April 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: rt715-sdca: fix clock stop prepare timeout issue
by Jack Yu 21 Apr '21

21 Apr '21

The Clock_Stop_Not_Finished bit is currently not cleared during the clock stop sequence,which results in the clock stop sequence timing out after 300ms. This is root-caused to a missing register configuration impacting the transition between the bus clock and an internal clock. This problem was exposed by cleanups in the SoundWire bus code, where previously the ClockStopPrepare was mistakenly skipped when using ClockStopMode0. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715-sdca-sdw.c | 1 + sound/soc/codecs/rt715-sdca-sdw.h | 1 + sound/soc/codecs/rt715-sdca.c | 3 +++ sound/soc/codecs/rt715-sdca.h | 1 + 4 files changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index 1350798406f0..47a8f04d7100 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -70,6 +70,7 @@ static bool rt715_sdca_mbq_readable_register(struct device *dev, unsigned int re case 0x2000036: case 0x2000037: case 0x2000039: + case 0x2000044: case 0x6100000: return true; default: diff --git a/sound/soc/codecs/rt715-sdca-sdw.h b/sound/soc/codecs/rt715-sdca-sdw.h index cd365bb60747..0cbc14844f8c 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.h +++ b/sound/soc/codecs/rt715-sdca-sdw.h @@ -113,6 +113,7 @@ static const struct reg_default rt715_mbq_reg_defaults_sdca[] = { { 0x2000036, 0x0000 }, { 0x2000037, 0x0000 }, { 0x2000039, 0xaa81 }, + { 0x2000044, 0x0202 }, { 0x6100000, 0x0100 }, { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC8_9_VOL, RT715_SDCA_FU_VOL_CTRL, CH_01), 0x00 }, diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index 7db76c19e048..936e3061ca1e 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -1054,6 +1054,9 @@ int rt715_sdca_io_init(struct device *dev, struct sdw_slave *slave) rt715_sdca_index_update_bits(rt715, RT715_VENDOR_REG, RT715_REV_1, 0x40, 0x40); } + /* DFLL Calibration trigger */ + rt715_sdca_index_update_bits(rt715, RT715_VENDOR_REG, + RT715_DFLL_VAD, 0x1, 0x1); /* trigger mode = VAD enable */ regmap_write(rt715->regmap, SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_SMPU_TRIG_ST_EN, diff --git a/sound/soc/codecs/rt715-sdca.h b/sound/soc/codecs/rt715-sdca.h index 85ce4d95e5eb..1c7fbff2ce1e 100644 --- a/sound/soc/codecs/rt715-sdca.h +++ b/sound/soc/codecs/rt715-sdca.h @@ -81,6 +81,7 @@ struct rt715_sdca_kcontrol_private { #define RT715_AD_FUNC_EN 0x36 #define RT715_REV_1 0x37 #define RT715_SDW_INPUT_SEL 0x39 +#define RT715_DFLL_VAD 0x44 #define RT715_EXT_DMIC_CLK_CTRL2 0x54 /* Index (NID:61h) */ -- 2.29.0

1 0

Question about soc_pcm_pointer()
by Kuninori Morimoto 21 Apr '21

21 Apr '21

Hi ALSA ML I noticed that current soc-pcm.c :: soc_pcm_pointer() is adding both CPU-DAI's max delay (= A) and Codec-DAI's max delay (= B). static snd_pcm_uframes_t soc_pcm_pointer(...) { ... ^ for_each_rtd_cpu_dais(rtd, i, cpu_dai) (A) cpu_delay = max(cpu_delay, ...); v delay += cpu_delay; ^ for_each_rtd_codec_dais(rtd, i, codec_dai) (B) codec_delay = max(codec_delay, ...); v delay += codec_delay; runtime->delay = delay; ... } But I guess All-DAI's max delay is very enough for it. What do you think ? Thank you for your help !! Best regards --- Kuninori Morimoto

