March 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH v3 00/17] ASoC: Intel: AVS - Audio DSP for cAVS
by Cezary Rojewski 09 Mar '22

09 Mar '22

A continuation of cleanup work of Intel SST solutions found in sound/soc/intel/. With two major chapters released last year catpt [1] and removal of haswell solution [2], time has come for Skylake-driver. Througout 2019, 2020 and 2021 Skylake-driver has had many fixes applied and even attempts of refactors as seen in fundamental overhaul [3], IPC flow adjustments [4] and LARGE_CONFIG overhaul [5] series. Unfortunately, story repeats itself - problems are found within the core of a driver. Painting it with different colors does not change the fact that is it still a house of cards. As changes needed to address those issues would make Skylake solution incompatible with its previous revisions, a decision has been made to provide a new solution instead. In time it would deprecate and replace Skylake-driver. That solution has been called AVS - from AudioDSP architecture name: Audio-Voice-Speech. It is meant to provide support for the exact same range of platforms as its predecessor: SKL, KBL, AML and APL. Note: this series is dependent upon HDA-series [6] which exposes several codec-organization functions allowing for reduced code size on avs-driver side. Note: this series does not add fully functional driver as its size would get out of control. Here, focus is put on adding IPC protocol and code loading code. Changes v2 -> v3: - fixed variable initialization errors mentioned by sparse Changes v1 -> v2: Almost all updates here are thanks to feedback from Pierre. - several comments and few kernel-docs have been added in areas which felt more or less unclear - avs_ipc_wait_busy_completion() now spins up to 'repeats_left' number of times before giving up - 'adsp_ba' field of struct avs_dev has been renamed to 'dsp_ba' - 'dops' field of struct avs_spec has been renamed to 'dsp_ops' - IPC abstraction has been simplified: SKL_ADSP_REG_HIP* regs are used directly - fixed allnoconfig with AVS enabled compilation issues - fixed code loading error paths: previously requested firmware is now released before function return to the caller - code and function arguments tied to D0IX support have been removed from this patchset and will be part of followup series adding that feature instead - enriched dev_err() messages in avs_dsp_get_core() and avs_dsp_put_core() - numerous wording fixes used in power/reset/stall DSP operations Changes RFC v1 [7]: -> v1: - separated HDA codec-organization patches, path and topology handling, PCM and complementary features such as recovery from this series to ease the review process - fixed EXPORT_SYMBOL_GPL for exported members of ASoC framework - result of stall() is now checked when sending ROM message - result of snd_hdac_ext_stream_set_spib() is now checked when loading basefw - if basefw is not ready, notification processing is now skipped - documented several topology parsing helpers Changes [internal] RFC v2 -> [public] RFC v1: - dropped any sysfs related changes from this series, moved to follow up one - dropped entire subscription-mechanism found in ipc.c. Handlers that are delegated to service certain firmware notifications are now called directly - fixed kernel doc for snd_soc_dapm_new_dai_widgets() as reported by ikp - prefixed snd_hda_codec_device_init() as suggested by Amadeo - improved comments for d0ix transitions for APL-based platforms as suggested by Pierre - a ton of spelling related fixes in most of the commit messages - fixed remaining warnings pointed by scan-build (variable assigned but not used) - replaced most of 'cAVS X.Y' expression usages with 'platform-based' equivalents as suggested by Pierre e.g.: cAVS 1.5 -> SKL-based Changes [internal] RFC v1 -> [internal] RFC v2: - fixed memleak caused by lack of kfree(vols) if memory allocation fails in avs_peakvol_create() as reported by Curtis - fixed missing 'i' iterator incrementation in avs_widget_ready() causing reference loss as reported by Curtis - replace hardcode: 0x40 usage with snd_hdac_calc_stream_format as suggested by Curtis. In consequence, readability for all code loading (CL) procedues has increased and such approach auto-documents the CL stream preparation - updated behavior of all index-fetching functions found in utils.c: avs_module_entry_index(), avs_module_id_entry_index() and follow ups: avs_get_module_entry(), avs_get_module_id_entry() to better conform to linux-kernel standard when no entry is found (return -ENOENT) rather than C++ standard (return -1, what in kernel case translated to -EPERM) as suggested by Curtis and Peter - several suggestions have been made regarding spacing, and so far, I've agreed and applied with all of them. None proposed seemed out of place or redundant - avs_path_stop() renamed to avs_path_pause() pipeline states are represented by RESET/PAUSED/RUNNING. avs_path_reset() and avs_path_run() were already there and avs_path_stop() just didn't look cohesive - added missing parsers for num_output_pin and num_input_pin which are required for custom modules such as WAVES or DSM - dropped 'priv_param_length' from custom module descriptor as this field is obsolete in firmware - parse_dictionary() has been split into parse_dictionary_header() and parse_dictionary_entries() to drop code duplication present in several parsing function which could not re-use entire parse_dictionary() - added avs_tplg_vendor_array_lookup_next() and