July 2020 - Alsa-devel - mailman.alsa-project.org

headphones detected but don't show up in alsa mixer, no audio on ubuntu 18.04
by GitHub issues - edited 27 Jul '20

27 Jul '20

alsa-project/alsa-lib issue #75 was edited from rbavery: Here is my ALSA info: http://alsa-project.org/db/?f=dd3d1d8a9fa0020900a0adc03bc6adfaf1f77ee5 I tried following these remedies (including reinstalling alsa, reloading, restarting computer) but nothing has panned out: https://support.system76.com/articles/audio/ Any help or suggestions are appreciated Issue URL : https://github.com/alsa-project/alsa-lib/issues/75 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: hda: Workaround for spurious wakeups on some Intel platforms
by Takashi Iwai 27 Jul '20

27 Jul '20

We've received a regression report on Intel HD-audio controller that wakes up immediately after S3 suspend. The bisection leads to the commit c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed"). This commit replaces the system-suspend to use pm_runtime_force_suspend() instead of the direct call of __azx_runtime_suspend(). However, by some really mysterious reason, pm_runtime_force_suspend() causes a spurious wakeup (although it calls the same __azx_runtime_suspend() internally). As an ugly workaround for now, revert the behavior to call __azx_runtime_suspend() and __azx_runtime_resume() for those old Intel platforms that may exhibit such a problem, while keeping the new standard pm_runtime_force_suspend() and pm_runtime_force_resume() pair for the remaining chips. Fixes: c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208649 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 17 ++++++++++++++--- 2 files changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index fe171685492d..be63ead8161f 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -41,7 +41,7 @@ /* 24 unused */ #define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ -/* 27 unused */ +#define AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP (1 << 27) /* Workaround for spurious wakeups after suspend */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ #define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9ba1fb8f0b7f..fb65450d8de1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -297,7 +297,8 @@ enum { /* PCH for HSW/BDW; with runtime PM */ /* no i915 binding for this as HSW/BDW has another controller for HDMI */ #define AZX_DCAPS_INTEL_PCH \ - (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME) + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ + AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) /* HSW HDMI */ #define AZX_DCAPS_INTEL_HASWELL \ @@ -1026,7 +1027,14 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - pm_runtime_force_suspend(dev); + /* An ugly workaround: direct call of __azx_runtime_suspend() and + * __azx_runtime_resume() for old Intel platforms that suffer from + * spurious wakeups after S3 suspend + */ + if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) + __azx_runtime_suspend(chip); + else + pm_runtime_force_suspend(dev); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1055,7 +1063,10 @@ static int azx_resume(struct device *dev) if (azx_acquire_irq(chip, 1) < 0) return -EIO; - pm_runtime_force_resume(dev); + if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) + __azx_runtime_resume(chip, false); + else + pm_runtime_force_resume(dev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); -- 2.26.2

2 2

Lenovo X270 with HDA ALC298
by Yasushi SHOJI 27 Jul '20

27 Jul '20

Hi all, I noticed that my 3 years old Lenovo X270 with Intel HDA & RealTek ALC298 doesn't work with 3.5 mm audio combo jack with 4 pole CTIA. A bit of googling showed me that `snd-hda-intel` module can take some options like `model` to run some fix ups. and I have found that `model=alc298-dell1` seems to at least work. But its gain is way too low. It's mostly just noise. It sounds like my mic is on the line-in. So, I opened `patch_realtek.c` to see what's going on. I found the following but not sure what the values for `.v.pins` are for. ``` [ALC298_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ { 0x1a, 0x01a1913d }, /* use as headphone mic, without its own jack detect */ { } }, .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, ``` In `apply_fix(), these `.v.pins` values are passed to `snd_hda_apply_pincfgs()` but I can't find what values mean. Would someone please enlighten me? My questions are 1) How can I found out which pin goes where. There is x270 schematics on the net but not sure it's correct. 2) What does the value, ie, 0x01a1913d, represent? Thanks, -- yashi

