November 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: omap-mcbsp: On OMAP1 and OMAP2420 only 16bit format is supported
by Peter Ujfalusi 05 Nov '18

05 Nov '18

For samples sizes of 20/24/32 bit the DRR2/DXR2 registers must be used to receive or transmit the most significant part of the 32bit sample. We can not trick the system by using the DRR2/DXR2 register offset and use 32bit element size since these (and other McBSP) registers are 16bit ones. We would need support for port_window in cyclic mode from the sDMA driver, but it is not straight forward as we would need to have the sDMA frame to cover the 32bit (DRR2+DRR1/DXR2+DXR1), but the frames must cover the ALSA period to be able to receive periodic interrupts. Since the 32bit samples are not working (DMA timeout), just remove it in case we have McBSP with reg_size of 2. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, I have noticed that on Nokia n810 32bit samples just don't work at all for audio Regards, Peter sound/soc/omap/omap-mcbsp.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d0ebb6b9bfac..caa24c463b3e 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -862,6 +862,11 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) if (ret) return ret; + if (mcbsp->pdata->reg_size == 2) { + omap_mcbsp_dai.playback.formats = SNDRV_PCM_FMTBIT_S16_LE; + omap_mcbsp_dai.capture.formats = SNDRV_PCM_FMTBIT_S16_LE; + } + ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, &omap_mcbsp_dai, 1); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] ASoC: pcm3168a: cleanup hw_constraint for LEFT_J/I2S
by Kuninori Morimoto 05 Nov '18

05 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> hw constraint for LEFT_J/I2S are same, but had duplicated code. This code cleanup these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/pcm3168a.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index e1c3a18..115f2165 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -497,6 +497,10 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, unsigned int fmt; unsigned int sample_min; unsigned int channel_max; + unsigned int channel_maxs[] = { + 6, /* rx */ + 8 /* tx */ + }; if (tx) fmt = pcm3168a->dac_fmt; @@ -521,18 +525,9 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, channel_max = 2; break; case PCM3168A_FMT_LEFT_J: - sample_min = 24; - if (tx) - channel_max = 8; - else - channel_max = 6; - break; case PCM3168A_FMT_I2S: sample_min = 24; - if (tx) - channel_max = 8; - else - channel_max = 6; + channel_max = channel_maxs[tx]; break; default: sample_min = 24; -- 2.7.4

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[alsa-devel] [PATCH 0/3] ASoC: rsnd: prepare patches for BUSIF support
by Kuninori Morimoto 05 Nov '18

05 Nov '18

Hi Mark These are prepare/cleanup for for Renesas rsnd driver BUSIF support. I will post BUSIF support patch soon, but it needs Jiada Wang <jiada_wang(a)mentor.com> 's check. If it was OK, I will focus to TDM split mode. Kuninori Morimoto (3): ASoC: rsnd: move .get_status under rsnd_mod_ops ASoC: rsnd: add .get_id/.get_id_sub ASoC: rsnd: rsnd_mod_name() handles both name and ID sound/soc/sh/rcar/adg.c | 2 +- sound/soc/sh/rcar/cmd.c | 11 ++-- sound/soc/sh/rcar/core.c | 72 +++++++++++++++++------ sound/soc/sh/rcar/ctu.c | 23 +++++++- sound/soc/sh/rcar/dma.c | 47 +++++++-------- sound/soc/sh/rcar/dvc.c | 3 +- sound/soc/sh/rcar/gen.c | 12 ++-- sound/soc/sh/rcar/mix.c | 3 +- sound/soc/sh/rcar/rsnd.h | 21 +++---- sound/soc/sh/rcar/src.c | 29 +++++----- sound/soc/sh/rcar/ssi.c | 147 +++++++++++++++++++++++------------------------ sound/soc/sh/rcar/ssiu.c | 37 ++++++------ 12 files changed, 232 insertions(+), 175 deletions(-) -- 2.7.4 Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0
by Hans de Goede 05 Nov '18

