May 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH RESEND] MAINTAINERS: Remove self from Cirrus Codec drivers
by Richard Fitzgerald 15 May '23

15 May '23

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> I'm leaving Cirrus Logic, and will no longer have access to hardware and documentation necessary to be effective in a maintainership role. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- MAINTAINERS | 1 - 1 file changed, 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index c919c282f599..5c37070138f2 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4923,7 +4923,6 @@ F: drivers/media/cec/i2c/ch7322.c CIRRUS LOGIC AUDIO CODEC DRIVERS M: James Schulman <james.schulman(a)cirrus.com> M: David Rhodes <david.rhodes(a)cirrus.com> -M: Lucas Tanure <tanureal(a)opensource.cirrus.com> M: Richard Fitzgerald <rf(a)opensource.cirrus.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) L: patches(a)opensource.cirrus.com -- 2.30.2

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[PATCH] ASoC: mediatek: mt8186: Fix use-after-free in driver remove path
by Douglas Anderson 15 May '23

15 May '23

When devm runs function in the "remove" path for a device it runs them in the reverse order. That means that if you have parts of your driver that aren't using devm or are using "roll your own" devm w/ devm_add_action_or_reset() you need to keep that in mind. The mt8186 audio driver didn't quite get this right. Specifically, in mt8186_init_clock() it called mt8186_audsys_clk_register() and then went on to call a bunch of other devm function. The caller of mt8186_init_clock() used devm_add_action_or_reset() to call mt8186_deinit_clock() but, because of the intervening devm functions, the order was wrong. Specifically at probe time, the order was: 1. mt8186_audsys_clk_register() 2. afe_priv->clk = devm_kcalloc(...) 3. afe_priv->clk[i] = devm_clk_get(...) At remove time, the order (which should have been 3, 2, 1) was: 1. mt8186_audsys_clk_unregister() 3. Free all of afe_priv->clk[i] 2. Free afe_priv->clk The above seemed to be causing a use-after-free. Luckily, it's easy to fix this by simply using devm more correctly. Let's move the devm_add_action_or_reset() to the right place. In addition to fixing the use-after-free, code inspection shows that this fixes a leak (missing call to mt8186_audsys_clk_unregister()) that would have happened if any of the syscon_regmap_lookup_by_phandle() calls in mt8186_init_clock() had failed. Fixes: 55b423d5623c ("ASoC: mediatek: mt8186: support audio clock control in platform driver") Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 6 --- sound/soc/mediatek/mt8186/mt8186-afe-clk.h | 1 - sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 4 -- sound/soc/mediatek/mt8186/mt8186-audsys-clk.c | 46 ++++++++++--------- sound/soc/mediatek/mt8186/mt8186-audsys-clk.h | 1 - 5 files changed, 24 insertions(+), 34 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-clk.c b/sound/soc/mediatek/mt8186/mt8186-afe-clk.c index a6b4f29049bb..539e3a023bc4 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-clk.