PCM that supports full rewinds can be set to 2 seconds of latency, while for a PCM that doesn't support rewinds, 150-200 ms is more appropriate. In other words, ideally, one bit of information about the future full-rewind support should be available at the server startup without any complex dance with test audio streams. Ideally, even before setting hw_params, but "just after" also seems to be acceptable. And at this stage (before or just after setting hw_params), snd_pcm_rewindable() is useless, because it returns 0 even for the hw plugin which does support full rewinds (rightfully, because the card initially is just about to underrun). Therefore, if the solution is indeed to use snd_pcm_rewindable(), then it is needed to write some test data to the pcm, and only then test for rewindability. Alternatives are below, please help choosing: 1) Use a test stream and call snd_pcm_rewindable() when there is a non-zero amount of data in the buffer: I don't like it. Too complex dance, with side effects, to get one bit of essentially-static information that the plugin knows for sure even before setting hw_params. 2) Use snd_pcm_forwardable() with an empty buffer just after setting hw_params: works (once the implementation in ioplug/extplug is fixed to return 0), doesn't require API additions, but David doesn't like it. He thinks that the sets of PCMs that support forwarding and rewinding may be different (here I disagree, but can't provide arguments). Additionally, forwarding the application pointer without writing anything and rewinding over that beforehand looks like undefined behaviour (which looks like a valid argument, but I am not 100% sure, and I am not actually forwarding, I am just asking how much I can forward). 3) Add a new function to ALSA that gets this bit of static information for a given pcm handle, use it. Drawbacks: a new function is needed, and it's a bad idea to send a patch that adds a new public ALSA API function without discussing here first (that's why the e-mail). 4) Write your own alternative here if you have one. -- Alexander E. Patrakov

4 25

[alsa-devel] [PATCH] ASoC: wm8804: Allow control of master clock divider in PLL generation
by Charles Keepax 29 May '14

29 May '14

From: Daniel Matuschek <info(a)crazy-audio.com> WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using an additional mclk_div divider, it is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. The behavior has to be controlled by the sound card driver, because some sample frequency share the same setting. e.g. 192kHz and 96kHz use 24.576MHz master clock. The only difference is the MCLK divider. Signed-off-by: Daniel Matuschek <daniel(a)matuschek.net> Tested-by: Florian Meier <florian.meier(a)koalo.de> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Hi, Looks like the version of this patch that was sent in didn't apply on the ASoC tree which is probably why it seems to have been missed. Not sure whether this was an email client fail or the patch was generated against the wrong tree but as it is a useful patch I have taken the liberty of fixing it up (code is exactly the same) and resending. Thanks, Charles sound/soc/codecs/wm8804.c | 17 ++++++++++++++--- sound/soc/codecs/wm8804.h | 4 ++++ 2 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index bbcad9f..d96e596 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -63,6 +63,7 @@ struct wm8804_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; + int mclk_div; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -318,7 +319,7 @@ static struct { #define FIXED_PLL_SIZE ((1ULL << 22) * 10) static int pll_factors(struct pll_div *pll_div, unsigned int target, - unsigned int source) + unsigned int source, unsigned int mclk_div) { u64 Kpart; unsigned long int K, Ndiv, Nmod, tmp; @@ -330,7 +331,8 @@ static int pll_factors(struct pll_div *pll_div, unsigned int target, */ for (i = 0; i < ARRAY_SIZE(post_table); i++) { tmp = target * post_table[i].div; - if (tmp >= 90000000 && tmp <= 100000000) { + if ((tmp >= 90000000 && tmp <= 100000000) && + (mclk_div == post_table[i].mclkdiv)) { pll_div->freqmode = post_table[i].freqmode; pll_div->mclkdiv = post_table[i].mclkdiv; target *= post_table[i].div; @@ -387,8 +389,12 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, } else { int ret; struct pll_div pll_div; + struct wm8804_priv *wm8804; - ret = pll_factors(&pll_div, freq_out, freq_in); + wm8804 = snd_soc_codec_get_drvdata(codec); + + ret = pll_factors(&pll_div, freq_out, freq_in, + wm8804->mclk_div); if (ret) return ret; @@ -452,6 +458,7 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { struct snd_soc_codec *codec; + struct wm8804_priv *wm8804; codec = dai->codec; switch (div_id) { @@ -459,6 +466,10 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, snd_soc_update_bits(codec, WM8804_PLL5, 0x30, (div & 0x3) << 4); break; + case WM8804_MCLK_DIV: + wm8804 = snd_soc_codec_get_drvdata(codec); + wm8804->mclk_div = div; + break; default: dev_err(dai->dev, "Unknown clock divider: %d\n", div_id); return -EINVAL; diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index 8ec14f5..e72d4f4 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -57,5 +57,9 @@ #define WM8804_CLKOUT_SRC_OSCCLK 4 #define WM8804_CLKOUT_DIV 1 +#define WM8804_MCLK_DIV 2 + +#define WM8804_MCLKDIV_256FS 0 +#define WM8804_MCLKDIV_128FS 1 #endif /* _WM8804_H */ -- 1.7.2.5

2 1

[alsa-devel] [PATCH 1/3] Revert "ALSA: hda - drop def association and sequence from pinconf comparing"
by Hui Wang 29 May '14

29 May '14

This reverts commit c687200b9d4ef60042a50f7d942cfef120cc7bf1. Dropping the def association and sequence from pinconf comparing is a bit risky, It will introduce a greater risk of catching unwanted machines. And in addition, so far no BIOS experts give us an explicit answer whether it makes senses to compare these two fields or not. For safety reason, we revert this commit. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- This is the first attempt to move existing machines from old quirk table to hda_pin_quirk table. During the working, I met serveral cases that the same codec has different default pin conf value just because of default association difference, Please see the following two patches (0x10ec0255 and 0x10ec0668). sound/pci/hda/hda_auto_parser.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 3cf9137..b684c6e 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -844,8 +844,7 @@ static bool pin_config_match(struct hda_codec *codec, { for (; pins->nid; pins++) { u32 def_conf = snd_hda_codec_get_pincfg(codec, pins->nid); - u32 mask = 0xffffff00; - if ((pins->val & mask) != (def_conf & mask)) + if (pins->val != def_conf) return false; } return true; -- 1.8.1.2

2 3

[alsa-devel] [patch] ALSA: bebob: sizeof() vs ARRAY_SIZE() typo
by Dan Carpenter 29 May '14

29 May '14

ARRAY_SIZE() was intended here instead of sizeof(). The "bridgeco_freq_table" array holds integers so the original condition is never true. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 514c7c9..bc4f827 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -749,7 +749,7 @@ parse_stream_formation(u8 *buf, unsigned int len, if (buf[2] == bridgeco_freq_table[i]) break; } - if (i == sizeof(bridgeco_freq_table)) + if (i == ARRAY_SIZE(bridgeco_freq_table)) return -ENOSYS; /* Avoid double count by different entries for the same rate. */

3 2

[alsa-devel] [patch 2/2] ALSA: fireworks: small leak on error path
by Dan Carpenter 29 May '14

29 May '14

There was a typo here so we return directly instead of freeing "hwinfo". Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 8e856cc..996fdc4 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -157,7 +157,7 @@ get_hardware_info(struct snd_efw *efw) /* Hardware metering. */ if (hwinfo->phys_in_grp_count > HWINFO_MAX_CAPS_GROUPS || hwinfo->phys_out_grp_count > HWINFO_MAX_CAPS_GROUPS) { - return -EIO; + err = -EIO; goto end; } efw->phys_in = hwinfo->phys_in;

3 2

[alsa-devel] [patch 1/2] ALSA: fireworks: remove some stray checks
by Dan Carpenter 29 May '14

29 May '14

We checked "err" earlier. These things seem to be left over code. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index a354d26..8e856cc 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -89,8 +89,6 @@ get_hardware_info(struct snd_efw *efw) snprintf(version, sizeof(version), "%u.%u", (hwinfo->arm_version >> 24) & 0xff, (hwinfo->arm_version >> 16) & 0xff); - if (err < 0) - goto end; efw->firmware_version = hwinfo->arm_version; strcpy(efw->card->driver, "Fireworks"); @@ -101,8 +99,6 @@ get_hardware_info(struct snd_efw *efw) hwinfo->vendor_name, hwinfo->model_name, version, hwinfo->guid_hi, hwinfo->guid_lo, dev_name(&efw->unit->device), 100 << fw_dev->max_speed); - if (err < 0) - goto end; if (hwinfo->flags & BIT(FLAG_RESP_ADDR_CHANGABLE)) efw->resp_addr_changable = true;

3 2

[alsa-devel] [PATCH 1/2] ASoC: export dapm_kcontrol_set/get_value functions
by Vinod Koul 29 May '14

29 May '14

The DSPs like Intel ones use the DPCM to represent the DSP toplogy. So when DAPM triggers the widget, the DSP needs to know the kcontrol values and pass on to DSP firmware, so export these for driver use Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/soc-dapm.h | 3 +++ sound/soc/soc-dapm.c | 6 ++++-- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 75020f5..abda467 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -368,6 +368,9 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_clock_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, + unsigned int value); +unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol); /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 98c1dc6..d831e73 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -327,14 +327,15 @@ static struct list_head *dapm_kcontrol_get_path_list( list_for_each_entry(path, dapm_kcontrol_get_path_list(kcontrol), \ list_kcontrol) -static unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) +unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); return data->value; } +EXPORT_SYMBOL_GPL(dapm_kcontrol_get_value); -static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, +bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, unsigned int value) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); @@ -349,6 +350,7 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, return true; } +EXPORT_SYMBOL_GPL(dapm_kcontrol_set_value); /** * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol -- 1.7.0.4

1 0

[alsa-devel] [PATCH 0/2] Use resource_size instead of computation
by Benoit Taine 28 May '14

28 May '14

These patches enhance kernel style usage, and make use of the resource_size() function instead of explicit computation. The semantic patch at scripts/coccinelle/api/resource_size.cocci was used to detect and edit this situation.

2 2

[alsa-devel] [RFC] AVB - network-based soundcards in ALSA
by Henrik Austad 28 May '14