avs_tplg_vendor_entry_next() to drop code duplicated present in several parsing functions. This change greatly impacted readability of all parsers - parsing helpers such avs_tplg_vendor_array_lookup() now return offset by updating specified in function argument list u32 *offset variable. This is to address problem when u32 offset would be greater than max int, which is the return type for these functions - AVS_DEFINE_PTR_PARSER() macro has been introduced to drop code duplication for all ptr-parsing users - all struct avs_path_module creators have had their declaration updated: function argument *owner ceased to exist as it could already be accessed by mod->owner - fixed the order of operation for conditional paths (e.g.: echo reference) so these are no longer controlled by "source" path and instead are impacted by state changes of source and sink paths both. Previously only source path e.g. playback sourcing echo reference would trigger RUNNING status for conditional path. Equivalent RUNNING on WoV path which is in this case sink path, would not do so, leading to order-of-operation problems. Behavior has been changed to: both source and sink need to be RUNNING for conditional path to be set to RUNNING too. PAUSED for either source or sink will cause PAUSED transition for conditional path. - to achieve the above, path states are now saved in 'state' i.e. new u32 field for struct avs_path - resigned from fw_filename field usage in favour of newly added tplg_filename for machine board descriptors as suggested by Pierre - platform descriptor fields have had their names update better reflect their purpose as suggested by Pierre - fixed comp_list missing locking when manipulated - all message senders now accept request as pointer as suggeseted by Peter - resigned of AZX_ usage for all ADSP-related registers, leaving them only for HOST memory space related operations - fixed disable path for core DSP operations: power/reset/stall as reported by Peter - safety when locking between received responses (reply vs notification) has been lowered as suggested by Pierre. Most usages are not performed in IRQ context and none is done in hard-IRQ one - s/master/main/ plus AVS_MAIN_CORE_MASK has replaced ->master_mask - several functions have had their logging updated - logs have been moved to lower level functions as suggested by Pierre - hdac_ext_stream usage has been streamlined to estream, hdac_streams are represented by hstream instead - hw_params() are resilient to scenarios when they are called mutliple times as reported by Pierre - avs_dsp_enable() now collapses if any of its steps fails as reported by Pierre and Peter - avs_module_ida_empty() now returns value of type bool as suggested by Bard [1]: https://www.spinics.net/lists/alsa-devel/msg116440.html [2]: https://www.spinics.net/lists/alsa-devel/msg116901.html [3]: https://www.spinics.net/lists/alsa-devel/msg94199.html [4]: https://www.spinics.net/lists/alsa-devel/msg92588.html [5]: https://lore.kernel.org/all/20190808181549.12521-1-cezary.rojewski@intel.co… [6]: https://lore.kernel.org/alsa-devel/20220207114906.3759800-1-cezary.rojewski… [7]: https://lore.kernel.org/all/20211208111301.1817725-1-cezary.rojewski@intel.… Cezary Rojewski (17): ALSA: hda: Add helper macros for DSP capable devices ASoC: Export DAI register and widget ctor and dctor functions ASoC: Intel: Introduce AVS driver ASoC: Intel: avs: Inter process communication ASoC: Intel: avs: Add code loading requests ASoC: Intel: avs: Add pipeline management requests ASoC: Intel: avs: Add module management requests ASoC: Intel: avs: Add power management requests ASoC: Intel: avs: Add ROM requests ASoC: Intel: avs: Add basefw runtime-parameter requests ASoC: Intel: avs: Firmware resources management utilities ASoC: Intel: avs: Declare module configuration types ASoC: Intel: avs: Dynamic firmware resources management ASoC: Intel: avs: General code loading flow ASoC: Intel: avs: Implement CLDMA transfer ASoC: Intel: avs: Code loading over CLDMA ASoC: Intel: avs: Code loading over HDA include/sound/hdaudio.h | 2 + include/sound/hdaudio_ext.h | 50 +++ include/sound/soc-dapm.h | 1 + sound/soc/intel/Kconfig | 12 + sound/soc/intel/Makefile | 1 + sound/soc/intel/avs/Makefile | 6 + sound/soc/intel/avs/avs.h | 248 +++++++++++ sound/soc/intel/avs/cldma.c | 317 ++++++++++++++ sound/soc/intel/avs/cldma.h | 29 ++ sound/soc/intel/avs/core.c | 62 +++ sound/soc/intel/avs/dsp.c | 303 +++++++++++++ sound/soc/intel/avs/ipc.c | 393 +++++++++++++++++ sound/soc/intel/avs/loader.c | 607 +++++++++++++++++++++++++ sound/soc/intel/avs/messages.c | 697 +++++++++++++++++++++++++++++ sound/soc/intel/avs/messages.h | 753 ++++++++++++++++++++++++++++++++ sound/soc/intel/avs/registers.h | 75 ++++ sound/soc/intel/avs/utils.c | 301 +++++++++++++ sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 15 + 19 files changed, 3873 insertions(+) create mode 100644 sound/soc/intel/avs/Makefile create mode 100644 sound/soc/intel/avs/avs.h create mode 100644 sound/soc/intel/avs/cldma.c create mode 100644 sound/soc/intel/avs/cldma.h create mode 100644 sound/soc/intel/avs/core.c create mode 100644 sound/soc/intel/avs/dsp.c create mode 100644 sound/soc/intel/avs/ipc.c create mode 100644 sound/soc/intel/avs/loader.c create mode 100644 sound/soc/intel/avs/messages.c create mode 100644 sound/soc/intel/avs/messages.h create mode 100644 sound/soc/intel/avs/registers.h create mode 100644 sound/soc/intel/avs/utils.c -- 2.25.1