2 2

[PATCH] ASoC: ak4458: enable daisy chain
by Shengjiu Wang 20 Apr '21

20 Apr '21

From: Viorel Suman <viorel.suman(a)nxp.com> Enable Daisy Chain if in TDM mode and the number of played channels is bigger than the maximum supported number of channels. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/ak4458.c | 47 ++++++++++++++++++++++++++------------- sound/soc/codecs/ak4458.h | 1 + 2 files changed, 32 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 3088128816bb..29eb78702bf3 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -306,6 +306,20 @@ static const struct snd_soc_dapm_route ak4497_intercon[] = { }; +static int ak4458_get_tdm_mode(struct ak4458_priv *ak4458) +{ + switch (ak4458->slots * ak4458->slot_width) { + case 128: + return 1; + case 256: + return 2; + case 512: + return 3; + default: + return 0; + } +} + static int ak4458_rstn_control(struct snd_soc_component *component, int bit) { int ret; @@ -333,13 +347,16 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); int pcm_width = max(params_physical_width(params), ak4458->slot_width); - u8 format, dsdsel0, dsdsel1; - int nfs1, dsd_bclk, ret; + u8 format, dsdsel0, dsdsel1, dchn; + int nfs1, dsd_bclk, ret, channels, channels_max; nfs1 = params_rate(params); ak4458->fs = nfs1; /* calculate bit clock */ + channels = params_channels(params); + channels_max = dai->driver->playback.channels_max; + switch (params_format(params)) { case SNDRV_PCM_FORMAT_DSD_U8: case SNDRV_PCM_FORMAT_DSD_U16_LE: @@ -419,6 +436,17 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, AK4458_00_CONTROL1, AK4458_DIF_MASK, format); + /* + * Enable/disable Daisy Chain if in TDM mode and the number of played + * channels is bigger than the maximum supported number of channels + */ + dchn = ak4458_get_tdm_mode(ak4458) && + (ak4458->fmt == SND_SOC_DAIFMT_DSP_B) && + (channels > channels_max) ? AK4458_DCHAIN_MASK : 0; + + snd_soc_component_update_bits(component, AK4458_0B_CONTROL7, + AK4458_DCHAIN_MASK, dchn); + ret = ak4458_rstn_control(component, 0); if (ret) return ret; @@ -519,20 +547,7 @@ static int ak4458_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, ak4458->slots = slots; ak4458->slot_width = slot_width; - switch (slots * slot_width) { - case 128: - mode = AK4458_MODE_TDM128; - break; - case 256: - mode = AK4458_MODE_TDM256; - break; - case 512: - mode = AK4458_MODE_TDM512; - break; - default: - mode = AK4458_MODE_NORMAL; - break; - } + mode = ak4458_get_tdm_mode(ak4458) << AK4458_MODE_SHIFT; snd_soc_component_update_bits(component, AK4458_0A_CONTROL6, AK4458_MODE_MASK, diff --git a/sound/soc/codecs/ak4458.h b/sound/soc/codecs/ak4458.h index 9548c5d78621..9ad869575f8d 100644 --- a/sound/soc/codecs/ak4458.h +++ b/sound/soc/codecs/ak4458.h @@ -82,6 +82,7 @@ * */ #define AK4458_ATS_SHIFT 6 #define AK4458_ATS_MASK GENMASK(7, 6) +#define AK4458_DCHAIN_MASK (0x1 << 1) #define AK4458_DSDSEL_MASK (0x1 << 0) #define AK4458_DP_MASK (0x1 << 7) -- 2.27.0

2 1

[PATCH] ASoC: simple-card-utils: Allocate link info structure on heap
by Thierry Reding 20 Apr '21