28 May '14

Hi all! This is an RFC for a new class of soundcards. I am not very familiar with how ALSA is tied together underneath the hood, so what you see here, is based on my naive understanding of ALSA. I wear asbestos underwear on a regular basis, so I prefer honesty over sugarcoating :) I use "I" and "we" interchangeably. By 'we' I mean a small R&D group at Cisco Norway, by "I", I mean.. well, me. So, we plan for AVB, I do the kernel side work. We plan to upstream this, given that the community accepts it. Also, I've used my private address as that is set up to track kernel-related lists, but added my Cisco-address so please keep that on the CC if you reply. We have recently begun working on Audio Video Bridging (AVB, [1]) and is looking into how this can be added to the Linux Kernel via ALSA and video4linux. But first; for those of you who are not familiar with AVB: In short, AVB is just a set of open standards governing network and timing configuration so that you can stream audio and video reliably and with low latency. Note that this is not the kind of streaming services currently associated with streaming (a few companies distributing movies and TV-shows comes to mind; one rhyming with lightsticks). It is the kind of streaming you use when connecting a pair of speakers to your computer - via ethernet. Or a webcam via the wireless network. (I'm aware of the security implications here, but bear with me). For the eager reader, AVB is being promoted by AVnu Alliance [2], they have a lot of information available. I also added a link to a very short intro to AVB that Hans held a few weeks back (focus on the network though) in [3]. Then the IEEE 802.1 working group [4] has a few standards, but these are probably not that relevant no this list, at least not right now. For AVB to work, you need support in the networking infrastructure. This is not prevalent but it is coming. There are a few manufacturers that provide AVB ready equipment and some networking gear. What you need of standards for AVB: * gPTP support (IEEE 802.1AS), this is an IEEE 1588 (PTP) profile for AVB. This is needed for accurate timestamping of samples, and all nodes in an AVB domain must agree to the _same_ time (not that the _correct_ time is not that important in this setting). .1AS should give you a <1us error between the clocks for the systems involved. * Stream Reservation (IEEE 802.1Qat, or 802.1Q:2011 Sec. #35) to make sure we have guaranteed bandwidth. This will avoid dropped etherframes due to congested network. It also caps the amount you can reserve to 75% of total BW, making sure AVB can coexist with normal traffic. * Traffic Shaping and adminssion control (IEEE 802.1Qav, or 802.1Q:2011 Sec. #34) to improve utilization but also avoid/minimize jitter due to queues inside switches/routers/bridges. * IEEE 802.1BA, default configuration for AVB devices and what the network looks like. * IEEE 1722 (and 1733 for layer-3) Layer 2 Transport for audio/video. The packing is similar to what is done in Firewire. You have 8kHz frame intervals for class A, 4kHz for class B. This gives relatively few samples pr. frame. Currently we only look at Layer 2 as small peripherals (microphones, speakers) will only have to implement L2 instead of the entire IP-stack. * IEEE 1722.1 Device discovery protocol (AVDECC) defines how Talkers and Listeners find each other and connect. Any talker will regularly announce its presence, and 1722.1 defines how to announce - and how to respond. Of all these standards, the 802.1BA and 1722 are probably the most interesting ones. AVnu also has a 'best practice' [5] document that gives a outline that serves as a nice starting point. Terminology (brief) - Bridge: Node in the network with more than 1 port (think switches) - End-station: Node in the network with 1 port. - Talkers: End-station that produce media (mic, camera) - Listeners: End-station that receives from Talkers - Streams & Channels: A talker creates a stream through the network to a Listener. Each stream is composed of 1..N channels where each sample is interleaved. - An end-station can act as both Talker and Listener. - gPTP domain: set of PTP-capable nodes connected (gPTP will not allow non-timeaware nodes in the domain). - SRP domain: nodes in a network that supports stream reservation. - AVB Domain: intersection of SRP domain and gPTP domain. To put it to easier terms, AVB gives you a way to add 'stuff' to your computer and play music to them via the network. Moving out into ALSA-land and introducing "The plan": * A central driver, an "avb_core" if you like. Once loaded it will create a configfs directory and start looking at etherframes to see if anything of interest comes along. This will be present from the start and is required for all the rest to work. * An "avb_media_driver" to split data going to ALSA and v4l as well as combining streams coming back. The easiest way is probably to combine snd_avb and the corresponding v4l driver into a single driver, but expose it as "snd_avb" to ALSA (and ditto for v4l2). * A userspace tool for tapping into the AVDECC data (for autodiscovery of nodes). Let's call this avdecclib for now (there are a few userspace libraries available on github). * ConfigFS [6] is then used by userspace to spawn an new avb_media_driver for each stream we want to connect to. Tree-structure will look something like this mkdir /config/avb/node0; config/ └── avb └── node0 ├── channels_in ├── channels_out ├── enable └── mac (the number of attributes will have to be adjusted as I figure out what makes sense to have in the configfs item. Writing 1 to enable will then trigger the negotiating phase and wait for the driver to come online. A new ALSA soundcard will then pop into existence, which can then be used as any regular soundcard attached to the computer. So, an attempt to bring this to life using state of the art ASCII skills +----------------------------------------------------+ | | | media application | | | +-------+-----------------+--------------------+-----+ | | | | | | +-------+-----+ +------+------+ +-------+-----+ | | | | | | | alsalib | | v4l2lib | | avdecclib | | | | | | | userspace +-------+--- -+ +------+------+ +-------+-----+ ................ | | | kernelspace | | | +-------+-----+ | | | | | | | alsa core | | | | | | | +-------+-----+ | | | | | +-------+-----------------+------+ | | | +-------+-----+ | snd_avb v4l2_avb | | | | | | ConfigFS | | avb_media_driver | | | | | +-------+-----+ +-------------+------------+-----+ | | | +-------+-----+ +---------------+ +------+------+ | | | | +----------+ +-----------------| avb_config | | time | | avb_core | | | | | +-----+ | | | +-------------+ +-------+-------+ | +------+------+ | | | | | | | | | +-------+-------+ | | | | | | | | | media_clock | | +------+------+ | | | | | | | +---------------+ +----+ net +-----+ | | +-------------+ So, why in the kernel and not completely in userspace? Primarily because we would like to make it as easy as possible to create a Talker or a Listener in an AVB domain. Sure, you would need some kind of tool to manage the ConfigFS interface and set up the detailed configuration, but once that is done, _any_ program on a standard GNU/Linux box can use AVB as if it was a regular soundcard. That is a real benefit, and what makes it really exciting. It is also a bit difficult to associate a physical location to a MAC-address. A userspace tool can be configured to remember this, but this is not information that belongs in the kernel. This needs to be persistent anyway, so setting 00:00:A4.. to be "L&R Speaker in Henrik's Den" doesn't really make sense to compile into the kernel. Then there is the notion of security. If the kernel triggers on every newly discovered device, it is pretty simple to write a metasploit plugin that will bring any AVB enabled Linux box to its knees by just flooding the network with Announce-messages. Also, I don't necessarily want the stream from my computer to my speakers to be accessed by someone (tm) on my network. I'd greatly appreciate feedback and comments, especially with regards to the rough outline and the usage of ConfigFS and ioctls. Stay tuned! Once we have something that doesn't crash and burn in the most horrible sense, I'll submit a few patches for people to look at. If the interest is high, I'll probably create a public repo that I'll update more frequently, but with more of the bleeding-part of the edge. Thanks! 1) http://en.wikipedia.org/wiki/Audio_Video_Bridging 2) http://www.avnu.org/ 3) http://www.slideshare.net/henrikau/avb-v4l2summit 4) http://en.wikipedia.org/wiki/IEEE_802.1 5) http://www.avnu.org/knowledge_center 6) http://events.linuxfoundation.org/sites/events/files/slides/USB%20Gadget%20… -- Henrik Austad

5 11

[alsa-devel] [PATCH 0/2] ASoC: tlv320aic3x: Fix snd_soc_dapm_put_volsw_aic3x callback
by Peter Ujfalusi 28 May '14

28 May '14

Hi, Commit "cf7c1de20c576 ASoC: dapm: Move 'value' field from widget to control" changed the way how the 'value' has been stored for a widget. This resulted issues with codec drivers needing and using custom dapm put callbacks to handle the hw underneath, like the tlv320aic3x driver. Since the driver is not updated following the mentioned commit it is mostly broken when we try to change Left/Righ PGA mixers: amixer sset 'Left PGA Mixer Line1L' off amixer: Invalid command! The reason for this is that the custom callback was not calling the dapm_kcontrol_set_value() to store the value which causes inconsistent state between the HW and dapm. Regards, Peter --- Peter Ujfalusi (2): ASoC: dapm: Export dapm_kcontrol_set_value() so it can be used by drivers ASoC: tlv320aic3x: Fix Left/Rigth PGA mixer widgets/kcontrols include/sound/soc-dapm.h | 3 +++ sound/soc/codecs/tlv320aic3x.c | 14 +++++++------- sound/soc/soc-dapm.c | 3 ++- 3 files changed, 12 insertions(+), 8 deletions(-) -- 1.9.3

3 12

[alsa-devel] [PATCH 1/5] ASoC: Add ADAU1X61 and ADAU1X81 CODECs common code
by Lars-Peter Clausen 28 May '14