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[PATCH 0/3] ASoC: Intel: boards: cleanups for 5.18
by Pierre-Louis Bossart 09 Mar '22

09 Mar '22

Updates to clean-up the GPIOLIB dependency and a quirk for HP SoundWire devices. Anthony I Gilea (1): ASoC: Intel: sof_sdw: fix quirks for 2022 HP Spectre x360 13" Pierre-Louis Bossart (2): ASoC: Intel: boards: remove explicit dependency on GPIOLIB when DMIC is used" ASoC: Intel: boards: add GPIOLIB dependency where missed drivers/soundwire/dmi-quirks.c | 2 +- sound/soc/intel/boards/Kconfig | 49 +++++++++++++++++++------------- sound/soc/intel/boards/sof_sdw.c | 2 +- 3 files changed, 31 insertions(+), 22 deletions(-) -- 2.30.2

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[PATCH v2] ALSA: hda/i915 - avoid hung task timeout in i915 wait
by Kai Vehmanen 09 Mar '22

09 Mar '22

If kernel is built with hung task detection enabled and CONFIG_DEFAULT_HUNG_TASK_TIMEOUT set to less than 60 seconds, snd_hdac_i915_init() will trigger the hung task timeout in case i915 is not available and taint the kernel. Split the 60sec wait into a loop of smaller waits to avoid this. Reviewed-by: Lucas De Marchi <lucas.demarchi(a)intel.com> Co-developed-by: Ramalingam C <ramalingam.c(a)intel.com> Signed-off-by: Ramalingam C <ramalingam.c(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/hda/hdac_i915.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) Changes V1->V2: - address local variable naming issue raised by Amadeusz and use Takashi's proposal diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 454474ac5716..aa48bed7baf7 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -143,7 +143,7 @@ static bool i915_gfx_present(void) int snd_hdac_i915_init(struct hdac_bus *bus) { struct drm_audio_component *acomp; - int err; + int err, i; if (!i915_gfx_present()) return -ENODEV; @@ -159,9 +159,11 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (!acomp->ops) { if (!IS_ENABLED(CONFIG_MODULES) || !request_module("i915")) { - /* 60s timeout */ - wait_for_completion_timeout(&acomp->master_bind_complete, - msecs_to_jiffies(60 * 1000)); + /* max 60s timeout */ + for (i = 0; i < 60; i++) + if (wait_for_completion_timeout(&acomp->master_bind_complete, + msecs_to_jiffies(1000))) + break; } } if (!acomp->ops) { base-commit: fd7698cf0858f8c5e659b655109cd93c2f15cdd3 -- 2.35.1

