October 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: Intel: switch from ioremap_cache to memremap" to the asoc tree
by Mark Brown 23 Oct '15

23 Oct '15

The patch ASoC: Intel: switch from ioremap_cache to memremap has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ba40a854ea4f9d81368dc023bd25ac4073058039 Mon Sep 17 00:00:00 2001 From: Dan Williams <dan.j.williams(a)intel.com> Date: Fri, 9 Oct 2015 18:16:36 -0400 Subject: [PATCH] ASoC: Intel: switch from ioremap_cache to memremap In preparation for deprecating ioremap_cache() convert its usage in skl-nhlt to memremap. Signed-off-by: Dan Williams <dan.j.williams(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-nhlt.c | 10 +++++----- sound/soc/intel/skylake/skl.h | 6 +++--- 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 13036b1..b0c7bd1 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -25,7 +25,7 @@ static u8 OSC_UUID[16] = {0x6E, 0x88, 0x9F, 0xA6, 0xEB, 0x6C, 0x94, 0x45, #define DSDT_NHLT_PATH "\\_SB.PCI0.HDAS" -void __iomem *skl_nhlt_init(struct device *dev) +void *skl_nhlt_init(struct device *dev) { acpi_handle handle; union acpi_object *obj; @@ -40,17 +40,17 @@ void __iomem *skl_nhlt_init(struct device *dev) if (obj && obj->type == ACPI_TYPE_BUFFER) { nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer; - return ioremap_cache(nhlt_ptr->min_addr, nhlt_ptr->length); + return memremap(nhlt_ptr->min_addr, nhlt_ptr->length, + MEMREMAP_WB); } dev_err(dev, "device specific method to extract NHLT blob failed\n"); return NULL; } -void skl_nhlt_free(void __iomem *addr) +void skl_nhlt_free(void *addr) { - iounmap(addr); - addr = NULL; + memunmap(addr); } static struct nhlt_specific_cfg *skl_get_specific_cfg( diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index e980d78..dd2e79a 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -62,7 +62,7 @@ struct skl { unsigned int init_failed:1; /* delayed init failed */ struct platform_device *dmic_dev; - void __iomem *nhlt; /* nhlt ptr */ + void *nhlt; /* nhlt ptr */ struct skl_sst *skl_sst; /* sst skl ctx */ struct skl_dsp_resource resource; @@ -83,8 +83,8 @@ struct skl_dma_params { int skl_platform_unregister(struct device *dev); int skl_platform_register(struct device *dev); -void __iomem *skl_nhlt_init(struct device *dev); -void skl_nhlt_free(void __iomem *addr); +void *skl_nhlt_init(struct device *dev); +void skl_nhlt_free(void *addr); struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance, u8 link_type, u8 s_fmt, u8 no_ch, u32 s_rate, u8 dirn); -- 2.6.1

1 0

[alsa-devel] [PATCH v2 0/3] ASoC: dapm: add PGA kcontrol, dai_link enhancement &
by Vinod Koul 23 Oct '15

23 Oct '15

This adds few small patches in DAPM which we require for Intel DSPs First adds support for having kcontrol on PGAs. This is required for modules which are modelled as PGA but can be user configurable. Next is new callback on snd_soc_dai_link_events(). Lastly a macro to return widget associated with a kcontrol Jeeja KP (2): ASoC: dapm: Add kcontrol support for PGAs ASoC: dapm: Add startup & shutdown for dai_links Mythri P K (1): ASoC: dapm: Add snd_soc_dapm_kcontrol_widget() include/sound/soc-dapm.h | 3 +++ sound/soc/soc-dapm.c | 59 +++++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 56 insertions(+), 6 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH 0/3] ASoC: dapm: add PGA kcontrol, dai_link enhancement &
by Vinod Koul 23 Oct '15

23 Oct '15

This adds few small patches in DAPM which we require for Intel DSPs Firsts adds support for having kcontrol on PGAs. This is required for modules which are modelled as PGA but cna be user configurable. Next is new callback on snd_soc_dai_link_events(). Lastly a macro to return widget associated with a kcontrol Jeeja KP (2): ASoC: dapm: Add kcontrol support for PGAs ASoC: dapm: Add startup & shutdown for dai_links Mythri P K (1): ASoC: dapm: Add snd_soc_dapm_kcontrol_widget() include/sound/soc-dapm.h | 3 +++ sound/soc/soc-dapm.c | 64 ++++++++++++++++++++++++++++++++++++++++++------ 2 files changed, 60 insertions(+), 7 deletions(-) -- 1.9.1

