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[PATCH AUTOSEL 5.4 1/5] ASoC: ops: Check bounds for second channel in snd_soc_put_volsw_sx()
by Sasha Levin 06 Dec '22

06 Dec '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 97eea946b93961fffd29448dcda7398d0d51c4b2 ] The bounds checks in snd_soc_put_volsw_sx() are only being applied to the first channel, meaning it is possible to write out of bounds values to the second channel in stereo controls. Add appropriate checks. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220511134137.169575-2-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 453b61b42dd9..00e6a6e46fe5 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -460,6 +460,12 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, if (snd_soc_volsw_is_stereo(mc)) { val_mask = mask << rshift; val2 = (ucontrol->value.integer.value[1] + min) & mask; + + if (mc->platform_max && val2 > mc->platform_max) + return -EINVAL; + if (val2 > max) + return -EINVAL; + val2 = val2 << rshift; err = snd_soc_component_update_bits(component, reg2, val_mask, -- 2.35.1

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[PATCH AUTOSEL 5.10 01/10] ASoC: fsl_micfil: explicitly clear software reset bit
by Sasha Levin 06 Dec '22

06 Dec '22

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 292709b9cf3ba470af94b62c9bb60284cc581b79 ] SRES is self-cleared bit, but REG_MICFIL_CTRL1 is defined as non volatile register, it still remain in regmap cache after set, then every update of REG_MICFIL_CTRL1, software reset happens. to avoid this, clear it explicitly. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Link: https://lore.kernel.org/r/1651925654-32060-1-git-send-email-shengjiu.wang@n… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_micfil.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index efc5daf53bba..ead4bfa13561 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -190,6 +190,17 @@ static int fsl_micfil_reset(struct device *dev) return ret; } + /* + * SRES is self-cleared bit, but REG_MICFIL_CTRL1 is defined + * as non-volatile register, so SRES still remain in regmap + * cache after set, that every update of REG_MICFIL_CTRL1, + * software reset happens. so clear it explicitly. + */ + ret = regmap_clear_bits(micfil->regmap, REG_MICFIL_CTRL1, + MICFIL_CTRL1_SRES); + if (ret) + return ret; + return 0; } -- 2.35.1

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[PATCH AUTOSEL 5.15 01/12] ASoC: fsl_micfil: explicitly clear software reset bit
by Sasha Levin 06 Dec '22

06 Dec '22

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 292709b9cf3ba470af94b62c9bb60284cc581b79 ] SRES is self-cleared bit, but REG_MICFIL_CTRL1 is defined as non volatile register, it still remain in regmap cache after set, then every update of REG_MICFIL_CTRL1, software reset happens. to avoid this, clear it explicitly. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Link: https://lore.kernel.org/r/1651925654-32060-1-git-send-email-shengjiu.wang@n… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_micfil.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 9f90989ac59a..cb84d95c3aac 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -191,6 +191,17 @@ static int fsl_micfil_reset(struct device *dev) return ret; } + /* + * SRES is self-cleared bit, but REG_MICFIL_CTRL1 is defined + * as non-volatile register, so SRES still remain in regmap + * cache after set, that every update of REG_MICFIL_CTRL1, + * software reset happens. so clear it explicitly. + */ + ret = regmap_clear_bits(micfil->regmap, REG_MICFIL_CTRL1, + MICFIL_CTRL1_SRES); + if (ret) + return ret; + return 0; } -- 2.35.1

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[PATCH AUTOSEL 6.0 01/13] ASoC: fsl_micfil: explicitly clear software reset bit
by Sasha Levin 06 Dec '22

06 Dec '22

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 292709b9cf3ba470af94b62c9bb60284cc581b79 ] SRES is self-cleared bit, but REG_MICFIL_CTRL1 is defined as non volatile register, it still remain in regmap cache after set, then every update of REG_MICFIL_CTRL1, software reset happens. to avoid this, clear it explicitly. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Link: https://lore.kernel.org/r/1651925654-32060-1-git-send-email-shengjiu.wang@n… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_micfil.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 79ef4e269bc9..8aa6871e0d42 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -194,6 +194,17 @@ static int fsl_micfil_reset(struct device *dev) if (ret) return ret; + /* + * SRES is self-cleared bit, but REG_MICFIL_CTRL1 is defined + * as non-volatile register, so SRES still remain in regmap + * cache after set, that every update of REG_MICFIL_CTRL1, + * software reset happens. so clear it explicitly. + */ + ret = regmap_clear_bits(micfil->regmap, REG_MICFIL_CTRL1, + MICFIL_CTRL1_SRES); + if (ret) + return ret; + return 0; } -- 2.35.1

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Query about xrun on usb/pcm
by Carl Hetherington 06 Dec '22