28 May '14

The ADAU1X61 and ADAU1X81 are very similar in the digital domain, but are quite different in the analog domain. This patch adds support for the common parts of the ADAU1X61 and ADAU1X81 CODECs. The patch also restores some of the alphabetical order in the Makfile and Kconfig. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/linux/platform_data/adau17x1.h | 23 + sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 6 +- sound/soc/codecs/adau17x1.c | 866 +++++++++++++++++++++++++++++++++ sound/soc/codecs/adau17x1.h | 124 +++++ 5 files changed, 1022 insertions(+), 3 deletions(-) create mode 100644 include/linux/platform_data/adau17x1.h create mode 100644 sound/soc/codecs/adau17x1.c create mode 100644 sound/soc/codecs/adau17x1.h diff --git a/include/linux/platform_data/adau17x1.h b/include/linux/platform_data/adau17x1.h new file mode 100644 index 0000000..f90bd92 --- /dev/null +++ b/include/linux/platform_data/adau17x1.h @@ -0,0 +1,23 @@ +/* + * Driver for ADAU1761/ADAU1461/ADAU1761/ADAU1961/ADAU1781/ADAU1781 codecs + * + * Copyright 2011-2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#ifndef __LINUX_PLATFORM_DATA_ADAU17X1_H__ +#define __LINUX_PLATFORM_DATA_ADAU17X1_H__ + +/** + * enum adau17x1_micbias_voltage - Microphone bias voltage + * @ADAU17X1_MICBIAS_0_90_AVDD: 0.9 * AVDD + * @ADAU17X1_MICBIAS_0_65_AVDD: 0.65 * AVDD + */ +enum adau17x1_micbias_voltage { + ADAU17X1_MICBIAS_0_90_AVDD = 0, + ADAU17X1_MICBIAS_0_65_AVDD = 1, +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3f7bbe6..1f308a7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -214,13 +214,17 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate +config SND_SOC_ADAU1373 + tristate + config SND_SOC_ADAU1701 tristate "Analog Devices ADAU1701 CODEC" depends on I2C select SND_SOC_SIGMADSP -config SND_SOC_ADAU1373 +config SND_SOC_ADAU17X1 tristate + select SND_SOC_SIGMADSP config SND_SOC_ADAU1977 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1ccdaf0..ee04f5a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,8 +7,9 @@ snd-soc-ad193x-spi-objs := ad193x-spi.o snd-soc-ad193x-i2c-objs := ad193x-i2c.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o -snd-soc-adau1701-objs := adau1701.o snd-soc-adau1373-objs := adau1373.o +snd-soc-adau1701-objs := adau1701.o +snd-soc-adau17x1-objs := adau17x1.o snd-soc-adau1977-objs := adau1977.o snd-soc-adau1977-spi-objs := adau1977-spi.o snd-soc-adau1977-i2c-objs := adau1977-i2c.o @@ -162,10 +163,11 @@ obj-$(CONFIG_SND_SOC_AD193X_I2C) += snd-soc-ad193x-i2c.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o +obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o +obj-$(CONFIG_SND_SOC_ADAU17X1) += snd-soc-adau17x1.o obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o -obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c new file mode 100644 index 0000000..2961fae --- /dev/null +++ b/sound/soc/codecs/adau17x1.c @@ -0,0 +1,866 @@ +/* + * Common code for ADAU1X61 and ADAU1X81 codecs + * + * Copyright 2011-2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <linux/gcd.h> +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <linux/regmap.h> + +#include "sigmadsp.h" +#include "adau17x1.h" + +static const char * const adau17x1_capture_mixer_boost_text[] = { + "Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3", +}; + +static SOC_ENUM_SINGLE_DECL(adau17x1_capture_boost_enum, + ADAU17X1_REC_POWER_MGMT, 5, adau17x1_capture_mixer_boost_text); + +static const char * const adau17x1_mic_bias_mode_text[] = { + "Normal operation", "High performance", +}; + +static SOC_ENUM_SINGLE_DECL(adau17x1_mic_bias_mode_enum, + ADAU17X1_MICBIAS, 3, adau17x1_mic_bias_mode_text); + +static const DECLARE_TLV_DB_MINMAX(adau17x1_digital_tlv, -9563, 0); + +static const struct snd_kcontrol_new adau17x1_controls[] = { + SOC_DOUBLE_R_TLV("Digital Capture Volume", + ADAU17X1_LEFT_INPUT_DIGITAL_VOL, + ADAU17X1_RIGHT_INPUT_DIGITAL_VOL, + 0, 0xff, 1, adau17x1_digital_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume", ADAU17X1_DAC_CONTROL1, + ADAU17X1_DAC_CONTROL2, 0, 0xff, 1, adau17x1_digital_tlv), + + SOC_SINGLE("ADC High Pass Filter Switch", ADAU17X1_ADC_CONTROL, + 5, 1, 0), + SOC_SINGLE("Playback De-emphasis Switch", ADAU17X1_DAC_CONTROL0, + 2, 1, 0), + + SOC_ENUM("Capture Boost", adau17x1_capture_boost_enum), + + SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), +}; + +static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct adau *adau = snd_soc_codec_get_drvdata(w->codec); + int ret; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + adau->pll_regs[5] = 1; + } else { + adau->pll_regs[5] = 0; + /* Bypass the PLL when disabled, otherwise registers will become + * inaccessible. */ + regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, + ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL, 0); + } + + /* The PLL register is 6 bytes long and can only be written at once. */ + ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, + adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); + + if (SND_SOC_DAPM_EVENT_ON(event)) { + mdelay(5); + regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, + ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL, + ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL); + } + + return 0; +} + +static const char * const adau17x1_mono_stereo_text[] = { + "Stereo", + "Mono Left Channel (L+R)", + "Mono Right Channel (L+R)", + "Mono (L+R)", +}; + +static SOC_ENUM_SINGLE_DECL(adau17x1_dac_mode_enum, + ADAU17X1_DAC_CONTROL0, 6, adau17x1_mono_stereo_text); + +static const struct snd_kcontrol_new adau17x1_dac_mode_mux = + SOC_DAPM_ENUM("DAC Mono-Stereo-Mode", adau17x1_dac_mode_enum); + +static const struct snd_soc_dapm_widget adau17x1_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY_S("PLL", 3, SND_SOC_NOPM, 0, 0, adau17x1_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY("AIFCLK", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("MICBIAS", ADAU17X1_MICBIAS, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Left Playback Enable", ADAU17X1_PLAY_POWER_MGMT, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Right Playback Enable", ADAU17X1_PLAY_POWER_MGMT, + 1, 0, NULL, 0), + + SND_SOC_DAPM_MUX("Left DAC Mode Mux", SND_SOC_NOPM, 0, 0, + &adau17x1_dac_mode_mux), + SND_SOC_DAPM_MUX("Right DAC Mode Mux", SND_SOC_NOPM, 0, 0, + &adau17x1_dac_mode_mux), + + SND_SOC_DAPM_ADC("Left Decimator", NULL, ADAU17X1_ADC_CONTROL, 0, 0), + SND_SOC_DAPM_ADC("Right Decimator", NULL, ADAU17X1_ADC_CONTROL, 1, 0), + SND_SOC_DAPM_DAC("Left DAC", NULL, ADAU17X1_DAC_CONTROL0, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", NULL, ADAU17X1_DAC_CONTROL0, 1, 0), +}; + +static const struct snd_soc_dapm_route adau17x1_dapm_routes[] = { + { "Left Decimator", NULL, "SYSCLK" }, + { "Right Decimator", NULL, "SYSCLK" }, + { "Left DAC", NULL, "SYSCLK" }, + { "Right DAC", NULL, "SYSCLK" }, + { "Capture", NULL, "SYSCLK" }, + { "Playback", NULL, "SYSCLK" }, + + { "Left DAC", NULL, "Left DAC Mode Mux" }, + { "Right DAC", NULL, "Right DAC Mode Mux" }, + + { "Capture", NULL, "AIFCLK" }, + { "Playback", NULL, "AIFCLK" }, +}; + +static const struct snd_soc_dapm_route adau17x1_dapm_pll_route = { + "SYSCLK", NULL, "PLL", +}; + +/* + * The MUX register for the Capture and Playback MUXs selects either DSP as + * source/destination or one of the TDM slots. The TDM slot is selected via + * snd_soc_dai_set_tdm_slot(), so we only expose whether to go to the DSP or + * directly to the DAI interface with this control. + */ +static int adau17x1_dsp_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct adau *adau = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + struct snd_soc_dapm_update update; + unsigned int stream = e->shift_l; + unsigned int val, change; + int reg; + + if (ucontrol->value.enumerated.item[0] >= e->items) + return -EINVAL; + + switch (ucontrol->value.enumerated.item[0]) { + case 0: + val = 0; + adau->dsp_bypass[stream] = false; + break; + default: + val = (adau->tdm_slot[stream] * 2) + 1; + adau->dsp_bypass[stream] = true; + break; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = ADAU17X1_SERIAL_INPUT_ROUTE; + else + reg = ADAU17X1_SERIAL_OUTPUT_ROUTE; + + change = snd_soc_test_bits(codec, reg, 0xff, val); + if (change) { + update.kcontrol = kcontrol; + update.mask = 0xff; + update.reg = reg; + update.val = val; + + snd_soc_dapm_mux_update_power(&codec->dapm, kcontrol, + ucontrol->value.enumerated.item[0], e, &update); + } + + return change; +} + +static int adau17x1_dsp_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct adau *adau = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int stream = e->shift_l; + unsigned int reg, val; + int ret; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = ADAU17X1_SERIAL_INPUT_ROUTE; + else + reg = ADAU17X1_SERIAL_OUTPUT_ROUTE; + + ret = regmap_read(adau->regmap, reg, &val); + if (ret) + return ret; + + if (val != 0) + val = 1; + ucontrol->value.enumerated.item[0] = val; + + return 0; +} + +#define DECLARE_ADAU17X1_DSP_MUX_CTRL(_name, _label, _stream, _text) \ + const struct snd_kcontrol_new _name = \ + SOC_DAPM_ENUM_EXT(_label, (const struct soc_enum)\ + SOC_ENUM_SINGLE(SND_SOC_NOPM, _stream, \ + ARRAY_SIZE(_text), _text), \ + adau17x1_dsp_mux_enum_get, adau17x1_dsp_mux_enum_put) + +static const char * const adau17x1_dac_mux_text[] = { + "DSP", + "AIFIN", +}; + +static const char * const adau17x1_capture_mux_text[] = { + "DSP", + "Decimator", +}; + +static DECLARE_ADAU17X1_DSP_MUX_CTRL(adau17x1_dac_mux, "DAC Playback Mux", + SNDRV_PCM_STREAM_PLAYBACK, adau17x1_dac_mux_text); + +static DECLARE_ADAU17X1_DSP_MUX_CTRL(adau17x1_capture_mux, "Capture Mux", + SNDRV_PCM_STREAM_CAPTURE, adau17x1_capture_mux_text); + +static const struct snd_soc_dapm_widget adau17x1_dsp_dapm_widgets[] = { + SND_SOC_DAPM_PGA("DSP", ADAU17X1_DSP_RUN, 0, 0, NULL, 0), + SND_SOC_DAPM_SIGGEN("DSP Siggen"), + + SND_SOC_DAPM_MUX("DAC Playback Mux", SND_SOC_NOPM, 0, 0, + &adau17x1_dac_mux), + SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, + &adau17x1_capture_mux), +}; + +static const struct snd_soc_dapm_route adau17x1_dsp_dapm_routes[] = { + { "DAC Playback Mux", "DSP", "DSP" }, + { "DAC Playback Mux", "AIFIN", "Playback" }, + + { "Left DAC Mode Mux", "Stereo", "DAC Playback Mux" }, + { "Left DAC Mode Mux", "Mono (L+R)", "DAC Playback Mux" }, + { "Left DAC Mode Mux", "Mono Left Channel (L+R)", "DAC Playback Mux" }, + { "Right DAC Mode Mux", "Stereo", "DAC Playback Mux" }, + { "Right DAC Mode Mux", "Mono (L+R)", "DAC Playback Mux" }, + { "Right DAC Mode Mux", "Mono Right Channel (L+R)", "DAC Playback Mux" }, + + { "Capture Mux", "DSP", "DSP" }, + { "Capture Mux", "Decimator", "Left Decimator" }, + { "Capture Mux", "Decimator", "Right Decimator" }, + + { "Capture", NULL, "Capture Mux" }, + + { "DSP", NULL, "DSP Siggen" }, + + { "DSP", NULL, "Left Decimator" }, + { "DSP", NULL, "Right Decimator" }, +}; + +static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { + { "Left DAC Mode Mux", "Stereo", "Playback" }, + { "Left DAC Mode Mux", "Mono (L+R)", "Playback" }, + { "Left DAC Mode Mux", "Mono Left Channel (L+R)", "Playback" }, + { "Right DAC Mode Mux", "Stereo", "Playback" }, + { "Right DAC Mode Mux", "Mono (L+R)", "Playback" }, + { "Right DAC Mode Mux", "Mono Right Channel (L+R)", "Playback" }, + { "Capture", NULL, "Left Decimator" }, + { "Capture", NULL, "Right Decimator" }, +}; + +bool adau17x1_has_dsp(struct adau *adau) +{ + switch (adau->type) { + case ADAU1761: + case ADAU1381: + case ADAU1781: + return true; + default: + return false; + } +} +EXPORT_SYMBOL_GPL(adau17x1_has_dsp); + +static int adau17x1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau *adau = snd_soc_codec_get_drvdata(codec); + unsigned int val, div, dsp_div; + unsigned int freq; + + if (adau->clk_src == ADAU17X1_CLK_SRC_PLL) + freq = adau->pll_freq; + else + freq = adau->sysclk; + + if (freq % params_rate(params) != 0) + return -EINVAL; + + switch (freq / params_rate(params)) { + case 1024: /* fs */ + div = 0; + dsp_div = 1; + break; + case 6144: /* fs / 6 */ + div = 1; + dsp_div = 6; + break; + case 4096: /* fs / 4 */ + div = 2; + dsp_div = 5; + break; + case 3072: /* fs / 3 */ + div = 3; + dsp_div = 4; + break; + case 2048: /* fs / 2 */ + div = 4; + dsp_div = 3; + break; + case 1536: /* fs / 1.5 */ + div = 5; + dsp_div = 2; + break; + case 512: /* fs / 0.5 */ + div = 6; + dsp_div = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0, + ADAU17X1_CONVERTER0_CONVSR_MASK, div); + if (adau17x1_has_dsp(adau)) { + regmap_write(adau->regmap, ADAU17X1_SERIAL_SAMPLING_RATE, div); + regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dsp_div); + } + + if (adau->dai_fmt != SND_SOC_DAIFMT_RIGHT_J) + return 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAU17X1_SERIAL_PORT1_DELAY16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAU17X1_SERIAL_PORT1_DELAY8; + break; + case SNDRV_PCM_FORMAT_S32_LE: + val = ADAU17X1_SERIAL_PORT1_DELAY0; + break; + default: + return -EINVAL; + } + + return regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT1, + ADAU17X1_SERIAL_PORT1_DELAY_MASK, val); +} + +static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau *adau = snd_soc_codec_get_drvdata(codec); + unsigned int r, n, m, i, j; + unsigned int div; + int ret; + + if (freq_in < 8000000 || freq_in > 27000000) + return -EINVAL; + + if (!freq_out) { + r = 0; + n = 0; + m = 0; + div = 0; + } else { + if (freq_out % freq_in != 0) { + div = DIV_ROUND_UP(freq_in, 13500000); + freq_in /= div; + r = freq_out / freq_in; + i = freq_out % freq_in; + j = gcd(i, freq_in); + n = i / j; + m = freq_in / j; + div--; + } else { + r = freq_out / freq_in; + n = 0; + m = 0; + div = 0; + } + if (n > 0xffff || m > 0xffff || div > 3 || r > 8 || r < 2) + return -EINVAL; + } + + adau->pll_regs[0] = m >> 8; + adau->pll_regs[1] = m & 0xff; + adau->pll_regs[2] = n >> 8; + adau->pll_regs[3] = n & 0xff; + adau->pll_regs[4] = (r << 3) | (div << 1); + if (m != 0) + adau->pll_regs[4] |= 1; /* Fractional mode */ + + /* The PLL register is 6 bytes long and can only be written