2 1

/dev/sequencer descriptions
by Andrey Konovalov 27 Jul '20

27 Jul '20

Hi Takashi, We're working on syzkaller descriptions for /dev/sequencer [1], but have some troubles understanding its functionality. The main question is: is sound/synth/emux/ code reachable when tested in a VM (we're using QEMU with -soundhw all), or does it require some specific hardware to be present? In our setup (with Debian Stretch userspace image) we have /dev/sequencer with SNDRV_MINOR_OSS_SEQUENCER and /dev/sequencer2 with SNDRV_MINOR_OSS_MUSIC. For the former, there are no synth devices as reported by SNDCTL_SEQ_NRSYNTHS. For the latter, synth devices are there, but the load_patch() callback is not reachable as this check [2] fails. Is that code impossible to test in a VM at all? Or are there some specific kernel options/drivers that need to be enabled? Thanks! [1] https://github.com/google/syzkaller/pull/1952 [2] https://elixir.bootlin.com/linux/v5.7.8/source/sound/core/seq/oss/seq_oss_s…

2 2

[PATCH v2 1/3] ASoC: samsung: Document DT bindings for Midas sound subsystem
by Sylwester Nawrocki 27 Jul '20

27 Jul '20

This patch adds documentation of DT biding for the Midas sound complex. Partially based on the *txt version by Simon Shields <simon(a)lineageos.org>. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes for v2: - fix wrong *-gpios entries in the example, - mark sound-dai properties as required. --- .../bindings/sound/samsung,midas-audio.yaml | 108 +++++++++++++++++++++ 1 file changed, 108 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml new file mode 100644 index 0000000..fb685e51 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -0,0 +1,108 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,midas-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Midas audio complex with WM1811 codec + +maintainers: + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + const: samsung,midas-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle to the I2S controller + required: + - sound-dai + + codec: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle to the WM1811 CODEC node + required: + - sound-dai + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio components; each entry is + a pair of strings, the first being the connection's sink, the second + being the connection's source; valid names for sources and sinks are + the WM1811's pins (as documented in its binding), and the jacks + on the board: HP, SPK, Main Mic, Sub Mic, Headset Mic. + + mic-bias-supply: + description: Supply for the micbias on the Main microphone + + submic-bias-supply: + description: Supply for the micbias on the Sub microphone + + fm-sel-gpios: + description: GPIO pin for FM selection + + lineout-sel-gpios: + description: GPIO pin for line out selection + +required: + - compatible + - model + - cpu + - codec + - samsung,audio-routing + - mic-bias-supply + - submic-bias-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,midas-audio"; + model = "Midas"; + + fm-sel-gpios = <&gpaa0 3 GPIO_ACTIVE_HIGH>; + + mic-bias-supply = <&mic_bias_reg>; + submic-bias-supply = <&submic_bias_reg>; + + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + + "SPK", "SPKOUTLN", + "SPK", "SPKOUTLP", + "SPK", "SPKOUTRN", + "SPK", "SPKOUTRP", + + "RCV", "HPOUT2N", + "RCV", "HPOUT2P", + + "IN1LP", "Main Mic", + "IN1LN", "Main Mic", + "IN1RP", "Sub Mic", + "IN1LP", "Sub Mic"; + + cpu { + sound-dai = <&i2s0>; + }; + + codec { + sound-dai = <&wm1811>; + }; + + }; -- 2.7.4

2 5

[PATCH] usb: endpoint : remove needless check before usb_free_coherent()
by Xu Wang 27 Jul '20

27 Jul '20

usb_free_coherent() is safe with NULL addr and this check is not required. Signed-off-by: Xu Wang <vulab(a)iscas.ac.cn> --- sound/usb/endpoint.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 88760268fb55..3a2b2a309a71 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -615,9 +615,8 @@ static void release_urbs(struct snd_usb_endpoint *ep, int force) for (i = 0; i < ep->nurbs; i++) release_urb_ctx(&ep->urb[i]); - if (ep->syncbuf) - usb_free_coherent(ep->chip->dev, SYNC_URBS * 4, - ep->syncbuf, ep->sync_dma); + usb_free_coherent(ep->chip->dev, SYNC_URBS * 4, + ep->syncbuf, ep->sync_dma); ep->syncbuf = NULL; ep->nurbs = 0; -- 2.17.1

2 1

[PATCH 1/3] ASoC: samsung: Document DT bindings for Midas sound subsystem
by Sylwester Nawrocki 27 Jul '20