06 Dec '22

Hi all, I wonder if anybody has any clues/suggestions about problem I'm seeing with an XMOS-based USB sound card. As far as I can see, the card has endpoint 0x82 set up for capture data, 0x2 for playback data, and 0x82 is also used as the sync endpoint for playback. I'm assuming that's a fairly common arrangement? I am testing it using simultaneous playback and capture, and simulating a high-CPU-load case by sleeping for long enough to cause a lot of xruns. After some random time, I see a failure case that I'm struggling to explain. It goes like this: There's an XRUN on the playback PCM, so __snd_pcm_xrun happens, then stop_endpoints() happens, and: it decides not to stop 0x82 because its running is > 1 it stops 0x2, so its state goes to EP_STATE_STOPPING Then the ALSA userspace code calls prepare on the playback PCM to get it going again. This ends up in wait_clear_urbs(), which does nothing with 0x82 as it is still running. At this point, the prepare thread is interrupted by an XRUN on the capture PCM. With this PCM, there is no sync endpoint, and 0x82 is the data endpoint. In the xrun handler: stop_urbs() sets 0x82 to EP_STATE_STOPPING. ... and the xrun handler finishes. Then we end up back in the prepare for the playback PCM. wait_clear_urbs() then sets 0x2 to STOPPED, and the prepare is finished. Now, snd_usb_endpoint_start() is called on 0x2 and that is fine. Next, snd_usb_endpoint_start() is called on 0x82 and that fails because its state is still STOPPING. At this point things seem broken. Does anyone have a hint about where in this sequence things are going wrong, and maybe even why? I'm more than happy to clarify anything I can, or provide more debugging information. Thanks in advance! Best, Carl

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[PATCH v2 00/16] ASoC: Intel: avs: Data probing and fw logging
by Cezary Rojewski 05 Dec '22

05 Dec '22

The patchset focuses on debug functionality for the avs-driver. Two major blocks are covered here: data probing and AudioDSP firmware logging. Both are configured and controlled through debugfs. Data probing is a AudioDSP debug functionality which allows for gathering the actual data that is being routed to or from a module. Helps in debugging its processing capabilities - navigate to a specific module which may have caused a glitch within a pipeline (set of modules bound together). First few allow for assigning compress stream to a HDAudio stream, what is currently limited to pcm substreams only. These patches were already present on this list and reviewed in the past [1]. The next few tidy existing debug-related code up so it's ready for addition of new functionalities and make it clear which part of the avs is debug related and which is not. These also simplify the existing locking around the trace fifo. Afterward, debug-related IPCs are defined along with stub soc-component and compress DAI operations. Not much is done there as it's not a standard PCM streaming scenario. Most code found in compress operations is inherited from the HOST side of HDAudio streaming found in pcm.c file of the driver. Finally, a debugfs file operations are defined. These facilitate connecting to DSP modules from which the data shall be gathered as well as control and configuration of firmware logging. Additionally, entries are added to allow for dumping snapshots of key memory windows. Changes in v2: - reordered sign offs in patch 03 as reported by Takashi - fixed issue reported by ikp - unintilialized 'runtime' variable in patch 03 - added missing module_get_upon_open in component driver declaration [1]: https://lore.kernel.org/alsa-devel/20200124190413.18154-1-cezary.rojewski@i… Cezary Rojewski (16): ALSA: hda: Allow for compress stream to hdac_ext_stream assignment ALSA: hda: Prepare for compress stream support ALSA: hda: Interrupt servicing and BDL setup for compress streams ASoC: Intel: avs: Introduce avs_log_buffer_status_locked() ASoC: Intel: avs: Drop fifo_lock ASoC: Intel: avs: Introduce debug-context aware helpers ASoC: Intel: avs: Make enable_logs() dependent on DEBUG_FS ASoC: Intel: avs: Drop usage of debug members in non-debug code ASoC: Intel: avs: Add data probing requests ASoC: Intel: avs: Probe compress operations ASoC: Intel: avs: Data probing soc-component ASoC: Intel: avs: Add probe machine board ASoC: Intel: avs: Probing and firmware tracing over debugfs ASoC: Intel: avs: Gather remaining logs on strace_release() ASoC: Intel: avs: Allow for dumping FW_REGS area ASoC: Intel: avs: Allow for dumping debug window snapshot include/sound/hdaudio_ext.h | 2 + sound/hda/ext/hdac_ext_stream.c | 41 +++ sound/hda/hdac_controller.c | 4 +- sound/hda/hdac_stream.c | 51 +-- sound/soc/intel/Kconfig | 1 + sound/soc/intel/avs/Makefile | 4 + sound/soc/intel/avs/apl.c | 24 +- sound/soc/intel/avs/avs.h | 80 ++++- sound/soc/intel/avs/board_selection.c | 33 ++ sound/soc/intel/avs/boards/Kconfig | 8 + sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/probe.c | 64 ++++ sound/soc/intel/avs/core.c | 2 + sound/soc/intel/avs/debugfs.c | 436 ++++++++++++++++++++++++++ sound/soc/intel/avs/ipc.c | 2 +- sound/soc/intel/avs/messages.c | 104 +++++- sound/soc/intel/avs/messages.h | 53 ++++ sound/soc/intel/avs/pcm.c | 6 +- sound/soc/intel/avs/probes.c | 313 ++++++++++++++++++ sound/soc/intel/avs/registers.h | 3 +- sound/soc/intel/avs/skl.c | 17 +- sound/soc/intel/avs/utils.c | 22 -- 22 files changed, 1172 insertions(+), 100 deletions(-) create mode 100644 sound/soc/intel/avs/boards/probe.c create mode 100644 sound/soc/intel/avs/debugfs.c create mode 100644 sound/soc/intel/avs/probes.c -- 2.25.1