at once. */ + ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, + adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); + if (ret) + return ret; + + adau->pll_freq = freq_out; + + return 0; +} + +static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + struct snd_soc_dapm_context *dapm = &dai->codec->dapm; + + switch (clk_id) { + case ADAU17X1_CLK_SRC_MCLK: + case ADAU17X1_CLK_SRC_PLL: + break; + default: + return -EINVAL; + } + + adau->sysclk = freq; + + if (adau->clk_src != clk_id) { + if (clk_id == ADAU17X1_CLK_SRC_PLL) { + snd_soc_dapm_add_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } else { + snd_soc_dapm_del_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } + } + + adau->clk_src = clk_id; + + return 0; +} + +static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl0, ctrl1; + int lrclk_pol; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl0 = ADAU17X1_SERIAL_PORT0_MASTER; + adau->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl0 = 0; + adau->master = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + lrclk_pol = 0; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY1; + break; + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + lrclk_pol = 1; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY0; + break; + case SND_SOC_DAIFMT_DSP_A: + lrclk_pol = 1; + ctrl0 |= ADAU17X1_SERIAL_PORT0_PULSE_MODE; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY1; + break; + case SND_SOC_DAIFMT_DSP_B: + lrclk_pol = 1; + ctrl0 |= ADAU17X1_SERIAL_PORT0_PULSE_MODE; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl0 |= ADAU17X1_SERIAL_PORT0_BCLK_POL; + break; + case SND_SOC_DAIFMT_NB_IF: + lrclk_pol = !lrclk_pol; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl0 |= ADAU17X1_SERIAL_PORT0_BCLK_POL; + lrclk_pol = !lrclk_pol; + break; + default: + return -EINVAL; + } + + if (lrclk_pol) + ctrl0 |= ADAU17X1_SERIAL_PORT0_LRCLK_POL; + + regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT0, ctrl0); + regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT1, ctrl1); + + adau->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + return 0; +} + +static int adau17x1_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ser_ctrl0, ser_ctrl1; + unsigned int conv_ctrl0, conv_ctrl1; + + /* I2S mode */ + if (slots == 0) { + slots = 2; + rx_mask = 3; + tx_mask = 3; + slot_width = 32; + } + + switch (slots) { + case 2: + ser_ctrl0 = ADAU17X1_SERIAL_PORT0_STEREO; + break; + case 4: + ser_ctrl0 = ADAU17X1_SERIAL_PORT0_TDM4; + break; + case 8: + if (adau->type == ADAU1361) + return -EINVAL; + + ser_ctrl0 = ADAU17X1_SERIAL_PORT0_TDM8; + break; + default: + return -EINVAL; + } + + switch (slot_width * slots) { + case 32: + if (adau->type == ADAU1761) + return -EINVAL; + + ser_ctrl1 = ADAU17X1_SERIAL_PORT1_BCLK32; + break; + case 64: + ser_ctrl1 = ADAU17X1_SERIAL_PORT1_BCLK64; + break; + case 48: + ser_ctrl1 = ADAU17X1_SERIAL_PORT1_BCLK48; + break; + case 128: + ser_ctrl1 = ADAU17X1_SERIAL_PORT1_BCLK128; + break; + case 256: + if (adau->type == ADAU1361) + return -EINVAL; + + ser_ctrl1 = ADAU17X1_SERIAL_PORT1_BCLK256; + break; + default: + return -EINVAL; + } + + switch (rx_mask) { + case 0x03: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(1); + adau->tdm_slot[SNDRV_PCM_STREAM_CAPTURE] = 0; + break; + case 0x0c: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(2); + adau->tdm_slot[SNDRV_PCM_STREAM_CAPTURE] = 1; + break; + case 0x30: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(3); + adau->tdm_slot[SNDRV_PCM_STREAM_CAPTURE] = 2; + break; + case 0xc0: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(4); + adau->tdm_slot[SNDRV_PCM_STREAM_CAPTURE] = 3; + break; + default: + return -EINVAL; + } + + switch (tx_mask) { + case 0x03: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(1); + adau->tdm_slot[SNDRV_PCM_STREAM_PLAYBACK] = 0; + break; + case 0x0c: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(2); + adau->tdm_slot[SNDRV_PCM_STREAM_PLAYBACK] = 1; + break; + case 0x30: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(3); + adau->tdm_slot[SNDRV_PCM_STREAM_PLAYBACK] = 2; + break; + case 0xc0: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(4); + adau->tdm_slot[SNDRV_PCM_STREAM_PLAYBACK] = 3; + break; + default: + return -EINVAL; + } + + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0, + ADAU17X1_CONVERTER0_DAC_PAIR_MASK, conv_ctrl0); + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER1, + ADAU17X1_CONVERTER1_ADC_PAIR_MASK, conv_ctrl1); + regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT0, + ADAU17X1_SERIAL_PORT0_TDM_MASK, ser_ctrl0); + regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT1, + ADAU17X1_SERIAL_PORT1_BCLK_MASK, ser_ctrl1); + + if (!adau17x1_has_dsp(adau)) + return 0; + + if (adau->dsp_bypass[SNDRV_PCM_STREAM_PLAYBACK]) { + regmap_write(adau->regmap, ADAU17X1_SERIAL_INPUT_ROUTE, + (adau->tdm_slot[SNDRV_PCM_STREAM_PLAYBACK] * 2) + 1); + } + + if (adau->dsp_bypass[SNDRV_PCM_STREAM_CAPTURE]) { + regmap_write(adau->regmap, ADAU17X1_SERIAL_OUTPUT_ROUTE, + (adau->tdm_slot[SNDRV_PCM_STREAM_CAPTURE] * 2) + 1); + } + + return 0; +} + +const struct snd_soc_dai_ops adau17x1_dai_ops = { + .hw_params = adau17x1_hw_params, + .set_sysclk = adau17x1_set_dai_sysclk, + .set_fmt = adau17x1_set_dai_fmt, + .set_pll = adau17x1_set_dai_pll, + .set_tdm_slot = adau17x1_set_dai_tdm_slot, +}; +EXPORT_SYMBOL_GPL(adau17x1_dai_ops); + +int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, + enum adau17x1_micbias_voltage micbias) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + + switch (micbias) { + case ADAU17X1_MICBIAS_0_90_AVDD: + case ADAU17X1_MICBIAS_0_65_AVDD: + break; + default: + return -EINVAL; + } + + return regmap_write(adau->regmap, ADAU17X1_MICBIAS, micbias << 2); +} +EXPORT_SYMBOL_GPL(adau17x1_set_micbias_voltage); + +bool adau17x1_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADAU17X1_CLOCK_CONTROL: + case ADAU17X1_PLL_CONTROL: + case ADAU17X1_REC_POWER_MGMT: + case ADAU17X1_MICBIAS: + case ADAU17X1_SERIAL_PORT0: + case ADAU17X1_SERIAL_PORT1: + case ADAU17X1_CONVERTER0: + case ADAU17X1_CONVERTER1: + case ADAU17X1_LEFT_INPUT_DIGITAL_VOL: + case ADAU17X1_RIGHT_INPUT_DIGITAL_VOL: + case ADAU17X1_ADC_CONTROL: + case ADAU17X1_PLAY_POWER_MGMT: + case ADAU17X1_DAC_CONTROL0: + case ADAU17X1_DAC_CONTROL1: + case ADAU17X1_DAC_CONTROL2: + case ADAU17X1_SERIAL_PORT_PAD: + case ADAU17X1_CONTROL_PORT_PAD0: + case ADAU17X1_CONTROL_PORT_PAD1: + case ADAU17X1_DSP_SAMPLING_RATE: + case ADAU17X1_SERIAL_INPUT_ROUTE: + case ADAU17X1_SERIAL_OUTPUT_ROUTE: + case ADAU17X1_DSP_ENABLE: + case ADAU17X1_DSP_RUN: + case ADAU17X1_SERIAL_SAMPLING_RATE: + return true; + default: + break; + } + return false; +} +EXPORT_SYMBOL_GPL(adau17x1_readable_register); + +bool adau17x1_volatile_register(struct device *dev, unsigned int reg) +{ + /* SigmaDSP parameter and program memory */ + if (reg < 0x4000) + return true; + + switch (reg) { + /* The PLL register is 6 bytes long */ + case ADAU17X1_PLL_CONTROL: + case ADAU17X1_PLL_CONTROL + 1: + case ADAU17X1_PLL_CONTROL + 2: + case ADAU17X1_PLL_CONTROL + 3: + case ADAU17X1_PLL_CONTROL + 4: + case ADAU17X1_PLL_CONTROL + 5: + return true; + default: + break; + } + + return false; +} +EXPORT_SYMBOL_GPL(adau17x1_volatile_register); + +int adau17x1_load_firmware(struct adau *adau, struct device *dev, + const char *firmware) +{ + int ret; + int dspsr; + + ret = regmap_read(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, &dspsr); + if (ret) + return ret; + + regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 1); + regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, 0xf); + + ret = process_sigma_firmware_regmap(dev, adau->regmap, firmware); + if (ret) { + regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 0); + return ret; + } + regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dspsr); + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_load_firmware); + +int adau17x1_add_widgets(struct snd_soc_codec *codec) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_add_codec_controls(codec, adau17x1_controls, + ARRAY_SIZE(adau17x1_controls)); + if (ret) + return ret; + ret = snd_soc_dapm_new_controls(&codec->dapm, adau17x1_dapm_widgets, + ARRAY_SIZE(adau17x1_dapm_widgets)); + if (ret) + return ret; + + if (adau17x1_has_dsp(adau)) { + ret = snd_soc_dapm_new_controls(&codec->dapm, + adau17x1_dsp_dapm_widgets, + ARRAY_SIZE(adau17x1_dsp_dapm_widgets)); + } + return ret; +} +EXPORT_SYMBOL_GPL(adau17x1_add_widgets); + +int adau17x1_add_routes(struct snd_soc_codec *codec) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_dapm_add_routes(&codec->dapm, adau17x1_dapm_routes, + ARRAY_SIZE(adau17x1_dapm_routes)); + if (ret) + return ret; + + if (adau17x1_has_dsp(adau)) { + ret = snd_soc_dapm_add_routes(&codec->dapm, + adau17x1_dsp_dapm_routes, + ARRAY_SIZE(adau17x1_dsp_dapm_routes)); + } else { + ret = snd_soc_dapm_add_routes(&codec->dapm, + adau17x1_no_dsp_dapm_routes, + ARRAY_SIZE(adau17x1_no_dsp_dapm_routes)); + } + return ret; +} +EXPORT_SYMBOL_GPL(adau17x1_add_routes); + +int adau17x1_suspend(struct snd_soc_codec *codec) +{ + codec->driver->set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_suspend); + +int adau17x1_resume(struct snd_soc_codec *codec) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + + if (adau->switch_mode) + adau->switch_mode(codec->dev); + + codec->driver->set_bias_level(codec, SND_SOC_BIAS_STANDBY); + regcache_sync(adau->regmap); + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_resume); + +int adau17x1_probe(struct device *dev, struct regmap *regmap, + enum adau17x1_type type, void (*switch_mode)(struct device *dev)) +{ + struct adau *adau; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + adau = devm_kzalloc(dev, sizeof(*adau), GFP_KERNEL); + if (!adau) + return -ENOMEM; + + adau->regmap = regmap; + adau->switch_mode = switch_mode; + adau->type = type; + + dev_set_drvdata(dev, adau); + + if (switch_mode) + switch_mode(dev); + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_probe); + +MODULE_DESCRIPTION("ASoC ADAU1X61/ADAU1X81 common code"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h new file mode 100644 index 0000000..3ffabaf --- /dev/null +++ b/sound/soc/codecs/adau17x1.h @@ -0,0 +1,124 @@ +#ifndef __ADAU17X1_H__ +#define __ADAU17X1_H__ + +#include <linux/regmap.h> +#include <linux/platform_data/adau17x1.h> + +enum adau17x1_type { + ADAU1361, + ADAU1761, + ADAU1381, + ADAU1781, +}; + +enum adau17x1_pll { + ADAU17X1_PLL, +}; + +enum adau17x1_pll_src { + ADAU17X1_PLL_SRC_MCLK, +}; + +enum adau17x1_clk_src { + ADAU17X1_CLK_SRC_MCLK, + ADAU17X1_CLK_SRC_PLL, +}; + +struct adau { + unsigned int sysclk; + unsigned int pll_freq; + + enum adau17x1_clk_src clk_src; + enum adau17x1_type type; + void (*switch_mode)(struct device *dev); + + unsigned int dai_fmt; + + uint8_t pll_regs[6]; + + bool master; + + unsigned int tdm_slot[2]; + bool dsp_bypass[2]; + + struct regmap *regmap; +}; + +int adau17x1_add_widgets(struct snd_soc_codec *codec); +int adau17x1_add_routes(struct snd_soc_codec *codec); +int adau17x1_probe(struct device *dev, struct regmap *regmap, + enum adau17x1_type type, void (*switch_mode)(struct device *dev)); +int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, + enum adau17x1_micbias_voltage micbias); +bool adau17x1_readable_register(struct device *dev, unsigned int reg); +bool adau17x1_volatile_register(struct device *dev, unsigned int reg); +int adau17x1_suspend(struct snd_soc_codec *codec); +int adau17x1_resume(struct snd_soc_codec *codec); + +extern const struct snd_soc_dai_ops adau17x1_dai_ops; + +int adau17x1_load_firmware(struct adau *adau, struct device *dev, + const char *firmware); +bool adau17x1_has_dsp(struct adau *adau); + +#define ADAU17X1_CLOCK_CONTROL 0x4000 +#define ADAU17X1_PLL_CONTROL 0x4002 +#define ADAU17X1_REC_POWER_MGMT 0x4009 +#define ADAU17X1_MICBIAS 0x4010 +#define ADAU17X1_SERIAL_PORT0 0x4015 +#define ADAU17X1_SERIAL_PORT1 0x4016 +#define ADAU17X1_CONVERTER0 0x4017 +#define ADAU17X1_CONVERTER1 0x4018 +#define ADAU17X1_LEFT_INPUT_DIGITAL_VOL 0x401a +#define ADAU17X1_RIGHT_INPUT_DIGITAL_VOL 0x401b +#define ADAU17X1_ADC_CONTROL 0x4019 +#define ADAU17X1_PLAY_POWER_MGMT 0x4029 +#define ADAU17X1_DAC_CONTROL0 0x402a +#define ADAU17X1_DAC_CONTROL1 0x402b +#define ADAU17X1_DAC_CONTROL2 0x402c +#define ADAU17X1_SERIAL_PORT_PAD 0x402d +#define ADAU17X1_CONTROL_PORT_PAD0 0x402f +#define ADAU17X1_CONTROL_PORT_PAD1 0x4030 +#define ADAU17X1_DSP_SAMPLING_RATE 0x40eb +#define ADAU17X1_SERIAL_INPUT_ROUTE 0x40f2 +#define ADAU17X1_SERIAL_OUTPUT_ROUTE 0x40f3 +#define ADAU17X1_DSP_ENABLE 0x40f5 +#define ADAU17X1_DSP_RUN 0x40f6 +#define ADAU17X1_SERIAL_SAMPLING_RATE 0x40f8 + +#define ADAU17X1_SERIAL_PORT0_BCLK_POL BIT(4) +#define ADAU17X1_SERIAL_PORT0_LRCLK_POL BIT(3) +#define ADAU17X1_SERIAL_PORT0_MASTER BIT(0) + +#define ADAU17X1_SERIAL_PORT1_DELAY1 0x00 +#define ADAU17X1_SERIAL_PORT1_DELAY0 0x01 +#define ADAU17X1_SERIAL_PORT1_DELAY8 0x02 +#define ADAU17X1_SERIAL_PORT1_DELAY16 0x03 +#define ADAU17X1_SERIAL_PORT1_DELAY_MASK 0x03 + +#define ADAU17X1_CLOCK_CONTROL_INFREQ_MASK 0x6 +#define ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL BIT(3) +#define ADAU17X1_CLOCK_CONTROL_SYSCLK_EN BIT(0) + +#define ADAU17X1_SERIAL_PORT1_BCLK32 (0x0 << 5) +#define ADAU17X1_SERIAL_PORT1_BCLK48 (0x1 << 5) +#define ADAU17X1_SERIAL_PORT1_BCLK64 (0x2 << 5) +#define ADAU17X1_SERIAL_PORT1_BCLK128 (0x3 << 5) +#define ADAU17X1_SERIAL_PORT1_BCLK256 (0x4 << 5) +#define ADAU17X1_SERIAL_PORT1_BCLK_MASK (0x7 << 5) + +#define ADAU17X1_SERIAL_PORT0_STEREO (0x0 << 1) +#define ADAU17X1_SERIAL_PORT0_TDM4 (0x1 << 1) +#define ADAU17X1_SERIAL_PORT0_TDM8 (0x2 << 1) +#define ADAU17X1_SERIAL_PORT0_TDM_MASK (0x3 << 1) +#define ADAU17X1_SERIAL_PORT0_PULSE_MODE BIT(5) + +#define ADAU17X1_CONVERTER0_DAC_PAIR(x) (((x) - 1) << 5) +#define ADAU17X1_CONVERTER0_DAC_PAIR_MASK (0x3 << 5) +#define ADAU17X1_CONVERTER1_ADC_PAIR(x) ((x) - 1) +#define ADAU17X1_CONVERTER1_ADC_PAIR_MASK 0x3 + +#define ADAU17X1_CONVERTER0_CONVSR_MASK 0x7 + + +#endif -- 1.8.0