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[PATCH] ASoC: tas5805m: fix pdn polarity
by John Keeping 09 Mar '22

09 Mar '22

The binding defines the GPIO as "pdn-gpios" so when the GPIO is active the expectation is that the power down signal is asserted, but the driver swaps this to represent the inverted logic of the electrical signal. The GPIO_ACTIVE_HIGH/LOW flags should be used to identify the inverted logic here with the driver treating power down as active when the mapped GPIO value is 1. Fixes: ec45268467f4 ("ASoC: add support for TAS5805M digital amplifier") Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/codecs/tas5805m.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tas5805m.c b/sound/soc/codecs/tas5805m.c index fa0e81ec875a..12146a860ef8 100644 --- a/sound/soc/codecs/tas5805m.c +++ b/sound/soc/codecs/tas5805m.c @@ -155,7 +155,7 @@ static const uint32_t tas5805m_volume[] = { struct tas5805m_priv { struct regulator *pvdd; - struct gpio_desc *gpio_pdn_n; + struct gpio_desc *gpio_pdn; uint8_t *dsp_cfg_data; int dsp_cfg_len; @@ -444,11 +444,11 @@ static int tas5805m_i2c_probe(struct i2c_client *i2c) dev_set_drvdata(dev, tas5805m); tas5805m->regmap = regmap; - tas5805m->gpio_pdn_n = devm_gpiod_get(dev, "pdn", GPIOD_OUT_LOW); - if (IS_ERR(tas5805m->gpio_pdn_n)) { + tas5805m->gpio_pdn = devm_gpiod_get(dev, "pdn", GPIOD_OUT_HIGH); + if (IS_ERR(tas5805m->gpio_pdn)) { dev_err(dev, "error requesting PDN gpio: %ld\n", - PTR_ERR(tas5805m->gpio_pdn_n)); - return PTR_ERR(tas5805m->gpio_pdn_n); + PTR_ERR(tas5805m->gpio_pdn)); + return PTR_ERR(tas5805m->gpio_pdn); } /* This configuration must be generated by PPC3. The file loaded @@ -505,7 +505,7 @@ static int tas5805m_i2c_probe(struct i2c_client *i2c) } usleep_range(100000, 150000); - gpiod_set_value(tas5805m->gpio_pdn_n, 1); + gpiod_set_value(tas5805m->gpio_pdn, 0); usleep_range(10000, 15000); /* Don't register through devm. We need to be able to unregister @@ -515,7 +515,7 @@ static int tas5805m_i2c_probe(struct i2c_client *i2c) &tas5805m_dai, 1); if (ret < 0) { dev_err(dev, "unable to register codec: %d\n", ret); - gpiod_set_value(tas5805m->gpio_pdn_n, 0); + gpiod_set_value(tas5805m->gpio_pdn, 1); regulator_disable(tas5805m->pvdd); return ret; } @@ -529,7 +529,7 @@ static int tas5805m_i2c_remove(struct i2c_client *i2c) struct tas5805m_priv *tas5805m = dev_get_drvdata(dev); snd_soc_unregister_component(dev); - gpiod_set_value(tas5805m->gpio_pdn_n, 0); + gpiod_set_value(tas5805m->gpio_pdn, 1); usleep_range(10000, 15000); regulator_disable(tas5805m->pvdd); return 0; -- 2.35.1

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[PATCH v3] arm64: dts: imx8mn-bsh-smm-s2pro: Add tlv320aic31xx audio card node
by Ariel D'Alessandro 09 Mar '22