3 7

[alsa-devel] [PATCH 1/2] ASoC: nau8825: Add FLL configuration
by Ben Zhang 23 Oct '15

23 Oct '15

snd_soc_codec_driver.set_pll is implemented to configure the FLL. The codec internal SYSCLK can be from either the MCLK pin directly, or the FLL. This is configured by snd_soc_codec_driver.set_pll. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 162 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8825.h | 28 ++++++-- 2 files changed, 185 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 5c1badf..010f50b 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -29,6 +29,58 @@ #include "nau8825.h" +#define NAU_FREF_MAX 13500000 +#define NAU_FVCO_MAX 100000000 +#define NAU_FVCO_MIN 90000000 + +struct nau88xx_fll { + int mclk_src; + int ratio; + int fll_frac; + int fll_int; + int clk_ref_div; +}; + +struct nau88xx_fll_attr { + unsigned int param; + unsigned int val; +}; + +/* scaling for mclk from sysclk_src output */ +static const struct nau88xx_fll_attr mclk_src_scaling[] = { + { 1, 0x0 }, + { 2, 0x2 }, + { 4, 0x3 }, + { 8, 0x4 }, + { 16, 0x5 }, + { 32, 0x6 }, + { 3, 0x7 }, + { 6, 0xa }, + { 12, 0xb }, + { 24, 0xc }, + { 48, 0xd }, + { 96, 0xe }, + { 5, 0xf }, +}; + +/* ratio for input clk freq */ +static const struct nau88xx_fll_attr fll_ratio[] = { + { 512000, 0x01 }, + { 256000, 0x02 }, + { 128000, 0x04 }, + { 64000, 0x08 }, + { 32000, 0x10 }, + { 8000, 0x20 }, + { 4000, 0x40 }, +}; + +static const struct nau88xx_fll_attr fll_pre_scalar[] = { + { 1, 0x0 }, + { 2, 0x1 }, + { 4, 0x2 }, + { 8, 0x3 }, +}; + static const struct reg_default nau8825_reg_defaults[] = { { NAU8825_REG_ENA_CTRL, 0x00ff }, { NAU8825_REG_CLK_DIVIDER, 0x0050 }, @@ -808,6 +860,115 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) return 0; } +/** + * nau88xx_calc_fll_param - Calculate FLL parameters. + * @fll_in: external clock provided to codec. + * @fs: sampling rate. + * @fll_param: Pointer to structure of FLL parameters. + * + * Calculate FLL parameters to configure codec. + * + * Returns 0 for success or negative error code. + */ +static int nau88xx_calc_fll_param(unsigned int fll_in, unsigned int fs, + struct nau88xx_fll *fll_param) +{ + u64 fref, fvco, fll_input; + int i; + + /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing + * freq_in by 1, 2, 4, or 8 using FLL pre-scalar. + * FREF = freq_in / NAU8825_FLL_REF_DIV_MASK + */ + for (i = 0; i < ARRAY_SIZE(fll_pre_scalar); i++) { + fref = fll_in / fll_pre_scalar[i].param; + if (fref <= NAU_FREF_MAX) + break; + } + if (i == ARRAY_SIZE(fll_pre_scalar)) + return -EINVAL; + fll_param->clk_ref_div = fll_pre_scalar[i].val; + + /* Choose the FLL ratio based on FREF */ + for (i = 0; i < ARRAY_SIZE(fll_ratio); i++) { + if (fref >= fll_ratio[i].param) + break; + } + if (i == ARRAY_SIZE(fll_ratio)) + return -EINVAL; + fll_param->ratio = fll_ratio[i].val; + + /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. + * FDCO must be within the 90MHz - 100MHz or the FFL cannot be + * guaranteed across the full range of operation. + * FDCO = freq_out * 2 * mclk_src_scaling + */ + for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { + fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + if (NAU_FVCO_MIN < fvco && fvco < NAU_FVCO_MAX) + break; + } + if (i == ARRAY_SIZE(mclk_src_scaling)) + return -EINVAL; + fll_param->mclk_src = mclk_src_scaling[i].val; + + /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional + * input based on FDCO, FREF and FLL ratio. + */ + fll_input = (fvco << 16) / (fref * fll_param->ratio); + fll_param->fll_int = fll_input >> 16; + fll_param->fll_frac = fll_input & 0xFFFF; + return 0; +} + +static void nau88xx_fll_apply(struct nau8825 *nau8825, + struct nau88xx_fll *fll_param) +{ + regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_MCLK_SRC_MASK, fll_param->mclk_src); + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL1, + NAU8825_FLL_RATIO_MASK, fll_param->ratio); + /* FLL 16-bit fractional input */ + regmap_write(nau8825->regmap, NAU8825_REG_FLL2, fll_param->fll_frac); + /* FLL 10-bit integer input */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL3, + NAU8825_FLL_INTEGER_MASK, fll_param->fll_int); + /* FLL pre-scaler */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL4, + NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); + /* select divided VCO input */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, + NAU8825_FLL_FILTER_SW_MASK, 0x0000); + /* FLL sigma delta modulator enable */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN_MASK, NAU8825_SDM_EN); +} + +/* freq_out must be 256*Fs in order to achieve the best performance */ +static int nau8825_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct nau88xx_fll fll_param; + int ret, fs; + + fs = freq_out / 256; + ret = nau88xx_calc_fll_param(freq_in, fs, &fll_param); + if (ret < 0) { + dev_err(codec->dev, "Unsupported input clock %d\n", freq_in); + return ret; + } + dev_dbg(codec->dev, "mclk_src=%x ratio=%x fll_frac=%x fll_int=%x clk_ref_div=%x\n", + fll_param.mclk_src, fll_param.ratio, fll_param.fll_frac, + fll_param.fll_int, fll_param.clk_ref_div); + + nau88xx_fll_apply(nau8825, &fll_param); + mdelay(2); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + return 0; +} + static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, unsigned int freq) { @@ -920,6 +1081,7 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, static struct snd_soc_codec_driver nau8825_codec_driver = { .probe = nau8825_codec_probe, .set_sysclk = nau8825_set_sysclk, + .set_pll = nau8825_set_pll, .set_bias_level = nau8825_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 8774923..dff8edb 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -101,13 +101,31 @@ #define NAU8825_ENABLE_SAR_SFT 1 /* CLK_DIVIDER (0x3) */ -#define NAU8825_CLK_SRC_SFT 15 -#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT) -#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT) -#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_SFT 15 +#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_MCLK_SRC_MASK (0xf << 0) + +/* FLL1 (0x04) */ +#define NAU8825_FLL_RATIO_MASK (0x7f << 0) + +/* FLL3 (0x06) */ +#define NAU8825_FLL_INTEGER_MASK (0x3ff << 0) + +/* FLL4 (0x07) */ +#define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) + +/* FLL5 (0x08) */ +#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14) /* FLL6 (0x9) */ -#define NAU8825_DCO_EN (1 << 15) +#define NAU8825_DCO_EN_MASK (0x1 << 15) +#define NAU8825_DCO_EN (0x1 << 15) +#define NAU8825_DCO_DIS (0x0 << 15) +#define NAU8825_SDM_EN_MASK (0x1 << 14) +#define NAU8825_SDM_EN (0x1 << 14) +#define NAU8825_SDM_DIS (0x0 << 14) /* HSD_CTRL (0xc) */ #define NAU8825_HSD_AUTO_MODE (1 << 6) -- 2.6.0.rc2.230.g3dd15c0