3 6

[alsa-devel] [PATCH RFC 02/11] sound: ASoC: kirkwood: Remove unused drivers
by Andrew Lunn 28 May '14

28 May '14

Both kirkwood-openrd and kirkwood-t5325 drivers have been replaced with DT based simple-card equivelents. Signed-off-by: Andrew Lunn <andrew(a)lunn.ch> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/kirkwood/Kconfig | 17 ----- sound/soc/kirkwood/Makefile | 4 -- sound/soc/kirkwood/kirkwood-openrd.c | 109 ----------------------------- sound/soc/kirkwood/kirkwood-t5325.c | 129 ----------------------------------- 4 files changed, 259 deletions(-) delete mode 100644 sound/soc/kirkwood/kirkwood-openrd.c delete mode 100644 sound/soc/kirkwood/kirkwood-t5325.c diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 06f4e8aa93ae..1f7c7ee3527a 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -15,20 +15,3 @@ config SND_KIRKWOOD_SOC_ARMADA370_DB Say Y if you want to add support for SoC audio on the Armada 370 Development Board. -config SND_KIRKWOOD_SOC_OPENRD - tristate "SoC Audio support for Kirkwood Openrd Client" - depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST) - depends on I2C - select SND_SOC_CS42L51 - help - Say Y if you want to add support for SoC audio on - Openrd Client. - -config SND_KIRKWOOD_SOC_T5325 - tristate "SoC Audio support for HP t5325" - depends on SND_KIRKWOOD_SOC && (MACH_T5325 || COMPILE_TEST) && I2C - select SND_SOC_ALC5623 - help - Say Y if you want to add support for SoC audio on - the HP t5325 thin client. - diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 7c1d8fe09e6b..c36b03d8006c 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -2,10 +2,6 @@ snd-soc-kirkwood-objs := kirkwood-dma.o kirkwood-i2s.o obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o -snd-soc-openrd-objs := kirkwood-openrd.o -snd-soc-t5325-objs := kirkwood-t5325.o snd-soc-armada-370-db-objs := armada-370-db.o obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c deleted file mode 100644 index 65f2a5b9ec3b..000000000000 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ /dev/null @@ -1,109 +0,0 @@ -/* - * kirkwood-openrd.c - * - * (c) 2010 Arnaud Patard <apatard(a)mandriva.com> - * (c) 2010 Arnaud Patard <arnaud.patard(a)rtp-net.org> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <sound/soc.h> -#include <linux/platform_data/asoc-kirkwood.h> -#include "../codecs/cs42l51.h" - -static int openrd_client_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int freq; - - switch (params_rate(params)) { - default: - case 44100: - freq = 11289600; - break; - case 48000: - freq = 12288000; - break; - case 96000: - freq = 24576000; - break; - } - - return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); - -} - -static struct snd_soc_ops openrd_client_ops = { - .hw_params = openrd_client_hw_params, -}; - - -static struct snd_soc_dai_link openrd_client_dai[] = { -{ - .name = "CS42L51", - .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "i2s", - .platform_name = "mvebu-audio", - .codec_dai_name = "cs42l51-hifi", - .codec_name = "cs42l51-codec.0-004a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, - .ops = &openrd_client_ops, -}, -}; - - -static struct snd_soc_card openrd_client = { - .name = "OpenRD Client", - .owner = THIS_MODULE, - .dai_link = openrd_client_dai, - .num_links = ARRAY_SIZE(openrd_client_dai), -}; - -static int openrd_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &openrd_client; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static int openrd_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - -static struct platform_driver openrd_driver = { - .driver = { - .name = "openrd-client-audio", - .owner = THIS_MODULE, - }, - .probe = openrd_probe, - .remove = openrd_remove, -}; - -module_platform_driver(openrd_driver); - -/* Module information */ -MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); -MODULE_DESCRIPTION("ALSA SoC OpenRD Client"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:openrd-client-audio"); diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c deleted file mode 100644 index d213832b0c72..000000000000 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ /dev/null @@ -1,129 +0,0 @@ -/* - * kirkwood-t5325.c - * - * (c) 2010 Arnaud Patard <arnaud.patard(a)rtp-net.org> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <sound/soc.h> -#include <linux/platform_data/asoc-kirkwood.h> -#include "../codecs/alc5623.h" - -static int t5325_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int freq; - - freq = params_rate(params) * 256; - - return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); - -} - -static struct snd_soc_ops t5325_ops = { - .hw_params = t5325_hw_params, -}; - -static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route t5325_route[] = { - { "Headphone Jack", NULL, "HPL" }, - { "Headphone Jack", NULL, "HPR" }, - - {"Speaker", NULL, "SPKOUT"}, - {"Speaker", NULL, "SPKOUTN"}, - - { "MIC1", NULL, "Mic Jack" }, - { "MIC2", NULL, "Mic Jack" }, -}; - -static int t5325_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin(dapm, "Speaker"); - - return 0; -} - -static struct snd_soc_dai_link t5325_dai[] = { -{ - .name = "ALC5621", - .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "i2s", - .platform_name = "mvebu-audio", - .codec_dai_name = "alc5621-hifi", - .codec_name = "alc562x-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, - .ops = &t5325_ops, - .init = t5325_dai_init, -}, -}; - -static struct snd_soc_card t5325 = { - .name = "t5325", - .owner = THIS_MODULE, - .dai_link = t5325_dai, - .num_links = ARRAY_SIZE(t5325_dai), - - .dapm_widgets = t5325_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(t5325_dapm_widgets), - .dapm_routes = t5325_route, - .num_dapm_routes = ARRAY_SIZE(t5325_route), -}; - -static int t5325_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &t5325; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static int t5325_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - -static struct platform_driver t5325_driver = { - .driver = { - .name = "t5325-audio", - .owner = THIS_MODULE, - }, - .probe = t5325_probe, - .remove = t5325_remove, -}; - -module_platform_driver(t5325_driver); - -MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); -MODULE_DESCRIPTION("ALSA SoC t5325 audio client"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:t5325-audio"); -- 2.0.0.rc2

3 6

[alsa-devel] [PATCH] ASoC: fix snd_soc_jack_add_gpiods stub
by Arnd Bergmann 27 May '14

27 May '14

f025d3b9c64e1 ("ASoC: jack: Add support for GPIO descriptor defined jack pins") added this new interface, but the stub function provided for non-gpiolib builds was defined in the header file as a global function, leading to lots of "multiple definition of `snd_soc_jack_add_gpiods'" warnings. This adds the obvious "static inline" annotation. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index 98dca42..559dc53 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -464,9 +464,9 @@ static inline int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, return 0; } -int snd_soc_jack_add_gpiods(struct device *gpiod_dev, - struct snd_soc_jack *jack, - int count, struct snd_soc_jack_gpio *gpios) +static inline int snd_soc_jack_add_gpiods(struct device *gpiod_dev, + struct snd_soc_jack *jack, + int count, struct snd_soc_jack_gpio *gpios) { return 0; }

2 2

[alsa-devel] [PATCH 00/12] fixes for firewire topic branch
by Takashi Sakamoto 27 May '14

27 May '14

This series of patch is to fix this patchset: [PATCH 00/49 v4] Enhancement for support of Firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-April/075841.html After the patchset was merged into firewire topic branch, kbuild test backend reports some warnings. And I found some bugs and minor issues by myself. If you want to see fixup for each patch in the patchset, please see: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-May/077140.html Takashi Sakamoto (12): firewire-lib: Use ARRAY_SIZE() instead of sizeof() for correct loop limit firewire-lib: Fix sparse warning of incorrect type in assignment fireworks: Fix wrong value as argument for PTR_ERR() fireworks: Use the same prototype for functions as actual declaration fireworks/bebob: Change type of argument for sampling rate fireworks/bebob: Use the same type of variables as function parameters fireworks/bebob: Add suffix for long long integer literal fireworks/bebob: Improve indentation bebob: Use different names for identifiers in the same file bebob: Add static specifier to identifier with file scope bebob: Remove meaningless mutex_unlock() bebob: Improve comments about stream format sound/firewire/amdtp.c | 4 ++-- sound/firewire/bebob/bebob.c | 9 ++++----- sound/firewire/bebob/bebob.h | 6 +++--- sound/firewire/bebob/bebob_focusrite.c | 24 ++++++++++++------------ sound/firewire/bebob/bebob_maudio.c | 4 ++-- sound/firewire/bebob/bebob_midi.c | 4 ++-- sound/firewire/bebob/bebob_stream.c | 7 ++++--- sound/firewire/bebob/bebob_terratec.c | 6 +++--- sound/firewire/cmp.c | 4 +++- sound/firewire/fireworks/fireworks.h | 5 +++-- sound/firewire/fireworks/fireworks_hwdep.c | 2 +- sound/firewire/fireworks/fireworks_midi.c | 4 ++-- sound/firewire/fireworks/fireworks_pcm.c | 4 ++-- sound/firewire/fireworks/fireworks_stream.c | 2 +- sound/firewire/fireworks/fireworks_transaction.c | 4 ++-- 15 files changed, 46 insertions(+), 43 deletions(-) -- 1.8.3.2

2 13

[alsa-devel] [PATCH] ASoC: wm_adsp: Use adsp_err/warn instead of dev_err/warn
by Charles Keepax 27 May '14

27 May '14

We have defines for adsp messages best to consistently use them. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 37 +++++++++++++++++-------------------- 1 files changed, 17 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ff15eec..0600271 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1543,16 +1543,16 @@ static void wm_adsp2_boot_work(struct work_struct *work) ret = regmap_read(dsp->regmap, dsp->base + ADSP2_CLOCKING, &val); if (ret != 0) { - dev_err(dsp->dev, "Failed to read clocking: %d\n", ret); + adsp_err(dsp, "Failed to read clocking: %d\n", ret); return; } if ((val & ADSP2_CLK_SEL_MASK) >= 3) { ret = regulator_enable(dsp->dvfs); if (ret != 0) { - dev_err(dsp->dev, - "Failed to enable supply: %d\n", - ret); + adsp_err(dsp, + "Failed to enable supply: %d\n", + ret); return; } @@ -1560,9 +1560,9 @@ static void wm_adsp2_boot_work(struct work_struct *work) 1800000, 1800000); if (ret != 0) { - dev_err(dsp->dev, - "Failed to raise supply: %d\n", - ret); + adsp_err(dsp, + "Failed to raise supply: %d\n", + ret); return; } } @@ -1672,15 +1672,15 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret = regulator_set_voltage(dsp->dvfs, 1200000, 1800000); if (ret != 0) - dev_warn(dsp->dev, - "Failed to lower supply: %d\n", - ret); + adsp_warn(dsp, + "Failed to lower supply: %d\n", + ret); ret = regulator_disable(dsp->dvfs); if (ret != 0) - dev_err(dsp->dev, - "Failed to enable supply: %d\n", - ret); + adsp_err(dsp, + "Failed to enable supply: %d\n", + ret); } list_for_each_entry(ctl, &dsp->ctl_list, list) @@ -1732,28 +1732,25 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); if (IS_ERR(adsp->dvfs)) { ret = PTR_ERR(adsp->dvfs); - dev_err(adsp->dev, "Failed to get DCVDD: %d\n", ret); + adsp_err(adsp, "Failed to get DCVDD: %d\n", ret); return ret; } ret = regulator_enable(adsp->dvfs); if (ret != 0) { - dev_err(adsp->dev, "Failed to enable DCVDD: %d\n", - ret); + adsp_err(adsp, "Failed to enable DCVDD: %d\n", ret); return ret; } ret = regulator_set_voltage(adsp->dvfs, 1200000, 1800000); if (ret != 0) { - dev_err(adsp->dev, "Failed to initialise DVFS: %d\n", - ret); + adsp_err(adsp, "Failed to initialise DVFS: %d\n", ret); return ret; } ret = regulator_disable(adsp->dvfs); if (ret != 0) { - dev_err(adsp->dev, "Failed to disable DCVDD: %d\n", - ret); + adsp_err(adsp, "Failed to disable DCVDD: %d\n", ret); return ret; } } -- 1.7.2.5