09 Mar '22

BSH SystemMaster (SMM) S2 PRO board comes with an audio card based on tlv320aic31xx family codec. The audio card exposes two playback devices, one of them using the EASRC (Enhanced Asynchronous Sample Rate Converter) module. Note that this would require SDMA and EASRC firmware in order to work. Signed-off-by: Ariel D'Alessandro <ariel.dalessandro(a)collabora.com> Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> --- .../dts/freescale/imx8mn-bsh-smm-s2pro.dts | 94 +++++++++++++++++++ 1 file changed, 94 insertions(+) diff --git a/arch/arm64/boot/dts/freescale/imx8mn-bsh-smm-s2pro.dts b/arch/arm64/boot/dts/freescale/imx8mn-bsh-smm-s2pro.dts index c6a8ed6745c1..44e2635d3096 100644 --- a/arch/arm64/boot/dts/freescale/imx8mn-bsh-smm-s2pro.dts +++ b/arch/arm64/boot/dts/freescale/imx8mn-bsh-smm-s2pro.dts @@ -7,6 +7,7 @@ /dts-v1/; #include "imx8mn-bsh-smm-s2-common.dtsi" +#include <dt-bindings/sound/tlv320aic31xx.h> / { model = "BSH SMM S2 PRO"; @@ -16,6 +17,69 @@ memory@40000000 { device_type = "memory"; reg = <0x0 0x40000000 0x0 0x20000000>; }; + + sound-tlv320aic31xx { + compatible = "fsl,imx-audio-tlv320aic31xx"; + model = "tlv320aic31xx-hifi"; + audio-cpu = <&sai3>; + audio-codec = <&codec>; + audio-asrc = <&easrc>; + audio-routing = + "Ext Spk", "SPL", + "Ext Spk", "SPR"; + mclk-id = <PLL_CLKIN_BCLK>; + }; + + vdd_input: vdd_input { + compatible = "regulator-fixed"; + regulator-name = "vdd_input"; + regulator-min-microvolt = <5000000>; + regulator-max-microvolt = <5000000>; + }; +}; + +&easrc { + fsl,asrc-rate = <48000>; + fsl,asrc-format = <10>; + status = "okay"; +}; + +&i2c2 { + clock-frequency = <400000>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2c2>; + status = "okay"; + + tlv320dac3101: audio-codec@18 { + #sound-dai-cells = <0>; + compatible = "ti,tlv320dac3101"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_dac_rst>; + reg = <0x18>; + + ai31xx-micbias-vg = <MICBIAS_AVDDV>; + + HPVDD-supply = <&buck4_reg>; + SPRVDD-supply = <&vdd_input>; + SPLVDD-supply = <&vdd_input>; + AVDD-supply = <&buck4_reg>; + IOVDD-supply = <&buck4_reg>; + DVDD-supply = <&buck5_reg>; + reset-gpios = <&gpio1 6 GPIO_ACTIVE_LOW>; + + clocks = <&clk IMX8MN_CLK_SAI3_ROOT>; + clock-names = "mclk"; + }; +}; + +&sai3 { + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai3>; + assigned-clocks = <&clk IMX8MN_CLK_SAI3>; + assigned-clock-parents = <&clk IMX8MN_AUDIO_PLL1_OUT>; + assigned-clock-rates = <24576000>; + fsl,sai-mclk-direction-output; + status = "okay"; }; /* eMMC */ @@ -30,6 +94,36 @@ &usdhc1 { }; &iomuxc { + pinctrl_dac_rst: dacrstgrp { + fsl,pins = < + MX8MN_IOMUXC_GPIO1_IO06_GPIO1_IO6 0x19 /* DAC_RST */ + >; + }; + + pinctrl_espi2: espi2grp { + fsl,pins = < + MX8MN_IOMUXC_ECSPI2_SCLK_ECSPI2_SCLK 0x082 + MX8MN_IOMUXC_ECSPI2_MOSI_ECSPI2_MOSI 0x082 + MX8MN_IOMUXC_ECSPI2_MISO_ECSPI2_MISO 0x082 + MX8MN_IOMUXC_ECSPI2_SS0_ECSPI2_SS0 0x040 + >; + }; + + pinctrl_i2c2: i2c2grp { + fsl,pins = < + MX8MN_IOMUXC_I2C2_SCL_I2C2_SCL 0x400000c3 + MX8MN_IOMUXC_I2C2_SDA_I2C2_SDA 0x400000c3 + >; + }; + + pinctrl_sai3: sai3grp { + fsl,pins = < + MX8MN_IOMUXC_SAI3_TXFS_SAI3_TX_SYNC 0xd6 + MX8MN_IOMUXC_SAI3_TXC_SAI3_TX_BCLK 0xd6 + MX8MN_IOMUXC_SAI3_TXD_SAI3_TX_DATA0 0xd6 + >; + }; + pinctrl_usdhc1: usdhc1grp { fsl,pins = < MX8MN_IOMUXC_SD1_CLK_USDHC1_CLK 0x40000090 -- 2.34.1

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[PATCH] ALSA: hda: Add AlderLake-PS variant PCI ID
by Kai Vehmanen 09 Mar '22