05 Nov '18

Some boards such as the Swanky model Chromebooks use pmc_plt_clk_0 for the mclk instead of pmc_plt_clk_3. This commit adds a DMI based quirk for this. This fixing audio no longer working on these devices after commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") that commit fixes us unnecessary keeping unused clocks on, but in case of the Swanky that was breaking audio support since we were not using the right clock in the cht_bsw_max98090_ti machine driver. Cc: stable(a)vger.kernel.org Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Reported-and-tested-by: Dean Wallace <duffydack73(a)gmail.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 32 ++++++++++++++++++-- 1 file changed, 29 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index db6976f4ddaa..9d9f6e41d81c 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -19,6 +19,7 @@ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ +#include <linux/dmi.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> @@ -35,6 +36,8 @@ #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI "HiFi" +#define QUIRK_PMC_PLT_CLK_0 0x01 + struct cht_mc_private { struct clk *mclk; struct snd_soc_jack jack; @@ -385,11 +388,29 @@ static struct snd_soc_card snd_soc_card_cht = { .num_controls = ARRAY_SIZE(cht_mc_controls), }; +static const struct dmi_system_id cht_max98090_quirk_table[] = { + { + /* Swanky model Chromebook (Toshiba Chromebook 2) */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Swanky"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + {} +}; + static int snd_cht_mc_probe(struct platform_device *pdev) { + const struct dmi_system_id *dmi_id; struct device *dev = &pdev->dev; int ret_val = 0; struct cht_mc_private *drv; + const char *mclk_name; + int quirks = 0; + + dmi_id = dmi_first_match(cht_max98090_quirk_table); + if (dmi_id) + quirks = (unsigned long)dmi_id->driver_data; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) @@ -411,11 +432,16 @@ static int snd_cht_mc_probe(struct platform_device *pdev) snd_soc_card_cht.dev = &pdev->dev; snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); - drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (quirks & QUIRK_PMC_PLT_CLK_0) + mclk_name = "pmc_plt_clk_0"; + else + mclk_name = "pmc_plt_clk_3"; + + drv->mclk = devm_clk_get(&pdev->dev, mclk_name); if (IS_ERR(drv->mclk)) { dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(drv->mclk)); + "Failed to get MCLK from %s: %ld\n", + mclk_name, PTR_ERR(drv->mclk)); return PTR_ERR(drv->mclk); } -- 2.19.0

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[alsa-devel] [PATCH] ASoC: qdsp6: q6afe: Fix wrong MI2S SD line mask
by Rohit kumar 05 Nov '18

05 Nov '18

SD line mask for MI2S starts from BIT 0 instead of BIT 1. Fix all bit mask for MI2S SD lines. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/qcom/qdsp6/q6afe.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 000775b..829b5e987 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -49,14 +49,14 @@ #define AFE_PORT_I2S_SD1 0x2 #define AFE_PORT_I2S_SD2 0x3 #define AFE_PORT_I2S_SD3 0x4 -#define AFE_PORT_I2S_SD0_MASK BIT(0x1) -#define AFE_PORT_I2S_SD1_MASK BIT(0x2) -#define AFE_PORT_I2S_SD2_MASK BIT(0x3) -#define AFE_PORT_I2S_SD3_MASK BIT(0x4) -#define AFE_PORT_I2S_SD0_1_MASK GENMASK(2, 1) -#define AFE_PORT_I2S_SD2_3_MASK GENMASK(4, 3) -#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(3, 1) -#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(4, 1) +#define AFE_PORT_I2S_SD0_MASK BIT(0x0) +#define AFE_PORT_I2S_SD1_MASK BIT(0x1) +#define AFE_PORT_I2S_SD2_MASK BIT(0x2) +#define AFE_PORT_I2S_SD3_MASK BIT(0x3) +#define AFE_PORT_I2S_SD0_1_MASK GENMASK(1, 0) +#define AFE_PORT_I2S_SD2_3_MASK GENMASK(3, 2) +#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(2, 0) +#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(3, 0) #define AFE_PORT_I2S_QUAD01 0x5 #define AFE_PORT_I2S_QUAD23 0x6 #define AFE_PORT_I2S_6CHS 0x7 -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[alsa-devel] [PATCH] ALSA: usb-audio: Define registers for CM6206
by Linus Walleij 05 Nov '18