2 1

[alsa-devel] [PATCH v5] ASoC: sgtl5000: Fix the cache handling
by Fabio Estevam 27 May '14

27 May '14

Since commit e5d80e82e32e (ASoC: sgtl5000: Convert to use regmap directly) a kernel oops is observed after a suspend/resume sequence. The kernel oops happens inside sgtl5000_restore_regs() as codec->reg_cache is no longer a valid pointer. Add the remaining register entries into sgtl5000_reg_defaults[] and remove sgtl5000_restore_regs() completely, which allows suspend/resume to work fine and make the code simpler. Tested on a im53-qsb board. Reported-by: Shawn Guo <shawn.guo(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v4: - Remove sgtl5000_restore_regs() as suggested by Lars-Peter sound/soc/codecs/sgtl5000.c | 75 +++++++++------------------------------------ 1 file changed, 15 insertions(+), 60 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 9626ee0..3d39f0b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -36,18 +36,32 @@ /* default value of sgtl5000 registers */ static const struct reg_default sgtl5000_reg_defaults[] = { + { SGTL5000_CHIP_DIG_POWER, 0x0000 }, { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, { SGTL5000_CHIP_SSS_CTRL, 0x0010 }, + { SGTL5000_CHIP_ADCDAC_CTRL, 0x020c }, { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, + { SGTL5000_CHIP_ANA_ADC_CTRL, 0x0000 }, { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, + { SGTL5000_CHIP_LINREG_CTRL, 0x0000 }, + { SGTL5000_CHIP_REF_CTRL, 0x0000 }, + { SGTL5000_CHIP_MIC_CTRL, 0x0000 }, + { SGTL5000_CHIP_LINE_OUT_CTRL, 0x0000 }, { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, { SGTL5000_CHIP_ANA_POWER, 0x7060 }, { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, + { SGTL5000_CHIP_CLK_TOP_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_STATUS, 0x0000 }, + { SGTL5000_CHIP_SHORT_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_TEST2, 0x0000 }, + { SGTL5000_DAP_CTRL, 0x0000 }, + { SGTL5000_DAP_PEQ, 0x0000 }, { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, + { SGTL5000_DAP_AUDIO_EQ, 0x0000 }, { SGTL5000_DAP_SURROUND, 0x0040 }, { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, @@ -55,6 +69,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, + { SGTL5000_DAP_MIX_CHAN, 0x0000 }, { SGTL5000_DAP_AVC_CTRL, 0x0510 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, @@ -1068,71 +1083,11 @@ static int sgtl5000_suspend(struct snd_soc_codec *codec) return 0; } -/* - * restore all sgtl5000 registers, - * since a big hole between dap and regular registers, - * we will restore them respectively. - */ -static int sgtl5000_restore_regs(struct snd_soc_codec *codec) -{ - u16 *cache = codec->reg_cache; - u16 reg; - - /* restore regular registers */ - for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) { - - /* These regs should restore in particular order */ - if (reg == SGTL5000_CHIP_ANA_POWER || - reg == SGTL5000_CHIP_CLK_CTRL || - reg == SGTL5000_CHIP_LINREG_CTRL || - reg == SGTL5000_CHIP_LINE_OUT_CTRL || - reg == SGTL5000_CHIP_REF_CTRL) - continue; - - snd_soc_write(codec, reg, cache[reg]); - } - - /* restore dap registers */ - for (reg = SGTL5000_DAP_REG_OFFSET; reg < SGTL5000_MAX_REG_OFFSET; reg += 2) - snd_soc_write(codec, reg, cache[reg]); - - /* - * restore these regs according to the power setting sequence in - * sgtl5000_set_power_regs() and clock setting sequence in - * sgtl5000_set_clock(). - * - * The order of restore is: - * 1. SGTL5000_CHIP_CLK_CTRL MCLK_FREQ bits (1:0) should be restore after - * SGTL5000_CHIP_ANA_POWER PLL bits set - * 2. SGTL5000_CHIP_LINREG_CTRL should be set before - * SGTL5000_CHIP_ANA_POWER LINREG_D restored - * 3. SGTL5000_CHIP_REF_CTRL controls Analog Ground Voltage, - * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored - */ - snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, - cache[SGTL5000_CHIP_LINREG_CTRL]); - - snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, - cache[SGTL5000_CHIP_ANA_POWER]); - - snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, - cache[SGTL5000_CHIP_CLK_CTRL]); - - snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, - cache[SGTL5000_CHIP_REF_CTRL]); - - snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - cache[SGTL5000_CHIP_LINE_OUT_CTRL]); - return 0; -} - static int sgtl5000_resume(struct snd_soc_codec *codec) { /* Bring the codec back up to standby to enable regulators */ sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Restore registers by cached in memory */ - sgtl5000_restore_regs(codec); return 0; } #else -- 1.8.3.2

3 2

[alsa-devel] [PATCH] ASoC: jack: Fix multiple definition of `snd_soc_jack_add_gpiods'
by Jarkko Nikula 27 May '14

27 May '14

Commit f025d3b9c64e ("ASoC: jack: Add support for GPIO descriptor defined jack pins") caused build error when CONFIG_GPIOLIB is not set: sound/include/sound/soc.h:470: multiple definition of `snd_soc_jack_add_gpiods' sound/soc/soc-core.o:sound/include/sound/soc.h:470: first defined here make[2]: *** [sound/soc/snd-soc-core.o] Error 1 make[2]: *** Waiting for unfinished jobs.... make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Fix this by marking snd_soc_jack_add_gpiods() as static inline in soc.h. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- include/sound/soc.h | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 98dca42487bc..15ee2f71b92e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -464,9 +464,10 @@ static inline int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, return 0; } -int snd_soc_jack_add_gpiods(struct device *gpiod_dev, - struct snd_soc_jack *jack, - int count, struct snd_soc_jack_gpio *gpios) +static inline int snd_soc_jack_add_gpiods(struct device *gpiod_dev, + struct snd_soc_jack *jack, + int count, + struct snd_soc_jack_gpio *gpios) { return 0; } -- 2.0.0.rc2

2 1

[alsa-devel] [PATCH] ASoC: jack: Clarify GPIO descriptor lookup in struct snd_soc_jack_gpio doc
by Jarkko Nikula 27 May '14

27 May '14

Clarify struct snd_soc_jack_gpio documentation for the idx and name fields. Because name is passed as connection ID to gpiod_get_index() when using GPIO descriptor defined jack pins it is not only used as a label in debugfs but also as function name lookup in systems that support functions names for GPIOs. Clarify also idx since the index is within the function of the GPIO consumer device and not within the device itself only. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- include/sound/soc.h | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 15ee2f71b92e..72e056529745 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -599,9 +599,11 @@ struct snd_soc_jack_zone { * struct snd_soc_jack_gpio - Describes a gpio pin for jack detection * * @gpio: legacy gpio number - * @idx: gpio descriptor index within the GPIO consumer device + * @idx: gpio descriptor index within the function of the GPIO + * consumer device * @gpiod_dev GPIO consumer device - * @name: gpio name + * @name: gpio name. Also as connection ID for the GPIO consumer + * device function name lookup * @report: value to report when jack detected * @invert: report presence in low state * @debouce_time: debouce time in ms -- 2.0.0.rc2

2 1

[alsa-devel] [PATCH 1/1] sound/aoa/codecs/onyx.c: use static const for texts
by Fabian Frederick 27 May '14

27 May '14

'texts' is only used as source in strcpy Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiway(a)suse.de> Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- This is untested. sound/aoa/codecs/onyx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index f01bffb..401107b 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -241,7 +241,7 @@ static struct snd_kcontrol_new inputgain_control = { static int onyx_snd_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "Line-In", "Microphone" }; + static const char * const texts[] = { "Line-In", "Microphone" }; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; -- 1.9.1

2 1

[alsa-devel] [PATCH] pcm: route: Correctly close slave pcm when no matching chmap is found
by David Henningsson 27 May '14

27 May '14

This patch fixes a bug where the slave pcm was not correctly closed on some error conditions, such as not finding a matching chmap. Reported-by: Raymond Yau <superquad.vortex2(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/pcm/pcm_route.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index 49567ea..751e36f 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -1429,8 +1429,10 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (tt_chmap) { err = find_matching_chmap(spcm, tt_chmap, &chmap, &schannels); free(tt_chmap); - if (err < 0) + if (err < 0) { + snd_pcm_close(spcm); return err; + } } err = _snd_pcm_route_determine_ttable(tt, &csize, &ssize, chmap); -- 1.9.1

2 1

[alsa-devel] [PATCH 1/5] ALSA: hda - Add fixup_forced flag
by Hui Wang 27 May '14

27 May '14

From: David Henningsson <david.henningsson(a)canonical.com> The "fixup_forced" flag will indicate whether a specific fixup (or nofixup) has been set by the user, to override the driver's default. This flag will help future patches. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 9 ++++++--- sound/pci/hda/hda_codec.h | 1 + 2 files changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 90d2fda..36961ab 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -852,15 +852,17 @@ void snd_hda_pick_fixup(struct hda_codec *codec, if (codec->modelname && !strcmp(codec->modelname, "nofixup")) { codec->fixup_list = NULL; codec->fixup_id = -1; + codec->fixup_forced = 1; return; } if (codec->modelname && models) { while (models->name) { if (!strcmp(codec->modelname, models->name)) { - id = models->id; - name = models->name; - break; + codec->fixup_id = models->id; + codec->fixup_name = models->name; + codec->fixup_forced = 1; + return; } models++; } @@ -889,6 +891,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, } } + codec->fixup_forced = 0; codec->fixup_id = id; if (id >= 0) { codec->fixup_list = fixlist; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index a423313..5825aa1 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -402,6 +402,7 @@ struct hda_codec { /* fix-up list */ int fixup_id; + unsigned int fixup_forced:1; /* fixup explicitly set by user */ const struct hda_fixup *fixup_list; const char *fixup_name; -- 1.8.1.2

4 15

[alsa-devel] [PATCH] hdajackretask: Make sure codecs do not show up twice under 3.15 kernel
by David Henningsson 27 May '14

27 May '14

On a 3.15 based kernel, both "hdaudioC*D*" and "hwC*D*" shows up under /sys/class/sound, and pointing to the same object. Make sure we only match on the hwC*D* name to avoid duplicate entries. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hdajackretask/sysfs-pin-configs.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/hdajackretask/sysfs-pin-configs.c b/hdajackretask/sysfs-pin-configs.c index 4044989..5bae0f9 100644 --- a/hdajackretask/sysfs-pin-configs.c +++ b/hdajackretask/sysfs-pin-configs.c @@ -20,8 +20,10 @@ int get_codec_name_list(codec_name_t* names, int entries) const gchar* dirname = g_dir_read_name(sysdir); if (!dirname) break; + /* Split e g "hwC0D1" into "hw", "0" and "1" */ cd = g_strsplit_set(dirname, "CD", 9); ok = g_strv_length(cd) == 3; + ok &= strcmp(cd[0], "hw") == 0; if (ok) { gchar* filetest = g_strdup_printf("/sys/class/sound/%s/init_pin_configs", dirname); ok = g_file_test(filetest, G_FILE_TEST_IS_REGULAR); -- 1.9.1

2 1

[alsa-devel] [PATCH v4 00/17] ASoC: fsl-ssi: Driver cleanup
by Markus Pargmann 27 May '14