20 Apr '21

From: Thierry Reding <treding(a)nvidia.com> struct link_info can grow fairly large and may cause the stack frame size to be exceeded when allocated on the stack. Some architectures such as 32-bit ARM, RISC-V or PowerPC have small stack frames where this causes a compiler warning, so allocate these structures on the heap instead of the stack. Fixes: 343e55e71877 ("ASoC: simple-card-utils: Increase maximum number of links to 128") Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/soc/generic/audio-graph-card.c | 18 +++++++++++------- sound/soc/generic/simple-card.c | 23 +++++++++++++---------- 2 files changed, 24 insertions(+), 17 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index c7369beee805..e45a560aa9b0 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -548,21 +548,24 @@ static int graph_get_dais_count(struct asoc_simple_priv *priv, int audio_graph_parse_of(struct asoc_simple_priv *priv, struct device *dev) { struct snd_soc_card *card = simple_priv_to_card(priv); - struct link_info li; + struct link_info *li; int ret; + li = devm_kzalloc(dev, sizeof(*li), GFP_KERNEL); + if (!li) + return -ENOMEM; + card->owner = THIS_MODULE; card->dev = dev; - memset(&li, 0, sizeof(li)); - ret = graph_get_dais_count(priv, &li); + ret = graph_get_dais_count(priv, li); if (ret < 0) return ret; - if (!li.link) + if (!li->link) return -EINVAL; - ret = asoc_simple_init_priv(priv, &li); + ret = asoc_simple_init_priv(priv, li); if (ret < 0) return ret; @@ -581,8 +584,8 @@ int audio_graph_parse_of(struct asoc_simple_priv *priv, struct device *dev) if (ret < 0) return ret; - memset(&li, 0, sizeof(li)); - ret = graph_for_each_link(priv, &li, + memset(li, 0, sizeof(*li)); + ret = graph_for_each_link(priv, li, graph_dai_link_of, graph_dai_link_of_dpcm); if (ret < 0) @@ -600,6 +603,7 @@ int audio_graph_parse_of(struct asoc_simple_priv *priv, struct device *dev) if (ret < 0) goto err; + devm_kfree(dev, li); return 0; err: diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 8b9964d25757..ca27cb9ff9e1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -442,10 +442,9 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, return ret; } -static int simple_parse_of(struct asoc_simple_priv *priv) +static int simple_parse_of(struct asoc_simple_priv *priv, struct link_info *li) { struct snd_soc_card *card = simple_priv_to_card(priv); - struct link_info li; int ret; ret = asoc_simple_parse_widgets(card, PREFIX); @@ -461,8 +460,8 @@ static int simple_parse_of(struct asoc_simple_priv *priv) return ret; /* Single/Muti DAI link(s) & New style of DT node */ - memset(&li, 0, sizeof(li)); - ret = simple_for_each_link(priv, &li, + memset(li, 0, sizeof(*li)); + ret = simple_for_each_link(priv, li, simple_dai_link_of, simple_dai_link_of_dpcm); if (ret < 0) @@ -612,7 +611,7 @@ static int asoc_simple_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; struct device_node *np = dev->of_node; struct snd_soc_card *card; - struct link_info li; + struct link_info *li; int ret; /* Allocate the private data and the DAI link array */ @@ -625,21 +624,24 @@ static int asoc_simple_probe(struct platform_device *pdev) card->dev = dev; card->probe = simple_soc_probe; - memset(&li, 0, sizeof(li)); - ret = simple_get_dais_count(priv, &li); + li = devm_kzalloc(dev, sizeof(*li), GFP_KERNEL); + if (!li) + return -ENOMEM; + + ret = simple_get_dais_count(priv, li); if (ret < 0) return ret; - if (!li.link) + if (!li->link) return -EINVAL; - ret = asoc_simple_init_priv(priv, &li); + ret = asoc_simple_init_priv(priv, li); if (ret < 0) return ret; if (np && of_device_is_available(np)) { - ret = simple_parse_of(priv); + ret = simple_parse_of(priv, li); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); @@ -698,6 +700,7 @@ static int asoc_simple_probe(struct platform_device *pdev) if (ret < 0) goto err; + devm_kfree(dev, li); return 0; err: asoc_simple_clean_reference(card); -- 2.30.2