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-clk.c @@ -644,9 +644,3 @@ int mt8186_init_clock(struct mtk_base_afe *afe) return 0; } - -void mt8186_deinit_clock(void *priv) -{ - struct mtk_base_afe *afe = priv; - mt8186_audsys_clk_unregister(afe); -} diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-clk.h b/sound/soc/mediatek/mt8186/mt8186-afe-clk.h index d5988717d8f2..a9d59e506d9a 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-clk.h +++ b/sound/soc/mediatek/mt8186/mt8186-afe-clk.h @@ -81,7 +81,6 @@ enum { struct mtk_base_afe; int mt8186_set_audio_int_bus_parent(struct mtk_base_afe *afe, int clk_id); int mt8186_init_clock(struct mtk_base_afe *afe); -void mt8186_deinit_clock(void *priv); int mt8186_afe_enable_cgs(struct mtk_base_afe *afe); void mt8186_afe_disable_cgs(struct mtk_base_afe *afe); int mt8186_afe_enable_clock(struct mtk_base_afe *afe); diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c index 41172a82103e..a868a04ed4e7 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c @@ -2848,10 +2848,6 @@ static int mt8186_afe_pcm_dev_probe(struct platform_device *pdev) return ret; } - ret = devm_add_action_or_reset(dev, mt8186_deinit_clock, (void *)afe); - if (ret) - return ret; - /* init memif */ afe->memif_32bit_supported = 0; afe->memif_size = MT8186_MEMIF_NUM; diff --git a/sound/soc/mediatek/mt8186/mt8186-audsys-clk.c b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.c index 578969ca91c8..5666be6b1bd2 100644 --- a/sound/soc/mediatek/mt8186/mt8186-audsys-clk.c +++ b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.c @@ -84,6 +84,29 @@ static const struct afe_gate aud_clks[CLK_AUD_NR_CLK] = { GATE_AUD2(CLK_AUD_ETDM_OUT1_BCLK, "aud_etdm_out1_bclk", "top_audio", 24), }; +static void mt8186_audsys_clk_unregister(void *data) +{ + struct mtk_base_afe *afe = data; + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct clk *clk; + struct clk_lookup *cl; + int i; + + if (!afe_priv) + return; + + for (i = 0; i < CLK_AUD_NR_CLK; i++) { + cl = afe_priv->lookup[i]; + if (!cl) + continue; + + clk = cl->clk; + clk_unregister_gate(clk); + + clkdev_drop(cl); + } +} + int mt8186_audsys_clk_register(struct mtk_base_afe *afe) { struct mt8186_afe_private *afe_priv = afe->platform_priv; @@ -124,27 +147,6 @@ int mt8186_audsys_clk_register(struct mtk_base_afe *afe) afe_priv->lookup[i] = cl; } - return 0; + return devm_add_action_or_reset(afe->dev, mt8186_audsys_clk_unregister, afe); } -void mt8186_audsys_clk_unregister(struct mtk_base_afe *afe) -{ - struct mt8186_afe_private *afe_priv = afe->platform_priv; - struct clk *clk; - struct clk_lookup *cl; - int i; - - if (!afe_priv) - return; - - for (i = 0; i < CLK_AUD_NR_CLK; i++) { - cl = afe_priv->lookup[i]; - if (!cl) - continue; - - clk = cl->clk; - clk_unregister_gate(clk); - - clkdev_drop(cl); - } -} diff --git a/sound/soc/mediatek/mt8186/mt8186-audsys-clk.h b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.h index b8d6a06e11e8..897a2914dc19 100644 --- a/sound/soc/mediatek/mt8186/mt8186-audsys-clk.h +++ b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.h @@ -10,6 +10,5 @@ #define _MT8186_AUDSYS_CLK_H_ int mt8186_audsys_clk_register(struct mtk_base_afe *afe); -void mt8186_audsys_clk_unregister(struct mtk_base_afe *afe); #endif -- 2.40.1.606.ga4b1b128d6-goog