05 Nov '18

The register map for CM6206 is known and we can define what the values written to the different registers actually mean. I tested to print the hex values before/after this change, there is no functional change. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- sound/usb/quirks.c | 129 +++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 124 insertions(+), 5 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 8a945ece9869..3d0f09108c98 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -19,6 +19,7 @@ #include <linux/usb.h> #include <linux/usb/audio.h> #include <linux/usb/midi.h> +#include <linux/bits.h> #include <sound/control.h> #include <sound/core.h> @@ -668,15 +669,133 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev) } /* - * C-Media CM6206 is based on CM106 with two additional - * registers that are not documented in the data sheet. - * Values here are chosen based on sniffing USB traffic - * under Windows. + * CM6206 registers from the CM6206 datasheet rev 2.1 */ +#define CM6206_REG0_DMA_MASTER BIT(15) +#define CM6206_REG0_SPDIFO_RATE_48K (2 << 12) +#define CM6206_REG0_SPDIFO_RATE_96K (7 << 12) +/* Bit 4 thru 11 is the S/PDIF category code */ +#define CM6206_REG0_SPDIFO_CAT_CODE_GENERAL (0 << 4) +#define CM6206_REG0_SPDIFO_EMPHASIS_CD BIT(3) +#define CM6206_REG0_SPDIFO_COPYRIGHT_NA BIT(2) +#define CM6206_REG0_SPDIFO_NON_AUDIO BIT(1) +#define CM6206_REG0_SPDIFO_PRO_FORMAT BIT(0) + +#define CM6206_REG1_TEST_SEL_CLK BIT(14) +#define CM6206_REG1_PLLBIN_EN BIT(13) +#define CM6206_REG1_SOFT_MUTE_EN BIT(12) +#define CM6206_REG1_GPIO4_OUT BIT(11) +#define CM6206_REG1_GPIO4_OE BIT(10) +#define CM6206_REG1_GPIO3_OUT BIT(9) +#define CM6206_REG1_GPIO3_OE BIT(8) +#define CM6206_REG1_GPIO2_OUT BIT(7) +#define CM6206_REG1_GPIO2_OE BIT(6) +#define CM6206_REG1_GPIO1_OUT BIT(5) +#define CM6206_REG1_GPIO1_OE BIT(4) +#define CM6206_REG1_SPDIFO_INVALID BIT(3) +#define CM6206_REG1_SPDIF_LOOP_EN BIT(2) +#define CM6206_REG1_SPDIFO_DIS BIT(1) +#define CM6206_REG1_SPDIFI_MIX BIT(0) + +#define CM6206_REG2_DRIVER_ON BIT(15) +#define CM6206_REG2_HEADP_SEL_SIDE_CHANNELS (0 << 13) +#define CM6206_REG2_HEADP_SEL_SURROUND_CHANNELS (1 << 13) +#define CM6206_REG2_HEADP_SEL_CENTER_SUBW (2 << 13) +#define CM6206_REG2_HEADP_SEL_FRONT_CHANNELS (3 << 13) +#define CM6206_REG2_MUTE_HEADPHONE_RIGHT BIT(12) +#define CM6206_REG2_MUTE_HEADPHONE_LEFT BIT(11) +#define CM6206_REG2_MUTE_REAR_SURROUND_RIGHT BIT(10) +#define CM6206_REG2_MUTE_REAR_SURROUND_LEFT BIT(9) +#define CM6206_REG2_MUTE_SIDE_SURROUND_RIGHT BIT(8) +#define CM6206_REG2_MUTE_SIDE_SURROUND_LEFT BIT(7) +#define CM6206_REG2_MUTE_SUBWOOFER BIT(6) +#define CM6206_REG2_MUTE_CENTER BIT(5) +#define CM6206_REG2_MUTE_RIGHT_FRONT BIT(3) +#define CM6206_REG2_MUTE_LEFT_FRONT BIT(3) +#define CM6206_REG2_EN_BTL BIT(2) +#define CM6206_REG2_MCUCLKSEL_1_5_MHZ (0) +#define CM6206_REG2_MCUCLKSEL_3_MHZ (1) +#define CM6206_REG2_MCUCLKSEL_6_MHZ (2) +#define CM6206_REG2_MCUCLKSEL_12_MHZ (3) + +/* Bit 11..13 sets the sensitivity to FLY tuner volume control VP/VD signal */ +#define CM6206_REG3_FLYSPEED_DEFAULT (2 << 11) +#define CM6206_REG3_VRAP25EN BIT(10) +#define CM6206_REG3_MSEL1 BIT(9) +#define CM6206_REG3_SPDIFI_RATE_44_1K BIT(0 << 7) +#define CM6206_REG3_SPDIFI_RATE_48K BIT(2 << 7) +#define CM6206_REG3_SPDIFI_RATE_32K BIT(3 << 7) +#define CM6206_REG3_PINSEL BIT(6) +#define CM6206_REG3_FOE BIT(5) +#define CM6206_REG3_ROE BIT(4) +#define CM6206_REG3_CBOE BIT(3) +#define CM6206_REG3_LOSE BIT(2) +#define CM6206_REG3_HPOE BIT(1) +#define CM6206_REG3_SPDIFI_CANREC BIT(0) + +#define CM6206_REG5_DA_RSTN BIT(13) +#define CM6206_REG5_AD_RSTN BIT(12) +#define CM6206_REG5_SPDIFO_AD2SPDO BIT(12) +#define CM6206_REG5_SPDIFO_SEL_FRONT (0 << 9) +#define CM6206_REG5_SPDIFO_SEL_SIDE_SUR (1 << 9) +#define CM6206_REG5_SPDIFO_SEL_CEN_LFE (2 << 9) +#define CM6206_REG5_SPDIFO_SEL_REAR_SUR (3 << 9) +#define CM6206_REG5_CODECM BIT(8) +#define CM6206_REG5_EN_HPF BIT(7) +#define CM6206_REG5_T_SEL_DSDA4 BIT(6) +#define CM6206_REG5_T_SEL_DSDA3 BIT(5) +#define CM6206_REG5_T_SEL_DSDA2 BIT(4) +#define CM6206_REG5_T_SEL_DSDA1 BIT(3) +#define CM6206_REG5_T_SEL_DSDAD_NORMAL 0 +#define CM6206_REG5_T_SEL_DSDAD_FRONT 4 +#define CM6206_REG5_T_SEL_DSDAD_S_SURROUND 5 +#define CM6206_REG5_T_SEL_DSDAD_CEN_LFE 6 +#define CM6206_REG5_T_SEL_DSDAD_R_SURROUND 7 + static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) { int err = 0, reg; - int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000}; + int val[] = { + /* + * Values here are chosen based on sniffing USB traffic + * under Windows. + * + * REG0: DAC is master, sample rate 48kHz, no copyright + */ + CM6206_REG0_SPDIFO_RATE_48K | + CM6206_REG0_SPDIFO_COPYRIGHT_NA, + /* + * REG1: PLL binary search enable, soft mute enable. + */ + CM6206_REG1_PLLBIN_EN | + CM6206_REG1_SOFT_MUTE_EN | + /* + * REG2: enable output drivers, + * select front channels to the headphone output, + * then mute the headphone channels, run the MCU + * at 1.5 MHz. + */ + CM6206_REG2_DRIVER_ON | + CM6206_REG2_HEADP_SEL_FRONT_CHANNELS | + CM6206_REG2_MUTE_HEADPHONE_RIGHT | + CM6206_REG2_MUTE_HEADPHONE_LEFT, + /* + * REG3: default flyspeed, set 2.5V mic bias + * enable all line out ports and enable SPDIF + */ + CM6206_REG3_FLYSPEED_DEFAULT | + CM6206_REG3_VRAP25EN | + CM6206_REG3_FOE | + CM6206_REG3_ROE | + CM6206_REG3_CBOE | + CM6206_REG3_LOSE | + CM6206_REG3_HPOE | + CM6206_REG3_SPDIFI_CANREC, + /* REG4 is just a bunch of GPIO lines */ + 0x0000, + /* REG5: de-assert AD/DA reset signals */ + CM6206_REG5_DA_RSTN | + CM6206_REG5_AD_RSTN }; for (reg = 0; reg < ARRAY_SIZE(val); reg++) { err = snd_usb_cm106_write_int_reg(dev, reg, val[reg]); -- 2.19.1