27 Jul '20

This patch adds documentation of DT biding for the Midas sound complex. Partially based on the *txt version by Simon Shields <simon(a)lineageos.org>. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- .../bindings/sound/samsung,midas-audio.yaml | 105 +++++++++++++++++++++ 1 file changed, 105 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml new file mode 100644 index 0000000..4a82829 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -0,0 +1,105 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,midas-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Midas audio complex with WM1811 codec + +maintainers: + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + const: samsung,midas-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle to the I2S controller + + codec: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle to the WM1811 CODEC node + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio components; each entry is + a pair of strings, the first being the connection's sink, the second + being the connection's source; valid names for sources and sinks are + the WM1811's pins (as documented in its binding), and the jacks + on the board: HP, SPK, Main Mic, Sub Mic, Headset Mic. + + mic-bias-supply: + description: Supply for the micbias on the Main microphone + + submic-bias-supply: + description: Supply for the micbias on the Sub microphone + + fm-sel-gpios: + description: GPIO pin for FM selection + + lineout-sel-gpios: + description: GPIO pin for line out selection + +required: + - compatible + - model + - cpu + - codec + - samsung,audio-routing + - mic-bias-supply + - submic-bias-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,midas-audio"; + model = "Midas"; + + mic-bias-gpios = <&gpf1 7 GPIO_ACTIVE_HIGH>; + submic-bias-gpios = <&gpf2 0 GPIO_ACTIVE_HIGH>; + + mic-bias-supply = <&mic_bias_reg>; + submic-bias-supply = <&submic_bias_reg>; + + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + + "SPK", "SPKOUTLN", + "SPK", "SPKOUTLP", + "SPK", "SPKOUTRN", + "SPK", "SPKOUTRP", + + "RCV", "HPOUT2N", + "RCV", "HPOUT2P", + + "IN1LP", "Main Mic", + "IN1LN", "Main Mic", + "IN1RP", "Sub Mic", + "IN1LP", "Sub Mic"; + + cpu { + sound-dai = <&i2s0>; + }; + + codec { + sound-dai = <&wm1811>; + }; + + }; -- 2.7.4

3 6

[PATCH 0/2] ASoC: intel: use asoc_substream_to_rtd()
by Kuninori Morimoto 27 Jul '20

27 Jul '20

Hi Mark I have posted "ASoC: add asoc_substream_to_rtd() macro" patch-set to ALSA SoC ML (= see Link), and then Pierre-Louis wanted that Intel patch was separated for boards. The patches which are not for Intel were already accepted. This is for Intel, and Intel boards. Link: https://lore.kernel.org/r/87y2nf0yw2.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/877duz0ysw.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (2): ASoC: intel/boards: use asoc_substream_to_rtd() ASoC: intel: use asoc_substream_to_rtd() sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +++--- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 16 ++++++++-------- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 4 ++-- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 4 ++-- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 4 ++-- sound/soc/intel/boards/ehl_rt5660.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 ++++---- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 ++-- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 2 +- sound/soc/intel/boards/sof_maxim_common.c | 4 ++-- sound/soc/intel/boards/sof_pcm512x.c | 4 ++-- sound/soc/intel/boards/sof_rt5682.c | 4 ++-- sound/soc/intel/boards/sof_sdw_rt1308.c | 2 +- sound/soc/intel/boards/sof_wm8804.c | 2 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 12 ++++++------ sound/soc/intel/keembay/kmb_platform.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 8 ++++---- 34 files changed, 62 insertions(+), 62 deletions(-) -- 2.25.1

2 3

[PATCH] ASoC: dt-bindings: q6asm: Add Q6ASM_DAI_{TX_RX, TX, RX} defines
by Stephan Gerhold 27 Jul '20

27 Jul '20

Right now the direction of a DAI has to be specified as a literal number in the device tree, e.g.: dai@0 { reg = <0>; direction = <2>; }; but this does not make it immediately clear that this is a playback/RX-only DAI. Actually, q6asm-dai.c has useful defines for this. Move them to the dt-bindings header to allow using them in the dts(i) files. The example above then becomes: dai@0 { reg = <0>; direction = <Q6ASM_DAI_RX>; }; which is immediately recognizable as playback/RX-only DAI. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- Documentation/devicetree/bindings/sound/qcom,q6asm.txt | 9 +++++---- include/dt-bindings/sound/qcom,q6asm.h | 4 ++++ sound/soc/qcom/qdsp6/q6asm-dai.c | 3 --- 3 files changed, 9 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt index 6b9a88d0ea3f..8c4883becae9 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt @@ -39,9 +39,9 @@ configuration of each dai. Must contain the following properties. Usage: Required for Compress offload dais Value type: <u32> Definition: Specifies the direction of the dai stream - 0 for both tx and rx - 1 for only tx (Capture/Encode) - 2 for only rx (Playback/Decode) + Q6ASM_DAI_TX_RX (0) for both tx and rx + Q6ASM_DAI_TX (1) for only tx (Capture/Encode) + Q6ASM_DAI_RX (2) for only rx (Playback/Decode) - is-compress-dai: Usage: Required for Compress offload dais @@ -50,6 +50,7 @@ configuration of each dai. Must contain the following properties. = EXAMPLE +#include <dt-bindings/sound/qcom,q6asm.h> apr-service@7 { compatible = "qcom,q6asm"; @@ -62,7 +63,7 @@ apr-service@7 { dai@0 { reg = <0>; - direction = <2>; + direction = <Q6ASM_DAI_RX>; is-compress-dai; }; }; diff --git a/include/dt-bindings/sound/qcom,q6asm.h b/include/dt-bindings/sound/qcom,q6asm.h index 1eb77d87c2e8..f59d74f14395 100644 --- a/include/dt-bindings/sound/qcom,q6asm.h +++ b/include/dt-bindings/sound/qcom,q6asm.h @@ -19,4 +19,8 @@ #define MSM_FRONTEND_DAI_MULTIMEDIA15 14 #define MSM_FRONTEND_DAI_MULTIMEDIA16 15 +#define Q6ASM_DAI_TX_RX 0 +#define Q6ASM_DAI_TX 1 +#define Q6ASM_DAI_RX 2 + #endif /* __DT_BINDINGS_Q6_ASM_H__ */ diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index a2acb7564eb8..9b7b218f2a20 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -37,9 +37,6 @@ #define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024) #define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4) #define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4) -#define Q6ASM_DAI_TX_RX 0 -#define Q6ASM_DAI_TX 1 -#define Q6ASM_DAI_RX 2 #define ALAC_CH_LAYOUT_MONO ((101 << 16) | 1) #define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2) -- 2.27.0

3 2

[PATCH] ASoC: AMD: Restore PME_EN state at Power On
by Akshu Agrawal 27 Jul '20

27 Jul '20

PME_EN state needs to restored to the value set by fmw. For the devices which are not using I2S wake event which gets enabled by PME_EN bit, keeping PME_EN enabled burns considerable amount of power as it blocks low power state. For the devices using I2S wake event, PME_EN gets enabled in fmw and the state should be maintained after ACP Power On. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index ebf4388b6262..31b797c8bfe6 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -19,10 +19,12 @@ struct acp3x_dev_data { bool acp3x_audio_mode; struct resource *res; struct platform_device *pdev[ACP3x_DEVS]; + u32 pme_en; }; -static int acp3x_power_on(void __iomem *acp3x_base) +static int acp3x_power_on(struct acp3x_dev_data *adata) { + void __iomem *acp3x_base = adata->acp3x_base; u32 val; int timeout; @@ -39,10 +41,10 @@ static int acp3x_power_on(void __iomem *acp3x_base) while (++timeout < 500) { val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS); if (!val) { - /* Set PME_EN as after ACP power On, - * PME_EN gets cleared + /* ACP power On clears PME_EN. + * Restore the value to its prior state */ - rv_writel(0x1, acp3x_base + mmACP_PME_EN); + rv_writel(adata->pme_en, acp3x_base + mmACP_PME_EN); return 0; } udelay(1); @@ -74,12 +76,13 @@ static int acp3x_reset(void __iomem *acp3x_base) return -ETIMEDOUT; } -static int acp3x_init(void __iomem *acp3x_base) +static int acp3x_init(struct acp3x_dev_data *adata) { + void __iomem *acp3x_base = adata->acp3x_base; int ret; /* power on */ - ret = acp3x_power_on(acp3x_base); + ret = acp3x_power_on(adata); if (ret) { pr_err("ACP3x power on failed\n"); return ret; @@ -151,7 +154,9 @@ static int snd_acp3x_probe(struct pci_dev *pci, } pci_set_master(pci); pci_set_drvdata(pci, adata); - ret = acp3x_init(adata->acp3x_base); + /* Save ACP_PME_EN state */ + adata->pme_en = rv_readl(adata->acp3x_base + mmACP_PME_EN); + ret = acp3x_init(adata); if (ret) goto disable_msi; @@ -274,7 +279,7 @@ static int snd_acp3x_resume(struct device *dev) struct acp3x_dev_data *adata; adata = dev_get_drvdata(dev); - ret = acp3x_init(adata->acp3x_base); + ret = acp3x_init(adata); if (ret) { dev_err(dev, "ACP init failed\n"); return ret; -- 2.20.1

2 1