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[PATCH 0/3] ASoC: SOF: Various runtime pm fixes, improvements
by Peter Ujfalusi 15 May '23

15 May '23

Three patch to correct error path PM runtime handling in few places. Regards, Peter --- Pierre-Louis Bossart (3): ASoC: SOF: debug: conditionally bump runtime_pm counter on exceptions ASoC: SOF: pcm: fix pm_runtime imbalance in error handling ASoC: SOF: sof-client-probes: fix pm_runtime imbalance in error handling sound/soc/sof/debug.c | 4 ++-- sound/soc/sof/pcm.c | 11 ++++++----- sound/soc/sof/sof-client-probes.c | 14 ++++++++------ 3 files changed, 16 insertions(+), 13 deletions(-) -- 2.40.1

2 4

[PATCH] ASoC: SOF: pm: save io region state in case of errors in resume
by Peter Ujfalusi 15 May '23

15 May '23

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> If there are failures in DSP runtime resume, the device state will not reach active and this makes it impossible e.g. to retrieve a possible DSP panic dump via "exception" debugfs node. If CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE=y is set, the data in cache is stale. If debugfs cache is not used, the region simply cannot be read. To allow debugging these scenarios, update the debugfs cache contents in resume error handler. User-space can then later retrieve DSP panic and other state via debugfs (requires SOF debugfs cache to be enabled in build). Reported-by: Curtis Malainey <cujomalainey(a)chromium.org> Link: https://github.com/thesofproject/linux/issues/4274 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Curtis Malainey <cujomalainey(a)chromium.org> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/pm.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 2fdbc53ca715..2b232442e84b 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -164,7 +164,7 @@ static int sof_resume(struct device *dev, bool runtime_resume) ret = tplg_ops->set_up_all_pipelines(sdev, false); if (ret < 0) { dev_err(sdev->dev, "Failed to restore pipeline after resume %d\n", ret); - return ret; + goto setup_fail; } } @@ -178,6 +178,18 @@ static int sof_resume(struct device *dev, bool runtime_resume) dev_err(sdev->dev, "ctx_restore IPC error during resume: %d\n", ret); } +setup_fail: +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) + if (ret < 0) { + /* + * Debugfs cannot be read in runtime suspend, so cache + * the contents upon failure. This allows to capture + * possible DSP coredump information. + */ + sof_cache_debugfs(sdev); + } +#endif + return ret; } -- 2.40.1

2 1

[PATCH] ASoC: SOF: topology: Fix logic for copying tuples
by Peter Ujfalusi 15 May '23

15 May '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Topology could have more instances of the tokens being searched for than the number of sets that need to be copied. Stop copying token after the limit of number of token instances has been reached. This worked before only by chance as we had allocated more size for the tuples array than the number of actual tokens being parsed. Fixes: 7006d20e5e9d ("ASoC: SOF: Introduce IPC3 ops") Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/topology.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index d3d536b0a8f5..b80566570512 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -586,6 +586,10 @@ static int sof_copy_tuples(struct snd_sof_dev *sdev, struct snd_soc_tplg_vendor_ if (*num_copied_tuples == tuples_size) return 0; } + + /* stop when we've found the required token instances */ + if (found == num_tokens * token_instance_num) + return 0; } /* next array */ -- 2.40.1

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[PATCH] ASoC: SOF: ipc3-topology: Make sure that only one cmd is sent in dai_config
by Peter Ujfalusi 15 May '23

15 May '23

The commands in sof_ipc_dai_config.flags are encoded as bits: 1 (bit0) - hw_params 2 (bit1) - hw_free 4 (bit2) - pause These are commands, they cannot be combined as one would assume, for example 3 (bit0 | bit1) is invalid. This can happen right at the second start of a stream as at the end of the first stream we set the hw_free command (bit1) and on the second start we would OR on top of it the hw_params (bit0). Fixes: b66bfc3a9810 ("ASoC: SOF: sof-audio: Fix broken early bclk feature for SSP") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/ipc3-topology.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index fc1eb8e2de2c..ba4ef290b634 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -2103,10 +2103,13 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * * For the case of PAUSE/HW_FREE, since there are no quirks, flags can be used as is. */ - if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) + if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) { + /* Clear stale command */ + config->flags &= ~SOF_DAI_CONFIG_FLAGS_CMD_MASK; config->flags |= flags; - else + } else { config->flags = flags; + } /* only send the IPC if the widget is set up in the DSP */ if (swidget->use_count > 0) { -- 2.40.1

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[PATCH] ASoC: MAINTAINERS: drop Krzysztof Kozlowski from Samsung audio
by Krzysztof Kozlowski 15 May '23

15 May '23

Remove Krzysztof Kozlowski from maintainer of Samsung SoC Audio drivers and change the status to maintenance (no one is reality being paid for looking at this). Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- MAINTAINERS | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index c7082004ac55..746ec43ff244 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -18581,10 +18581,9 @@ F: Documentation/admin-guide/LSM/SafeSetID.rst F: security/safesetid/ SAMSUNG AUDIO (ASoC) DRIVERS -M: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> M: Sylwester Nawrocki <s.nawrocki(a)samsung.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -S: Supported +S: Maintained B: mailto:linux-samsung-soc@vger.kernel.org F: Documentation/devicetree/bindings/sound/samsung* F: sound/soc/samsung/ -- 2.34.1

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[PATCH 1/2] pcm: hw: reinterpret the drain_silence setting
by Oswald Buddenhagen 15 May '23