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Re: [alsa-devel] [PATCH] ASoC: wm8996: fix small typo
by Mark Brown 05 Nov '18

05 Nov '18

On Fri, Nov 02, 2018 at 09:30:06AM +0800, sh liu wrote: > From e9b923690675ca8fa883fd25dcead5b457856735 Mon Sep 17 00:00:00 2001 > From: liush <liush(a)allwinnertech.com> > Date: Fri, 2 Nov 2018 08:57:00 +0800 > Subject: [PATCH] ASoC: wm8996: fix small typo > > atleast -> at least > > Change-Id: Icc970b438166daef13518b7d1a62b13eb8752f5f > Signed-off-by: liush <liush(a)allwinnertech.com> This patch is corrupted and can't be applied: > *codec, int fll_id, int source, > if (i2c->irq) > timeout *= 10; > else > - /* ensure timeout of atleast 1 jiffies */ > + /* ensure timeout of at least 1 jiffies */ > timeout = timeout/2 ? : 1; > It looks like tabs have been replaced by spaces in whatever you're using to send it.

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[alsa-devel] [PATCH] SoC: nau8825: remove unnecessary unlikely()
by Yangtao Li 04 Nov '18

04 Nov '18

WARN_ON() already contains an unlikely(), so it's not necessary to use unlikely. Signed-off-by: Yangtao Li <tiny.windzz(a)gmail.com> --- sound/soc/codecs/nau8825.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index b9fed99d8b5e..7bbcbf5f05c8 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -424,10 +424,8 @@ static u32 nau8825_xtalk_sidetone(u32 sig_org, u32 sig_cros) { u32 gain, sidetone; - if (unlikely(sig_org == 0) || unlikely(sig_cros == 0)) { - WARN_ON(1); + if (WARN_ON(sig_org == 0 || sig_cros == 0)) return 0; - } sig_org = nau8825_intlog10_dec3(sig_org); sig_cros = nau8825_intlog10_dec3(sig_cros); -- 2.17.0

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[alsa-devel] Trying to get Roland TD-30 soundcard to work on Linux
by Cupertino Miranda 02 Nov '18

02 Nov '18

Hi, I have the Rolands TD-30 edrum module that includes a USB sound card in it. The card lists but no capture or recording devices work, although they are listed. I am new to audio device drivers and for that reason need some guidance to contribute to such driver. A full log can be found at: www.alsa-project.org/db/?f=e1a28cbd7e9f1201d3b0cbe44d8ec1908272a1d4 Looking forward to ear from you. Best regards, Cupertino Miranda

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[alsa-devel] [PATCH 1/2] ALSA: x86: Fix runtime PM for hdmi-lpe-audio
by Ville Syrjala 02 Nov '18

02 Nov '18

From: Ville Syrjälä <ville.syrjala(a)linux.intel.com> Commit 46e831abe864 ("drm/i915/lpe: Mark LPE audio runtime pm as "no callbacks"") broke runtime PM with lpe audio. We can no longer runtime suspend the GPU since the sysfs power/control for the lpe-audio device no longer exists and the device is considered always active. We can fix this by not marking the device as active. Cc: Chris Wilson <chris(a)chris-wilson.co.uk> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Fixes: 46e831abe864 ("drm/i915/lpe: Mark LPE audio runtime pm as "no callbacks"") Signed-off-by: Ville Syrjälä <ville.syrjala(a)linux.intel.com> --- sound/x86/intel_hdmi_audio.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 83d76c345940..bbed4acaafd1 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1877,7 +1877,6 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) pm_runtime_use_autosuspend(&pdev->dev); pm_runtime_mark_last_busy(&pdev->dev); - pm_runtime_set_active(&pdev->dev); dev_dbg(&pdev->dev, "%s: handle pending notification\n", __func__); for_each_port(card_ctx, port) { -- 2.18.1

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