3 7

[alsa-devel] [PATCH] ASoC: Document DAI signal polarity
by Anatol Pomozov 23 Oct '15

23 Oct '15

Currently there is no clear definition of what FSYNC polarity is. Different drivers use its own definition of what is "normal" and what is "inverted" fsync. This leads to compatibility problems between drivers. For example TegraX1 driver assumes that DSP-A format with frames starting at rising FSYNC edge has "inverted" polarity, while RT5677 assumes it is "normal" polarity. Explicitly specify meaning of BCLK/FSYNC polarity to avoid future compatibility problems. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- include/sound/soc-dai.h | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1..91e2e61 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -48,10 +48,25 @@ struct snd_compr_stream; #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ /* - * DAI hardware signal inversions. + * DAI hardware signal polarity. * * Specifies whether the DAI can also support inverted clocks for the specified * format. + * + * BCLK: + * - "normal" polarity means signal is available at rising edge of BCLK + * - "inverted" polarity means signal is available at falling edge of BCLK + * + * FSYNC "normal" polarity depends on the frame format: + * - I2S: frame consists of left then right channel data. Left channel starts + * with falling FSYNC edge, right channel starts with rising FSYNC edge. + * - Left/Right Justified: frame consists of left then right channel data. + * Left channel starts with rising FSYNC edge, right channel starts with + * falling FSYNC edge. + * - DSP A/B: Frame starts with rising FSYNC edge. + * - AC97: Frame starts with rising FSYNC edge. + * + * "Negative" FSYNC polarity is the one opposite of "normal" polarity. */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ -- 2.6.0.rc2.230.g3dd15c0