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[PATCH] ASoC: sof_es8336: fix possible use-after-free in sof_es8336_remove()
by Yang Yingliang 05 Dec '22

05 Dec '22

sof_es8336_remove() calls cancel_delayed_work(). However, that function does not wait until the work function finishes. This means that the callback function may still be running after the driver's remove function has finished, which would result in a use-after-free. Fix by calling cancel_delayed_work_sync(), which ensures that the work is properly cancelled, no longer running, and unable to re-schedule itself. Fixes: 89cdb224f2ab ("ASoC: sof_es8336: reduce pop noise on speaker") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/intel/boards/sof_es8336.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index 70713e4b07dc..773e5d1d87d4 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -783,7 +783,7 @@ static int sof_es8336_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); - cancel_delayed_work(&priv->pcm_pop_work); + cancel_delayed_work_sync(&priv->pcm_pop_work); gpiod_put(priv->gpio_speakers); device_remove_software_node(priv->codec_dev); put_device(priv->codec_dev); -- 2.25.1

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[PATCH 0/2] Add SOF panic dump support for AMD platform.
by V sujith kumar Reddy 05 Dec '22

05 Dec '22

This patch series consists of >> Use poll function instead of simple read ACP_SHA_DSP_FW_QUALIFIER >> Add support for IPC and DSP dumps Ajye Huang (1): ASoC: SOF: amd: Use poll function instead to read ACP_SHA_DSP_FW_QUALIFIER V sujith kumar Reddy (1): ASoC : SOF: amd: Add support for IPC and DSP dumps sound/soc/sof/amd/Kconfig | 1 + sound/soc/sof/amd/acp-common.c | 108 +++++++++++++++++++++++++++++++++ sound/soc/sof/amd/acp-ipc.c | 13 ++++ sound/soc/sof/amd/acp.c | 8 ++- sound/soc/sof/amd/acp.h | 6 ++ 5 files changed, 133 insertions(+), 3 deletions(-) -- 2.25.1

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[PATCH] ASoC: qcom: lpass-sc7280: Add maybe_unused tag for system PM ops
by Srinivasa Rao Mandadapu 05 Dec '22

05 Dec '22

Add __maybe_unused tag for system PM ops suspend and resume. This is required to fix allmodconfig compilation issue. Fixes: a3a96e93cc88 ("ASoC: qcom: lpass-sc7280: Add system suspend/resume PM ops") Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> --- sound/soc/qcom/lpass-sc7280.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/lpass-sc7280.c b/sound/soc/qcom/lpass-sc7280.c index 65a04d3..d43f480 100644 --- a/sound/soc/qcom/lpass-sc7280.c +++ b/sound/soc/qcom/lpass-sc7280.c @@ -235,14 +235,14 @@ static int sc7280_lpass_exit(struct platform_device *pdev) return 0; } -static int sc7280_lpass_dev_resume(struct device *dev) +static int __maybe_unused sc7280_lpass_dev_resume(struct device *dev) { struct lpass_data *drvdata = dev_get_drvdata(dev); return clk_bulk_prepare_enable(drvdata->num_clks, drvdata->clks); } -static int sc7280_lpass_dev_suspend(struct device *dev) +static int __maybe_unused sc7280_lpass_dev_suspend(struct device *dev) { struct lpass_data *drvdata = dev_get_drvdata(dev); @@ -253,6 +253,7 @@ static int sc7280_lpass_dev_suspend(struct device *dev) static const struct dev_pm_ops sc7280_lpass_pm_ops = { SET_SYSTEM_SLEEP_PM_OPS(sc7280_lpass_dev_suspend, sc7280_lpass_dev_resume) }; + static struct lpass_variant sc7280_data = { .i2sctrl_reg_base = 0x1000, .i2sctrl_reg_stride = 0x1000, -- 2.7.4