27 May '14

Hi, This series is a cleanup of the fsl-ssi driver. v4 includes the changes I made to fix the baudclock handling. I moved the baudclock enable/disable function calls into hw_params()/hw_free(). To avoid inconsistent reference counters for the baudclock I use a variable baudclk_streams which describes the active streams to be able to decide if we have to enable/disable the clock and set the rate. I currently don't have a board to test i2s-master, maybe I am able to test it next week. The new patch which includes these changes is "ASoC: fsl-ssi: Fix baudclock handling". Best regards, Markus Changes in v4: - New patch to fix baudclock handling (enable/disable/set_rate) Changes in v3: - Some new patches to improve/fix i2s master mode by Sascha - baudclock is enabled/disabled in startup/shutdown now - bitclock setup moved to a seperate function (not set_dai_sysclk) - Regmap config changed to NATIVE now. Markus Pargmann (12): ASoC: fsl-ssi: Fix register values when disabling ASoC: fsl-ssi: Move debugging to seperate file ASoC: fsl-ssi: Use dev_name for DAI driver struct ASoC: fsl-ssi: Move imx-specific probe to seperate function ASoC: fsl-ssi: Remove useless DMA code ASoC: fsl-ssi: Cleanup probe function ASoC: fsl-ssi: Remove unnecessary variables from ssi_private ASoC: fsl-ssi: make fsl,mode property optional ASoC: fsl-ssi: Transmit enable synchronization ASoC: fsl-ssi: Fix baudclock handling ASoC: fsl-ssi: reorder and document fsl_ssi_private ASoC: fsl-ssi: Use regmap Sascha Hauer (5): ASoC: fsl-ssi: introduce SoC specific data ASoC: fsl-ssi: Move fsl_ssi_set_dai_sysclk above fsl_ssi_hw_params ASoC: fsl-ssi: set bitclock in master mode from hw_params ASoC: fsl-ssi: remove unnecessary spinlock ASoC: fsl-ssi: Set framerate divider correctly for i2s master mode sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 3 +- sound/soc/fsl/fsl_ssi.c | 1259 +++++++++++++++++++------------------------ sound/soc/fsl/fsl_ssi.h | 112 +++- sound/soc/fsl/fsl_ssi_dbg.c | 163 ++++++ 5 files changed, 815 insertions(+), 723 deletions(-) create mode 100644 sound/soc/fsl/fsl_ssi_dbg.c -- 1.9.1

3 25

[alsa-devel] [PATCH v2] ALSA: hda: fix tegra build
by Arnd Bergmann 27 May '14

27 May '14

When CONFIG_PM is disabled, the CONFIG_SND_HDA_POWER_SAVE_DEFAULT symbol does not get defined, which causes a build error for the hda-tegra driver: hda/hda_tegra.c:80:25: error: 'CONFIG_SND_HDA_POWER_SAVE_DEFAULT' undeclared here (not in a function) static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; ^ /git/arm-soc/sound/pci/hda/hda_tegra.c:235:13: warning: 'hda_tegra_disable_clocks' defined but not used [-Wunused-function] static void hda_tegra_disable_clocks(struct hda_tegra *data) ^ This works around the problem by not referencing that macro when CONFIG_PM is disabled. Instead, we assume that it's disabled unconditionally and cannot be enabled at runtime. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Dylan Reid <dgreid(a)chromium.org> Cc: Stephen Warren <swarren(a)nvidia.com> Cc: Takashi Iwai <tiwai(a)suse.de> --- Same patch, but this time with correct Signed-off-by line. diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 086fd06..a366ba9 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -77,10 +77,14 @@ struct hda_tegra { void __iomem *regs; }; +#ifdef CONFIG_PM static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; module_param(power_save, bint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout (in seconds, 0 = disable)."); +#else +static int power_save = 0; +#endif /* * DMA page allocation ops.

2 1

[alsa-devel] aplay: main:722: audio open error: No such file or directory
by Christopher Head 27 May '14

27 May '14

Hello! Apologies if this is not the right place to report this. I tried to use the bug tracker at <https://bugtrack.alsa-project.org/alsa-bug>, but both times I went to it (a couple of weeks ago and again today) it was offline. Also, I am not subscribed to the list, so please include me in replies (or tell me if this is not acceptable and I will subscribe). I can't play audio on non-default devices. Running aplay with no options, or with -D default, works fine. Running aplay with -D hdmi does not work, failing with this message: aplay: main:722: audio open error: No such file or directory Running it in strace, I see that the last interesting thing it does before printing that message is it tries to open /dev/snd/pcmC0D3p, which does not exist; it should probably be addressing pcmC1D3p instead. I have no ALSA configuration files anywhere. kruskal asound # aplay -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=MID HDA Intel MID, ALC663 Analog Default Audio Device sysdefault:CARD=MID HDA Intel MID, ALC663 Analog Default Audio Device front:CARD=MID,DEV=0 HDA Intel MID, ALC663 Analog Front speakers surround40:CARD=MID,DEV=0 HDA Intel MID, ALC663 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=MID,DEV=0 HDA Intel MID, ALC663 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=MID,DEV=0 HDA Intel MID, ALC663 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=MID,DEV=0 HDA Intel MID, ALC663 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=MID,DEV=0 HDA Intel MID, ALC663 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=MID,DEV=0 HDA Intel MID, ALC663 Digital IEC958 (S/PDIF) Digital Audio Output hdmi:CARD=NVidia,DEV=0 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=1 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=2 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 HDMI Audio Output kruskal asound # ls -1 /dev/snd by-path controlC0 controlC1 hwC0D0 hwC1D0 hwC1D1 hwC1D2 hwC1D3 pcmC0D0c pcmC0D0p pcmC0D1p pcmC0D2c pcmC1D3p pcmC1D7p pcmC1D8p pcmC1D9p seq timer kruskal asound # cat cards 0 [MID ]: HDA-Intel - HDA Intel MID HDA Intel MID at 0xf4900000 irq 53 1 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xf3080000 irq 17 kruskal asound # cat devices 1: : sequencer 2: [ 0- 2]: digital audio capture 3: [ 0- 1]: digital audio playback 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0- 0]: hardware dependent 7: [ 0] : control 8: [ 1- 9]: digital audio playback 9: [ 1- 8]: digital audio playback 10: [ 1- 7]: digital audio playback 11: [ 1- 3]: digital audio playback 12: [ 1- 3]: hardware dependent 13: [ 1- 2]: hardware dependent 14: [ 1- 1]: hardware dependent 15: [ 1- 0]: hardware dependent 16: [ 1] : control 33: : timer kruskal asound # cat pcm 00-00: ALC663 Analog : ALC663 Analog : playback 1 : capture 1 00-01: ALC663 Digital : ALC663 Digital : playback 1 00-02: ALC663 Alt Analog : ALC663 Alt Analog : capture 1 01-03: HDMI 0 : HDMI 0 : playback 1 01-07: HDMI 0 : HDMI 0 : playback 1 01-08: HDMI 0 : HDMI 0 : playback 1 01-09: HDMI 0 : HDMI 0 : playback 1 -- Christopher Head

2 2

[alsa-devel] [RFC][PATCH 00/13 v1] Enhancement for Dice driver
by Takashi Sakamoto 27 May '14

27 May '14

This patchset is for enhancement of Dice driver. This patchset is based on this series. (This series is under reviewing.) [alsa-devel] [PATCH 00/49 v4] Enhancement for support of Firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-April/075841.html If you try this patchset, please apply the series in advance then apply this patchset. Or just use backport codes in my github repository: https://github.com/takaswie/snd-firewire-improve Or if you use Ubuntu 14.04, DKMS package is available from David's PPA. (Special thanks to David Henningsson.) https://launchpad.net/~diwic/+archive/snd-firewire-improve If you have any Dice based devices, please test this patchset because I can't test this driver with actual devices. == Background Dice is 'Digital Interface Communications Engine', produced by TC Applied Technologies. Several vendors produced devices with Dice chipsets. Currently, ALSA driver for Dice chipset has some restrictions: - Support playback of PCM samples only - Support SYT-Match mode for sampling clock source only This patchset adds more functionality into the driver: - Support both of capture/playback - Support both of PCM/MIDI - Support both of SYT-Match or non SYT-Match mode == Call for testing I don't have any Dice based devices. So I work for this patchset with my experiences for the other drivers and some stub codes. So it may not work fine, perhaps. If you have some Dice based devices, I hope you to test these patches and report the result. If it works badly, please get transaction logs and send it to me. The way to get transaction logs is: 1. Set debug parameter of 'firewire-ohci' driver. With root privillege: $ echo 1 > /sys/module/firewire_ohci/parameters/debug 2. Then transaction logs are output to /var/log/syslog $ tail -n 10 /var/log/syslog kernel: [ 2376.107790] snd_dice fw1.0: transaction failed: address error kernel: [ 2376.108015] firewire_core 0000:02:00.0: created device fw1: GUID 00a0de000002e247, S400 kernel: [ 2382.044469] firewire_ohci 0000:02:00.0: AR evt_bus_reset, generation 25 kernel: [ 2384.095868] snd_dice fw1.0: transaction failed: bus reset kernel: [ 2384.143898] snd_dice fw1.0: transaction failed: bus reset kernel: [ 2389.517686] firewire_ohci 0000:02:00.0: AR evt_bus_reset, generation 26 kernel: [ 2390.015821] firewire_ohci 0000:02:00.0: AT spd 0 tl 1f, ffc0 -> ffc1, evt_missing_ack, QR req, fffff0000400 kernel: [ 2392.885455] firewire_ohci 0000:02:00.0: AR evt_bus_reset, generation 27 kernel: [ 2392.885504] firewire_core 0000:02:00.0: phy config: new root=ffc1, gap_count=5 kernel: [ 2392.885534] firewire_ohci 0000:02:00.0: AT ack_complete, PHY 01c50000 fe3affff kernel: [ 2392.885556] firewire_ohci 0000:02:00.0: AR evt_bus_reset, generation 28 kernel: [ 2393.025757] firewire_ohci 0000:02:00.0: AT spd 0 tl 2a, ffc1 -> ffc0, ack_pending , QR req, fffff0000400 kernel: [ 2393.026007] firewire_ohci 0000:02:00.0: AR spd 2 tl 2a, ffc0 -> ffc1, ack_complete, QR resp = 041fa8cb kernel: [ 2393.026076] firewire_ohci 0000:02:00.0: AT spd 0 tl 2b, ffc1 -> ffc0, ack_pending , QR req, fffff0000404 kernel: [ 2393.026172] firewire_ohci 0000:02:00.0: AR spd 2 tl 2b, ffc0 -> ffc1, ack_complete, QR resp = 31333934 kernel: [ 2393.026251] firewire_ohci 0000:02:00.0: AT spd 0 tl 2c, ffc1 -> ffc0, ack_pending , QR req, fffff0000408 kernel: [ 2393.030558] firewire_ohci 0000:02:00.0: AR spd 2 tl 2c, ffc0 -> ffc1, ack_complete, QR resp = f0646122 3. Please send me the output since you plug-in your device (not whole) 4. And please tell me how you test, which devices you test with and what result you got == All patches My work for ALSA firewire drivers are available in my remote repository: https://github.com/takaswie/sound It includes 77 patches. I'm always welcome to receive some comments about them. Takashi Sakamoto (13): dice: Rename structure and its members dice: Move file to its own directory dice: Split transaction functionality into a file dice: Split stream functionality into a file dice: Split PCM functionality into a file dice: Split hwdep functionality into a file dice: Split proc functionality into a file dice: Add new functions for limitation of PCM parameters dice: Change the way to start stream dice: Add support for duplex streams with synchronization dice: Support for non SYT-Match sampling clock source mode dice: Add support for PCM capture dice: Add support for MIDI capture/playback sound/firewire/Kconfig | 4 +- sound/firewire/Makefile | 3 +- sound/firewire/dice-interface.h | 371 -------- sound/firewire/dice.c | 1500 -------------------------------- sound/firewire/dice/Makefile | 3 + sound/firewire/dice/dice-interface.h | 371 ++++++++ sound/firewire/dice/dice.c | 374 ++++++++ sound/firewire/dice/dice.h | 189 ++++ sound/firewire/dice/dice_hwdep.c | 190 ++++ sound/firewire/dice/dice_midi.c | 173 ++++ sound/firewire/dice/dice_pcm.c | 389 +++++++++ sound/firewire/dice/dice_proc.c | 252 ++++++ sound/firewire/dice/dice_stream.c | 410 +++++++++ sound/firewire/dice/dice_transaction.c | 359 ++++++++ 14 files changed, 2712 insertions(+), 1876 deletions(-) delete mode 100644 sound/firewire/dice-interface.h delete mode 100644 sound/firewire/dice.c create mode 100644 sound/firewire/dice/Makefile create mode 100644 sound/firewire/dice/dice-interface.h create mode 100644 sound/firewire/dice/dice.c create mode 100644 sound/firewire/dice/dice.h create mode 100644 sound/firewire/dice/dice_hwdep.c create mode 100644 sound/firewire/dice/dice_midi.c create mode 100644 sound/firewire/dice/dice_pcm.c create mode 100644 sound/firewire/dice/dice_proc.c create mode 100644 sound/firewire/dice/dice_stream.c create mode 100644 sound/firewire/dice/dice_transaction.c -- 1.8.3.2

4 19

[alsa-devel] [PATCH] ALSA: hda: fix tegra build
by Arnd Bergmann 26 May '14

26 May '14

When CONFIG_PM is disabled, the CONFIG_SND_HDA_POWER_SAVE_DEFAULT symbol does not get defined, which causes a build error for the hda-tegra driver: hda/hda_tegra.c:80:25: error: 'CONFIG_SND_HDA_POWER_SAVE_DEFAULT' undeclared here (not in a function) static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; ^ /git/arm-soc/sound/pci/hda/hda_tegra.c:235:13: warning: 'hda_tegra_disable_clocks' defined but not used [-Wunused-function] static void hda_tegra_disable_clocks(struct hda_tegra *data) ^ This works around the problem by not referencing that macro when CONFIG_PM is disabled. Instead, we assume that it's disabled unconditionally and cannot be enabled at runtime. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> Reviewed-by: Stephen Warren <swarren(a)nvidia.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 086fd06..a366ba9 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -77,10 +77,14 @@ struct hda_tegra { void __iomem *regs; }; +#ifdef CONFIG_PM static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; module_param(power_save, bint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout (in seconds, 0 = disable)."); +#else +static int power_save = 0; +#endif /* * DMA page allocation ops.