3 2

[alsa-devel] Applied "ASoC: fsl-asoc-card: add AC'97 support" to the asoc tree
by Mark Brown 23 Oct '15

23 Oct '15

The patch ASoC: fsl-asoc-card: add AC'97 support has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 50760cad9de969fe85b24465afe6396b8bbc6a3f Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" <mail(a)maciej.szmigiero.name> Date: Sat, 19 Sep 2015 02:00:25 +0200 Subject: [PATCH] ASoC: fsl-asoc-card: add AC'97 support Add AC'97 support to fsl-asoc-card using generic ASoC AC'97 CODEC. The SSI controller will silently enable any TX AC'97 slots that have their bits set in SLOTREQ received from CODEC and then will redirect some of playback samples there. That's why it is important to make sure that any of CODEC playback slots that can pull samples are set to slots 3/4 (standard PCM playback slots). Currently, this applies to S/PDIF slots as they were seen to pull samples sometimes even with S/PDIF output being disabled. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/fsl-asoc-card.txt | 10 +- sound/soc/fsl/fsl-asoc-card.c | 140 ++++++++++++++++----- 2 files changed, 116 insertions(+), 34 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index a96774c..ce55c0a 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -13,13 +13,15 @@ So having this generic sound card allows all Freescale SoC users to benefit from the simplification of a new card support and the capability of the wide sample rates support through ASRC. -Note: The card is initially designed for those sound cards who use I2S and - PCM DAI formats. However, it'll be also possible to support those non - I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long - as the driver has been properly upgraded. +Note: The card is initially designed for those sound cards who use AC'97, I2S + and PCM DAI formats. However, it'll be also possible to support those non + AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as + long as the driver has been properly upgraded. The compatible list for this generic sound card currently: + "fsl,imx-audio-ac97" + "fsl,imx-audio-cs42888" "fsl,imx-audio-wm8962" diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5aeb6ed..33628a0 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -14,6 +14,9 @@ #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_platform.h> +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) +#include <sound/ac97_codec.h> +#endif #include <sound/pcm_params.h> #include <sound/soc.h> @@ -115,6 +118,11 @@ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; +static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv) +{ + return priv->dai_fmt == SND_SOC_DAIFMT_AC97; +} + static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -133,7 +141,9 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, * set_bias_level(), bypass the remaining settings in hw_params(). * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. */ - if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) + if ((priv->card.set_bias_level && + priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || + fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -300,7 +310,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, ext_port--; /* - * Use asynchronous mode (6 wires) for all cases. + * Use asynchronous mode (6 wires) for all cases except AC97. * If only 4 wires are needed, just set SSI into * synchronous mode and enable 4 PADs in IOMUX. */ @@ -346,15 +356,30 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, IMX_AUDMUX_V2_PTCR_TCLKDIR; break; default: - return -EINVAL; + if (!fsl_asoc_card_is_ac97(priv)) + return -EINVAL; + } + + if (fsl_asoc_card_is_ac97(priv)) { + int_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR; } /* Asynchronous mode can not be set along with RCLKDIR */ - ret = imx_audmux_v2_configure_port(int_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(dev, "audmux internal port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port); + + ret = imx_audmux_v2_configure_port(int_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(int_port, int_ptcr, @@ -364,11 +389,16 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, return ret; } - ret = imx_audmux_v2_configure_port(ext_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(dev, "audmux external port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port); + + ret = imx_audmux_v2_configure_port(ext_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, @@ -389,6 +419,23 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + if (fsl_asoc_card_is_ac97(priv)) { +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + /* + * Use slots 3/4 for S/PDIF so SSI won't try to enable + * other slots and send some samples there + * due to SLOTREQ bits for S/PDIF received from codec + */ + snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, + AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4); +#endif + + return 0; + } + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { @@ -407,7 +454,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; struct i2c_client *codec_dev; - struct clk *codec_clk; const char *codec_dai_name; u32 width; int ret; @@ -420,9 +466,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Give a chance to old DT binding */ if (!cpu_np) cpu_np = of_parse_phandle(np, "ssi-controller", 0); - codec_np = of_parse_phandle(np, "audio-codec", 0); - if (!cpu_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); + if (!cpu_np) { + dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); ret = -EINVAL; goto fail; } @@ -434,22 +479,24 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto fail; } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; - } + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (codec_np) + codec_dev = of_find_i2c_device_by_node(codec_np); + else + codec_dev = NULL; asrc_np = of_parse_phandle(np, "audio-asrc", 0); if (asrc_np) asrc_pdev = of_find_device_by_node(asrc_np); /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ - codec_clk = clk_get(&codec_dev->dev, NULL); - if (!IS_ERR(codec_clk)) { - priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); - clk_put(codec_clk); + if (codec_dev) { + struct clk *codec_clk = clk_get(&codec_dev->dev, NULL); + + if (!IS_ERR(codec_clk)) { + priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } } /* Default sample rate and format, will be updated in hw_params() */ @@ -486,11 +533,21 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { + codec_dai_name = "ac97-hifi"; + priv->card.set_bias_level = NULL; + priv->dai_fmt = SND_SOC_DAIFMT_AC97; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); return -EINVAL; } + if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { + dev_err(&pdev->dev, "failed to find codec device\n"); + ret = -EINVAL; + goto asrc_fail; + } + /* Common settings for corresponding Freescale CPU DAI driver */ if (strstr(cpu_np->name, "ssi")) { /* Only SSI needs to configure AUDMUX */ @@ -507,7 +564,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } - sprintf(priv->name, "%s-audio", codec_dev->name); + snprintf(priv->name, sizeof(priv->name), "%s-audio", + fsl_asoc_card_is_ac97(priv) ? "ac97" : + codec_dev->name); /* Initialize sound card */ priv->pdev = pdev; @@ -531,8 +590,26 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; - priv->dai_link[0].codec_of_node = codec_np; priv->dai_link[0].codec_dai_name = codec_dai_name; + + if (!fsl_asoc_card_is_ac97(priv)) + priv->dai_link[0].codec_of_node = codec_np; + else { + u32 idx; + + ret = of_property_read_u32(cpu_np, "cell-index", &idx); + if (ret) { + dev_err(&pdev->dev, + "cannot get CPU index property\n"); + goto asrc_fail; + } + + priv->dai_link[0].codec_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, + "ac97-codec.%u", + (unsigned int)idx); + } + priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -543,6 +620,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_link[1].platform_of_node = asrc_np; priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].codec_name = + priv->dai_link[0].codec_name; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->card.num_links = 3; @@ -578,14 +657,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); -fail: of_node_put(codec_np); +fail: of_node_put(cpu_np); return ret; } static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, -- 2.6.1