09 Mar '22

Add HD Audio PCI ID for a variant of Intel AlderLake-P. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5af7a1e18013..0a83eb6b88b1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2499,6 +2499,8 @@ static const struct pci_device_id azx_ids[] = { /* Alderlake-P */ { PCI_DEVICE(0x8086, 0x51c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x51c9), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, { PCI_DEVICE(0x8086, 0x51cd), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Alderlake-M */ base-commit: fd7698cf0858f8c5e659b655109cd93c2f15cdd3 prerequisite-patch-id: 0259e96fec7d4c6138df07807ca8fbc6f73e0c31 -- 2.35.1

2 1

[PATCH v7 0/2] Add support for SoundWire1.6 audio cgcr register control
by Srinivasa Rao Mandadapu 08 Mar '22

08 Mar '22

This patch series is to add v1.6.0 compatible name for qcom soundwire driver and corresponding dt bindings. Changes Since V6: -- Update commit message -- Update Example with reset control in dt bindings. Changes Since V5: -- Constify static struct qcom_swrm_data global variables. -- Remove redundant swrm_v1_6_data variable. Changes Since V4: -- Dropped audio cgcr control patch due to dependency on clock patches. -- Update dt-bindings as per new reset control properties. Changes Since V3: -- Add v1.6.0 compatible name and soundwire data structure. -- Change macro define name properly. -- Update bindings for new property. -- Change commit message description. -- Change signedoff by sequence. Changes since v2: -- Update error check after ioremap. Changes since v1: -- Add const name to mask value. Srinivasa Rao Mandadapu (2): soundwire: qcom: Add compatible name for v1.6.0 dt-bindings: soundwire: qcom: Add bindings for audio clock reset control property Documentation/devicetree/bindings/soundwire/qcom,sdw.txt | 14 ++++++++++++++ drivers/soundwire/qcom.c | 1 + 2 files changed, 15 insertions(+) -- 2.7.4

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[PATCH] ASoC: mxs: Fix error handling in mxs_sgtl5000_probe
by Miaoqian Lin 08 Mar '22

08 Mar '22

This function only calls of_node_put() in the regular path. And it will cause refcount leak in error paths. For example, when codec_np is NULL, saif_np[0] and saif_np[1] are not NULL, it will cause leaks. of_node_put() will check if the node pointer is NULL, so we can call it directly to release the refcount of regular pointers. Fixes: e968194b45c4 ("ASoC: mxs: add device tree support for mxs-sgtl5000") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/mxs/mxs-sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 2412dc7e65d4..746f40938675 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -118,6 +118,9 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev) codec_np = of_parse_phandle(np, "audio-codec", 0); if (!saif_np[0] || !saif_np[1] || !codec_np) { dev_err(&pdev->dev, "phandle missing or invalid\n"); + of_node_put(codec_np); + of_node_put(saif_np[0]); + of_node_put(saif_np[1]); return -EINVAL; } -- 2.17.1

2 1

[PATCH] ASoC: rk817: Fix missing clk_disable_unprepare() in rk817_platform_probe
by Miaoqian Lin 08 Mar '22

08 Mar '22

Fix the missing clk_disable_unprepare() before return from rk817_platform_probe() in the error handling case. Fixes: 0d6a04da9b25 ("ASoC: Add Rockchip rk817 audio CODEC support") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/codecs/rk817_codec.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 03f24edfe4f6..8fffe378618d 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -508,12 +508,14 @@ static int rk817_platform_probe(struct platform_device *pdev) if (ret < 0) { dev_err(&pdev->dev, "%s() register codec error %d\n", __func__, ret); - goto err_; + goto err_clk; } return 0; -err_: +err_clk: + clk_disable_unprepare(rk817_codec_data->mclk); +err_: return ret; } -- 2.17.1

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[PATCH] ASoC: SOF: Add missing of_node_put() in imx8m_probe
by Miaoqian Lin 08 Mar '22

08 Mar '22

The device_node pointer is returned by of_parse_phandle() with refcount incremented. We should use of_node_put() on it when done. Fixes: afb93d716533 ("ASoC: SOF: imx: Add i.MX8M HW support") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/sof/imx/imx8m.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index 788e77bcb603..60251486b24b 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -224,6 +224,7 @@ static int imx8m_probe(struct snd_sof_dev *sdev) } ret = of_address_to_resource(res_node, 0, &res); + of_node_put(res_node); if (ret) { dev_err(&pdev->dev, "failed to get reserved region address\n"); goto exit_pdev_unregister; -- 2.17.1

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