15 May '23

It makes no sense to have a config option which determines the entire silencing time, because useful values strongly depend on the period size runtime setting. So instead we interpret the setting as an override for the "overhang" which is used to compensate FIFOs and IRQ delays. The reasonable worst case of this is determined by the hardware, so it makes sense to have this as a config option. In a next step, we may want to derive the default value from the declared FIFO size (whiche few drivers do) and the minimal period size (which is generally linked to the FIFO size). Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- src/pcm/pcm_hw.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 8ffebed9..ecc47a76 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -737,18 +737,19 @@ static int snd_pcm_hw_drain(snd_pcm_t *pcm) { snd_pcm_hw_t *hw = pcm->private_data; snd_pcm_sw_params_t sw_params; + snd_pcm_uframes_t silence_slack; snd_pcm_uframes_t silence_size; int err; if (pcm->stream != SND_PCM_STREAM_PLAYBACK) goto __skip_silence; if (hw->drain_silence == 0 || hw->perfect_drain) goto __skip_silence; snd_pcm_sw_params_current_no_lock(pcm, &sw_params); - if (hw->drain_silence > 0) { - silence_size = (pcm->rate * hw->drain_silence) / 1000; - goto __manual_silence; - } + if (hw->drain_silence > 0) + silence_slack = (pcm->rate * hw->drain_silence) / 1000; + else + silence_slack = pcm->rate / 10; /* 1/10th of second */ /* compute end silence size, align to period size + extra time */ if ((pcm->boundary % pcm->period_size) == 0) { silence_size = pcm->period_size - (*pcm->appl.ptr % pcm->period_size); @@ -761,8 +762,7 @@ static int snd_pcm_hw_drain(snd_pcm_t *pcm) */ silence_size = pcm->period_size; } - silence_size += pcm->rate / 10; /* 1/10th of second */ -__manual_silence: + silence_size += silence_slack; if (sw_params.silence_size < silence_size) { /* fill the silence soon as possible (in the bellow ioctl * or the next period wake up) -- 2.40.0.152.g15d061e6df

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[PATCH] ASoC: dt-bindings: nau8824: Convert to dtschema
by David Lin 15 May '23

15 May '23

Convert the NAU8824 audio CODEC bindings to DT schema. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8824.txt | 88 --------- .../bindings/sound/nuvoton,nau8824.yaml | 182 ++++++++++++++++++ 2 files changed, 182 insertions(+), 88 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nau8824.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml diff --git a/Documentation/devicetree/bindings/sound/nau8824.txt b/Documentation/devicetree/bindings/sound/nau8824.txt deleted file mode 100644 index e0058b97e49a..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8824.txt +++ /dev/null @@ -1,88 +0,0 @@ -Nuvoton NAU8824 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8824" - - - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). - -Optional properties: - - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. - - - nuvoton,vref-impedance: VREF Impedance selection - 0 - Open - 1 - 25 kOhm - 2 - 125 kOhm - 3 - 2.5 kOhm - - - nuvoton,micbias-voltage: Micbias voltage level. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,sar-threshold-num: Number of buttons supported - - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as - SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) - where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. - Refer datasheet section 10.2 for more information about threshold calculation. - - - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. - - - nuvoton,sar-voltage: Reference voltage for button impedance measurement. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,sar-compare-time: SAR compare time - 0 - 500 ns - 1 - 1 us - 2 - 2 us - 3 - 4 us - - - nuvoton,sar-sampling-time: SAR sampling time - 0 - 2 us - 1 - 4 us - 2 - 8 us - 3 - 16 us - - - nuvoton,short-key-debounce: Button short key press debounce time. - 0 - 30 ms - 1 - 50 ms - 2 - 100 ms - - - nuvoton,jack-eject-debounce: Jack ejection debounce time. - 0 - 0 ms - 1 - 1 ms - 2 - 10 ms - - -Example: - - headset: nau8824@1a { - compatible = "nuvoton,nau8824"; - reg = <0x1a>; - interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; - nuvoton,vref-impedance = <2>; - nuvoton,micbias-voltage = <6>; - // Setup 4 buttons impedance according to Android specification - nuvoton,sar-threshold-num = <4>; - nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; - nuvoton,sar-hysteresis = <0>; - nuvoton,sar-voltage = <6>; - nuvoton,sar-compare-time = <1>; - nuvoton,sar-sampling-time = <1>; - nuvoton,short-key-debounce = <0>; - nuvoton,jack-eject-debounce = <1>; - }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml new file mode 100644 index 000000000000..e03d7e2704b4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml @@ -0,0 +1,182 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8824.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8824 audio CODEC + +maintainers: + - John Hsu <KCHSU0(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nuvoton,nau8824 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-polarity: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + JKDET pin polarity. + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,vref-impedance: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + VREF Impedance selection. + enum: + - 0 # Open + - 1 # 25 kOhm + - 2 # 125 kOhm + - 3 # 2.5 kOhm + default: 2 + + nuvoton,micbias-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Micbias voltage level. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-threshold-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Number of buttons supported. + minimum: 1 + maximum: 4 + default: 4 + + nuvoton,sar-threshold: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Impedance threshold for each button. Array that contains up to 8 buttons + configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) where MICBIAS is + configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by + 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold + calculation. + minItems: 1 + maxItems: 4 + items: + minimum: 0 + maximum: 255 + + nuvoton,sar-hysteresis: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button impedance measurement hysteresis. + default: 0 + + nuvoton,sar-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Reference voltage for button impedance measurement. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-compare-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR compare time. + enum: + - 0 # 500ns + - 1 # 1us + - 2 # 2us + - 3 # 4us + default: 1 + + nuvoton,sar-sampling-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR sampling time. + enum: + - 0 # 2us + - 1 # 4us + - 2 # 8us + - 3 # 16us + default: 1 + + nuvoton,short-key-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button short key press debounce time. + enum: + - 0 # 30 ms + - 1 # 50 ms + - 2 # 100 ms + default: 0 + + nuvoton,jack-eject-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Jack ejection debounce time. + enum: + - 0 # 0 ms + - 1 # 1 ms + - 2 # 10 ms + default: 1 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + #sound-dai-cells = <0>; + compatible = "nuvoton,nau8824"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <38 IRQ_TYPE_LEVEL_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + nuvoton,sar-threshold-num = <4>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <0>; + nuvoton,sar-voltage = <6>; + nuvoton,sar-compare-time = <1>; + nuvoton,sar-sampling-time = <1>; + nuvoton,short-key-debounce = <0>; + nuvoton,jack-eject-debounce = <1>; + }; + }; -- 2.25.1