1 0

[alsa-devel] [PATCH] ASoC: rt5645: Add dmi "Google Reks" for chrome
by John Lin 23 Oct '15

23 Oct '15

Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index b0d96b6..84c5261 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3212,6 +3212,13 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"), }, }, + { + .ident = "Google Reks", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Reks"), + }, + }, { } }; -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [RESEND PATCH v3 0/2] ASoC: atmel-classd: add driver for Atmel CLASSD
by Songjun Wu 23 Oct '15

23 Oct '15

The Audio Class D Amplifier driver includes two parts. 1) Driver code to implement the Audio Class D Amplifier function. 2) Device tree binding document, it describes how to add the Audio Class D Amplifier in device tree. Changes in v3: - Remove '.owner' field in variable 'atmel_classd_driver', since the core will do it. - Correct 'IS_ERR(dd->aclk)' to 'IS_ERR(dd->gclk)', when geting the clock 'dd->gclk'. Changes in v2: - Change the "Mono", "Swap" and "Deemphasis" controls to "Mono Switch", "Swap Switch" and "Deemphasis Switch". - Merge "EQ Bass", "EQ Medium" and "EQ Treble" controls into one "EQ" control. - Change the "single-ended" and "differential" to "Single ended", "Differential". - Merge separate left and right controls into one single stereo control. - Add a warning if the user trys to specify an invalid value in function "atmel_classd_codec_probe". - Use the "dev_get_regmap()" in function "atmel_classd_codec_get_remap". - Check for errors when invoking the "clk_prepare_enable". - Remove the "dev_info" in function "atmel_classd_probe". - Add some code to create a sound card in function "atmel_classd_probe". - Remove the DT node "Sound" and the related code. - Delete the device node "Sound" in dts. Songjun Wu (2): ASoC: atmel-classd: add the Audio Class D Amplifier ASoC: atmel-classd: DT binding for Class D audio amplifier driver .../devicetree/bindings/sound/atmel-classd.txt | 52 ++ sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-classd.c | 679 ++++++++++++++++++++ sound/soc/atmel/atmel-classd.h | 120 ++++ 5 files changed, 862 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt create mode 100644 sound/soc/atmel/atmel-classd.c create mode 100644 sound/soc/atmel/atmel-classd.h -- 1.7.9.5

2 2

[alsa-devel] Applied "ASoC: atmel-classd: DT binding for Class D audio amplifier driver" to the asoc tree
by Mark Brown 23 Oct '15

23 Oct '15

The patch ASoC: atmel-classd: DT binding for Class D audio amplifier driver has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 391ac3ef50b9ec575c4d5c6f055efe5096ac1957 Mon Sep 17 00:00:00 2001 From: Songjun Wu <songjun.wu(a)atmel.com> Date: Thu, 8 Oct 2015 18:13:32 +0800 Subject: [PATCH] ASoC: atmel-classd: DT binding for Class D audio amplifier driver DT binding documentation for this new ASoC driver. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/atmel-classd.txt | 52 ++++++++++++++++++++++ 1 file changed, 52 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt diff --git a/Documentation/devicetree/bindings/sound/atmel-classd.txt b/Documentation/devicetree/bindings/sound/atmel-classd.txt new file mode 100644 index 0000000..0018451 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel-classd.txt @@ -0,0 +1,52 @@ +* Atmel ClassD driver under ALSA SoC architecture + +Required properties: +- compatible + Should be "atmel,sama5d2-classd". +- reg + Should contain ClassD registers location and length. +- interrupts + Should contain the IRQ line for the ClassD. +- dmas + One DMA specifiers as described in atmel-dma.txt and dma.txt files. +- dma-names + Must be "tx". +- clock-names + Tuple listing input clock names. + Required elements: "pclk", "gclk" and "aclk". +- clocks + Please refer to clock-bindings.txt. + +Optional properties: +- pinctrl-names, pinctrl-0 + Please refer to pinctrl-bindings.txt. +- atmel,model + The user-visible name of this sound complex. + The default value is "CLASSD". +- atmel,pwm-type + PWM modulation type, "single" or "diff". + The default value is "single". +- atmel,non-overlap-time + Set non-overlapping time, the unit is nanosecond(ns). + There are four values, + <5>, <10>, <15>, <20>, the default value is <10>. + Non-overlapping will be disabled if not specified. + +Example: +classd: classd@fc048000 { + compatible = "atmel,sama5d2-classd"; + reg = <0xfc048000 0x100>; + interrupts = <59 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) + | AT91_XDMAC_DT_PERID(47))>; + dma-names = "tx"; + clocks = <&classd_clk>, <&classd_gclk>, <&audio_pll_pmc>; + clock-names = "pclk", "gclk", "aclk"; + + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_classd_default>; + atmel,model = "classd @ SAMA5D2-Xplained"; + atmel,pwm-type = "diff"; + atmel,non-overlap-time = <10>; +}; -- 2.6.1

1 0

[alsa-devel] [PATCH v2] ASoC: Intel: Skylake: Convert to devm_snd_soc_register_card
by Axel Lin 23 Oct '15

23 Oct '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/intel/boards/skl_rt286.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 488c98e..a73a431 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -240,18 +240,11 @@ static int skylake_audio_probe(struct platform_device *pdev) { skylake_rt286.dev = &pdev->dev; - return snd_soc_register_card(&skylake_rt286); -} - -static int skylake_audio_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&skylake_rt286); - return 0; + return devm_snd_soc_register_card(&pdev->dev, &skylake_rt286); } static struct platform_driver skylake_audio = { .probe = skylake_audio_probe, - .remove = skylake_audio_remove, .driver = { .name = "skl_alc286s_i2s", }, -- 2.1.4

2 1