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[PATCH] ASoC: dt-bindings: maxim,max98504: Convert to DT schema
by Krzysztof Kozlowski 05 Dec '22

05 Dec '22

Convert the Maxim Integrated MAX98504 amplifier bindings to DT schema. Few properties are made optional: 1. interrupts: current Linux driver implementation does not use them, 2. supplies: on some boards these might be wired to battery, for which no regulator is provided. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../devicetree/bindings/sound/max98504.txt | 44 ---------- .../bindings/sound/maxim,max98504.yaml | 86 +++++++++++++++++++ 2 files changed, 86 insertions(+), 44 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/max98504.txt create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98504.yaml diff --git a/Documentation/devicetree/bindings/sound/max98504.txt b/Documentation/devicetree/bindings/sound/max98504.txt deleted file mode 100644 index 583ed5fdfb28..000000000000 --- a/Documentation/devicetree/bindings/sound/max98504.txt +++ /dev/null @@ -1,44 +0,0 @@ -Maxim MAX98504 class D mono speaker amplifier - -This device supports I2C control interface and an IRQ output signal. It features -a PCM and PDM digital audio interface (DAI) and a differential analog input. - -Required properties: - - - compatible : "maxim,max98504" - - reg : should contain the I2C slave device address - - DVDD-supply, DIOVDD-supply, PVDD-supply: power supplies for the device, - as covered in ../regulator/regulator.txt - - interrupts : should specify the interrupt line the device is connected to, - as described in ../interrupt-controller/interrupts.txt - -Optional properties: - - - maxim,brownout-threshold - the PVDD brownout threshold, the value must be - from 0, 1...21 range, corresponding to 2.6V, 2.65V...3.65V voltage range - - maxim,brownout-attenuation - the brownout attenuation to the speaker gain - applied during the "attack hold" and "timed hold" phase, the value must be - from 0...6 (dB) range - - maxim,brownout-attack-hold-ms - the brownout attack hold phase time in ms, - 0...255 (VBATBROWN_ATTK_HOLD, register 0x0018) - - maxim,brownout-timed-hold-ms - the brownout timed hold phase time in ms, - 0...255 (VBATBROWN_TIME_HOLD, register 0x0019) - - maxim,brownout-release-rate-ms - the brownout release phase step time in ms, - 0...255 (VBATBROWN_RELEASE, register 0x001A) - -The default value when the above properties are not specified is 0, -the maxim,brownout-threshold property must be specified to actually enable -the PVDD brownout protection. - -Example: - - max98504@31 { - compatible = "maxim,max98504"; - reg = <0x31>; - interrupt-parent = <&gpio_bank_0>; - interrupts = <2 0>; - - DVDD-supply = <&regulator>; - DIOVDD-supply = <&regulator>; - PVDD-supply = <&regulator>; -}; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98504.yaml b/Documentation/devicetree/bindings/sound/maxim,max98504.yaml new file mode 100644 index 000000000000..23f19a9d2c06 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98504.yaml @@ -0,0 +1,86 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98504.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98504 class D mono speaker amplifier + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> + +description: + Maxim Integrated MAX98504 speaker amplifier supports I2C control interface + with an IRQ output signal, PCM and PDM digital audio interface (DAI) and a + differential analog input. + +properties: + compatible: + const: maxim,max98504 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + DIOVDD-supply: true + DVDD-supply: true + PVDD-supply: true + + maxim,brownout-threshold: + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 21 + default: 0 + description: + PVDD brownout threshold, where values correspond to 2.6V, 2.65V...3.65V + voltage range. Property also enables the PVDD brownout protection. + + maxim,brownout-attenuation: + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 6 + default: 0 + description: + Brownout attenuation to the speaker gain applied during the "attack hold" + and "timed hold" phase, the value must be from 0...6 (dB) range. + + maxim,brownout-attack-hold-ms: + maximum: 255 + default: 0 + description: + Brownout attack hold phase time in ms, VBATBROWN_ATTK_HOLD, register 0x0018. + + maxim,brownout-timed-hold-ms: + maximum: 255 + default: 0 + description: + Brownout timed hold phase time in ms, VBATBROWN_TIME_HOLD, register 0x0019. + + maxim,brownout-release-rate-ms: + maximum: 255 + default: 0 + description: + Brownout release phase step time in ms, VBATBROWN_RELEASE, register 0x001A. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@31 { + compatible = "maxim,max98504"; + reg = <0x31>; + + DIOVDD-supply = <&ldo3_reg>; + DVDD-supply = <&ldo3_reg>; + }; + }; -- 2.34.1

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