2 2

[alsa-devel] [PATCH 1/2] ASoC: add api for dapm kcontrol configiuration
by Vinod Koul 26 May '14

26 May '14

From: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> For DSPs we need to set and get the value for snd_kcontrol. This is currently done by dapm_kcontrol_set/get_value, so create a wrapper snd_soc_dapm_kcontrol_get/set_value APIs to let drivers use this Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/soc-dapm.h | 4 ++++ sound/soc/soc-dapm.c | 13 +++++++++++++ 2 files changed, 17 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 75020f5..9e826e9 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -368,6 +368,10 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_clock_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +bool snd_soc_dapm_kcontrol_set_value( + const struct snd_kcontrol *kcontrol, unsigned int value); +unsigned int snd_soc_dapm_kcontrol_get_value( + const struct snd_kcontrol *kcontrol); /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 98c1dc6..b7dfa1a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -333,6 +333,12 @@ static unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) return data->value; } +unsigned int snd_soc_dapm_kcontrol_get_value( + const struct snd_kcontrol *kcontrol) +{ + dapm_kcontrol_get_value(kcontrol); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_get_value); static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, unsigned int value) @@ -350,6 +356,13 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, return true; } +bool snd_soc_dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, + unsigned int value) +{ + return dapm_kcontrol_set_value(kcontrol, value); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_set_value); + /** * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol * @kcontrol: The kcontrol -- 1.7.0.4

4 13

[alsa-devel] Fwd: Re: No sound for a rather old laptop (ALC260).
by Fidel Leon Martin 26 May '14

26 May '14

Hello, list. Is there any way to know if this support request has been taken by anyone...? ---------- Forwarded Message ---------- Subject: Re: [alsa-devel] No sound for a rather old laptop (ALC260). Date: Tuesday 06 May 2014, 16:40:56 From: Fidel Leon <fidelleon(a)mykolab.com> To: Takashi Iwai <tiwai(a)suse.de>, alsa-devel(a)alsa-project.org On Tuesday 06 May 2014 11:27:07 you wrote: > At Mon, 05 May 2014 16:34:38 +0200, > > Fidel Leon wrote: > > I own a rather old laptop which features a snd_hda_intel soundcard, with a > > ALC260 chipset. > > > > I know it works when using rather old kernels (Debian 7, for example), and > > > > I already have an alsa-info.txt file to debug, may I post it here? > > Please attach in your post. (But maybe better to compress so that it > won't hit the max size of ML.) Also, at best, take alsa-info.sh from > the working state with the old kernel, so that we can compare > directly. Many thanks, Takashi. I've tested a Debian 6 live image and my soundcard runs fine, so I can tell my soundcard works. Attached are alsa-infos from kernel 2.6 (which works) and 3.14 (Fedora 20, which doesnt work). Regards, ----------------------------------------- -- Fidel Leon fidelleon(a)mykolab.com GPG: AD74FB97

2 5

[alsa-devel] [PATCH] ASoC: samsung: Use params_width()
by Tushar Behera 26 May '14

26 May '14

commit 8c5178fca4ce ("ALSA: Add params_width() helpers") introduces a helper to get the sample width. Updating Samsung related sound drivers to use this helper. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- sound/soc/samsung/i2s.c | 8 ++++---- sound/soc/samsung/pcm.c | 4 ++-- sound/soc/samsung/s3c-i2s-v2.c | 8 ++++---- sound/soc/samsung/s3c2412-i2s.c | 6 +++--- sound/soc/samsung/s3c24xx-i2s.c | 6 +++--- sound/soc/samsung/smdk_wm8580.c | 8 +++----- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/spdif.c | 4 ++-- 8 files changed, 22 insertions(+), 24 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 07ff3e7..6b96168 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -672,8 +672,8 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, if (is_manager(i2s)) mod &= ~MOD_BLC_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: + switch (params_width(params)) { + case 8: if (is_secondary(i2s)) mod |= MOD_BLCS_8BIT; else @@ -681,7 +681,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, if (is_manager(i2s)) mod |= MOD_BLC_8BIT; break; - case SNDRV_PCM_FORMAT_S16_LE: + case 16: if (is_secondary(i2s)) mod |= MOD_BLCS_16BIT; else @@ -689,7 +689,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, if (is_manager(i2s)) mod |= MOD_BLC_16BIT; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: if (is_secondary(i2s)) mod |= MOD_BLCS_24BIT; else diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index a3c9c9c..4c5f97f 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -283,8 +283,8 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); /* Strictly check for sample size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; default: return -EINVAL; diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 77a2ae5..0ff4bbe 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -322,13 +322,13 @@ static int s3c_i2sv2_hw_params(struct snd_pcm_substream *substream, iismod &= ~S3C64XX_IISMOD_BLC_MASK; /* Sample size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: + switch (params_width(params)) { + case 8: iismod |= S3C64XX_IISMOD_BLC_8BIT; break; - case SNDRV_PCM_FORMAT_S16_LE: + case 16: break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iismod |= S3C64XX_IISMOD_BLC_24BIT; break; } diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 843f315..08c059b 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -120,11 +120,11 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, iismod = readl(i2s->regs + S3C2412_IISMOD); pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: + switch (params_width(params)) { + case 8: iismod |= S3C2412_IISMOD_8BIT; break; - case SNDRV_PCM_FORMAT_S16_LE: + case 16: iismod &= ~S3C2412_IISMOD_8BIT; break; } diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 4a6d206..9aba9fb 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -248,12 +248,12 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); pr_debug("hw_params r: IISMOD: %x\n", iismod); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: + switch (params_width(params)) { + case 8: iismod &= ~S3C2410_IISMOD_16BIT; dma_data->dma_size = 1; break; - case SNDRV_PCM_FORMAT_S16_LE: + case 16: iismod |= S3C2410_IISMOD_16BIT; dma_data->dma_size = 2; break; diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 7a16b32..b1a519f 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -37,13 +37,11 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, unsigned int pll_out; int bfs, rfs, ret; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_U8: - case SNDRV_PCM_FORMAT_S8: + switch (params_width(params)) { + case 8: bfs = 16; break; - case SNDRV_PCM_FORMAT_U16_LE: - case SNDRV_PCM_FORMAT_S16_LE: + case 16: bfs = 32; break; default: diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index fc25cc0..3d6272a 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -57,7 +57,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, int ret; /* AIF1CLK should be >=3MHz for optimal performance */ - if (params_format(params) == SNDRV_PCM_FORMAT_S24_LE) + if (params_width(params) == 24) pll_out = params_rate(params) * 384; else if (params_rate(params) == 8000 || params_rate(params) == 11025) pll_out = params_rate(params) * 512; diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index e93a93e..d9ffc48 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -211,8 +211,8 @@ static int spdif_hw_params(struct snd_pcm_substream *substream, con |= CON_PCM_DATA; con &= ~CON_PCM_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: con |= CON_PCM_16BIT; break; default: -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: sirf-audio-codec: Simplify the new bitmask value in regmap_update_bits
by Axel Lin 26 May '14

26 May '14

Having the binary ones complement operator in the new bitmak value makes the code hard to read. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- I think the binary ones complement operator in such use case is error prone: For example: To set BIT(1) and clear both BIT(2) and BIT(3): The following code has wrong result: regmap_update_bits(regmap, the_register, BIT(1), BIT(2), BIT(3), BIT(1) | ~BIT(2) | ~BIT(3)); Axel sound/soc/codecs/sirf-audio-codec.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index c5177bc..d90cb0f 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -109,7 +109,7 @@ static void enable_and_reset_codec(struct regmap *regmap, { regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1, codec_enable_bits | codec_reset_bits, - codec_enable_bits | ~codec_reset_bits); + codec_enable_bits); msleep(20); regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1, codec_reset_bits, codec_reset_bits); @@ -128,8 +128,7 @@ static int atlas6_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMD: regmap_update_bits(sirf_audio_codec->regmap, - AUDIO_IC_CODEC_CTRL1, ATLAS6_CODEC_ENABLE_BITS, - ~ATLAS6_CODEC_ENABLE_BITS); + AUDIO_IC_CODEC_CTRL1, ATLAS6_CODEC_ENABLE_BITS, 0); break; default: break; @@ -151,8 +150,7 @@ static int prima2_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMD: regmap_update_bits(sirf_audio_codec->regmap, - AUDIO_IC_CODEC_CTRL1, PRIMA2_CODEC_ENABLE_BITS, - ~PRIMA2_CODEC_ENABLE_BITS); + AUDIO_IC_CODEC_CTRL1, PRIMA2_CODEC_ENABLE_BITS, 0); break; default: break; -- 1.8.3.2

2 2

[alsa-devel] [PATCH RFC 03/11] sound: ASoC: kirkwood: Remove ARCH_KIRKWOOD dependency
by Andrew Lunn 26 May '14

26 May '14

mach-kirkwood has been removed, now that kirkwood lives in mach-mvebu. Signed-off-by: Andrew Lunn <andrew(a)lunn.ch> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/kirkwood/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 1f7c7ee3527a..6a0f79d2bb33 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,6 @@ config SND_KIRKWOOD_SOC tristate "SoC Audio for the Marvell Kirkwood and Dove chips" - depends on ARCH_KIRKWOOD || ARCH_DOVE || ARCH_MVEBU || MACH_KIRKWOOD || COMPILE_TEST + depends on ARCH_DOVE || ARCH_MVEBU || MACH_KIRKWOOD || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the -- 2.0.0.rc2

3 3

[alsa-devel] [PATCH 1/4] ASoC: tegra: Replace instances of rtd->codec->card with rtd->card
by Lars-Peter Clausen 26 May '14

26 May '14

No need to go via the CODEC to get a pointer to the card. This will help to eventually remove the card field from the snd_soc_codec struct. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/tegra/tegra_alc5632.c | 5 ++--- sound/soc/tegra/tegra_max98090.c | 5 ++--- sound/soc/tegra/tegra_rt5640.c | 5 ++--- sound/soc/tegra/tegra_wm8753.c | 3 +-- sound/soc/tegra/tegra_wm8903.c | 5 ++--- sound/soc/tegra/trimslice.c | 3 +-- 6 files changed, 10 insertions(+), 16 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index c61ea3a..21ad9bd 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -41,8 +41,7 @@ static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -105,7 +104,7 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card); + struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card); snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &tegra_alc5632_hs_jack); diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 0283cfb..b867b27 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -49,8 +49,7 @@ static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -127,7 +126,7 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct tegra_max98090 *machine = snd_soc_card_get_drvdata(codec->card); + struct tegra_max98090 *machine = snd_soc_card_get_drvdata(rtd->card); if (gpio_is_valid(machine->gpio_hp_det)) { snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 4511c5a..d720bd7 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -51,8 +51,7 @@ static int tegra_rt5640_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -110,7 +109,7 @@ static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(codec->card); + struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(rtd->card); snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, &tegra_rt5640_hp_jack); diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index 8e774d1..769e28f 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -55,8 +55,7 @@ static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 4ac7373..e795424 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -60,8 +60,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -173,7 +172,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); if (gpio_is_valid(machine->gpio_hp_det)) { diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 734bfcd..589d2d9 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -50,8 +50,7 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); int srate, mclk; int err; -- 1.8.0

4 8