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[PATCH 1/3] docs: admin-guide: add valsa driver documentation
by Ivan Orlov 15 May '23

15 May '23

Add documentation for the new Virtual ALSA driver. It covers all possible usage cases: errors and delay injections, random and pattern-based data generation, playback and ioctl redefinition functionalities testing. We have a lot of different virtual media drivers, which can be used for testing of the userspace applications and media subsystem middle layer. However, all of them are aimed at testing the video functionality and simulating the video devices. For audio devices we have only snd-dummy module, which is good in simulating the correct behavior of an ALSA device. I decided to write a tool, which would help to test the userspace ALSA programs (and the PCM middle layer as well) under unusual circumstances to figure out how they would behave. So I came up with this Virtual ALSA Driver. This new Virtual ALSA Driver has several features which can be useful during the userspace ALSA applications testing/fuzzing, or testing/fuzzing of the PCM middle layer. Not all of them can be implemented using the existing virtual drivers (like dummy or loopback). Here is what can this driver do: - Simulate both capture and playback processes - Check the playback stream for containing the looped pattern - Generate random or pattern-based capture data - Inject delays into the playback and capturing processes - Inject errors during the PCM callbacks Also, this driver can check the playback stream for containing the predefined pattern, which is used in the corresponding selftest to check the PCM middle layer data transferring functionality. Additionally, this driver redefines the default RESET ioctl, and the selftest covers this PCM API functionality as well. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- Documentation/admin-guide/index.rst | 1 + Documentation/admin-guide/valsa.rst | 114 ++++++++++++++++++++++++++++ 2 files changed, 115 insertions(+) create mode 100644 Documentation/admin-guide/valsa.rst diff --git a/Documentation/admin-guide/index.rst b/Documentation/admin-guide/index.rst index 43ea35613dfc..328cc59275a1 100644 --- a/Documentation/admin-guide/index.rst +++ b/Documentation/admin-guide/index.rst @@ -131,6 +131,7 @@ configure specific aspects of kernel behavior to your liking. thunderbolt ufs unicode + valsa vga-softcursor video-output xfs diff --git a/Documentation/admin-guide/valsa.rst b/Documentation/admin-guide/valsa.rst new file mode 100644 index 000000000000..64ffc130fb4c --- /dev/null +++ b/Documentation/admin-guide/valsa.rst @@ -0,0 +1,114 @@ +.. SPDX-License-Identifier: GPL-2.0 + +The Virtual ALSA Driver +======================= + +The Virtual ALSA Driver emulates a generic ALSA device, and can be used for +testing/fuzzing of the userspace ALSA applications, as well as for testing/fuzzing of +the ALSA middle layer. Additionally, it can be used for simulating hard to reproduce +problems with PCM devices. + +What can this driver do? +~~~~~~~~~~~~~~~~~~~~~~~~ + +At this moment the driver can do the following things: + * Simulate both capture and playback processes + * Generate random or pattern-based capturing data + * Inject delays into the playback and capturing processes + * Inject errors during the PCM callbacks + +Also, this driver can check the playback stream for containing the +predefined pattern, which is used in the corresponding selftest (alsa/valsa-test.sh) +to check the PCM middle layer data transferring functionality. Additionally, this +driver redefines the default RESET ioctl, and the selftest covers this PCM +API functionality as well. + +Configuration +------------- + +The driver has several parameters besides the common ALSA module parameters: + + * fill_mode (bool) - Buffer fill mode (see below) + * inject_delay (int) + * inject_hwpars_err (bool) + * inject_prepare_err (bool) + * inject_trigger_err (bool) + + +Capture Data Generation +----------------------- + +The driver has two modes of data generation: the first (0 in the fill_mode parameter) +means random data generation, the second (1 in the fill_mode) - pattern-based +data generation. Let's look at the second mode. + +First of all, you may want to specify the pattern for data generation. You can do it +by writing the pattern to the debugfs file (/sys/kernel/debug/valsa/fill_pattern). +Like that: + +.. code-block:: bash + + echo -n mycoolpattern > /sys/kernel/debug/valsa/fill_pattern + +After this, every capture action performed on the 'valsa' device will return +'mycoolpatternmycoolpatternmycoolpatternmy...' in the capturing buffer. + +The pattern itself can be up to 4096 bytes long. + +Delay injection +--------------- + +The driver has 'inject_delay' parameter, which has very self-descriptive name and +can be used for time delay/speedup simulations. The parameter has integer type, and +it means the delay added between module's internal timer ticks. + +If the 'inject_delay' value is positive, the buffer will be filled slower, if it is +negative - faster. You can try it yourself by starting a recording in any +audiorecording application (like Audacity) and selecting the 'valsa' device as a +source. + +This parameter can be also used for generating a huge amount of sound data in a very +short period of time (with the negative 'inject_delay' value). + +Errors injection +---------------- + +This module can be used for injecting errors into the PCM communication process. This +action can help you to figure out how the userspace ALSA program behaves under unusual +circumstances. + +For example, you can make all 'hw_params' PCM callback calls return EBUSY error by +writing '1' to the 'inject_hwpars_err' module parameter: + +.. code-block:: bash + + echo 1 > /sys/module/snd_valsa/parameters/inject_hwpars_err + +Errors can be injected into the following PCM callbacks: + + * hw_params (EBUSY) + * prepare (EINVAL) + * trigger (EINVAL) + + +Playback test +------------- + +This driver can be also used for the playback functionality testing - every time you +write the playback data to the 'valsa' PCM device and close it, the driver checks the +buffer for containing the looped pattern (which is specified in the fill_pattern +debugfs file). If the playback buffer content represents the looped pattern, 'pc_test' +debugfs entry is set into '1'. Otherwise, the driver sets it to '0'. + +ioctl redefinition test +----------------------- + +The driver redefines the 'reset' ioctl, which is default for all PCM devices. To test +this functionality, we can trigger the reset ioctl and check the 'ioctl_test' debugfs +entry: + +.. code-block:: bash + + cat /sys/kernel/debug/valsa/ioctl_test + +If the ioctl is triggered successfully, this file will contain '1', and '0' otherwise. -- 2.34.1

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