3 2

[PATCH] ASoC: rt1015p: add support on format S32_LE
by Jack Yu 20 Apr '21

20 Apr '21

Add support on format S32_LE for rt1015p. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015p.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1015p.c b/sound/soc/codecs/rt1015p.c index 39f4c4ed2ab8..40f2063aefbe 100644 --- a/sound/soc/codecs/rt1015p.c +++ b/sound/soc/codecs/rt1015p.c @@ -96,7 +96,8 @@ static struct snd_soc_dai_driver rt1015p_dai_driver = { .name = "HiFi", .playback = { .stream_name = "HiFi Playback", - .formats = SNDRV_PCM_FMTBIT_S24, + .formats = SNDRV_PCM_FMTBIT_S24 | + SNDRV_PCM_FMTBIT_S32, .rates = SNDRV_PCM_RATE_48000, .channels_min = 1, .channels_max = 2, -- 2.29.0

2 1

[PATCH] ALSA: usb: midi: don't return -ENOMEM when usb_urb_ep_type_check fails
by Colin King 20 Apr '21

20 Apr '21

From: Colin Ian King <colin.king(a)canonical.com> Currently when the call to usb_urb_ep_type_check fails (returning -EINVAL) the error return path returns -ENOMEM via the exit label "error". Other uses of the same error exit label set the err variable to -ENOMEM but this is not being used. I believe the original intent was for the error exit path to return the value in err rather than the hard coded -ENOMEM, so return this rather than the hard coded -ENOMEM. Addresses-Coverity: ("Unused value") Fixes: 738d9edcfd44 ("ALSA: usb-audio: Add sanity checks for invalid EPs") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/midi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 9efda4b06acb..a10ac75969a8 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1316,7 +1316,7 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi *umidi, error: snd_usbmidi_in_endpoint_delete(ep); - return -ENOMEM; + return err; } /* -- 2.30.2

2 1

[v6 3/3] dt-bindings: Add Rockchip rk817 audio CODEC support
by Chris Morgan 20 Apr '21

20 Apr '21

Create dt-binding documentation to document rk817 codec. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- Changes in v6: - Included additional project maintainers for correct subsystems. - Removed unneeded compatible from DT documentation. - Removed binding update for Odroid Go Advance (will do in seperate series). Changes in v5: - Move register definitions from rk817_codec.h to main rk808.h register definitions. - Add volatile register for codec bits. - Add default values for codec bits. - Removed of_compatible from mtd driver (not necessary). - Switched to using parent regmap instead of private regmap for codec. Changes in v4: - Created set_pll() call. - Created user visible gain control in mic. - Check for return value of clk_prepare_enable(). - Removed duplicate clk_prepare_enable(). - Split DT documentation to separate commit. Changes in v3: - Use DAPM macros to set audio path. - Updated devicetree binding (as every rk817 has this codec chip). - Changed documentation to yaml format. - Split MFD changes to separate commit. Changes in v2: - Fixed audio path registers to solve some bugs. .../bindings/sound/rockchip,rk817-codec.yaml | 57 +++++++++++++++++++ 1 file changed, 57 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml new file mode 100644 index 000000000000..0059ef54d6b5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml @@ -0,0 +1,57 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-rk817.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk817 audio codec + +description: + The rk817 codec is an I2C codec integrated with every Rockchip + rk817 PMIC MFD. + +properties: + + "#sound-dai-cells": + const: 1 + + clocks: + maxItems: 1 + + clock-names: + const: "mclk" + + mic-in-differential: + description: the microphone is in differential mode. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - clocks + - clock-names + +additionalProperties: false + +examples: + - | +rk817: pmic@20 { + compatible = "rockchip,rk817"; + reg = <0x20>; + interrupt-parent = <&gpio0>; + interrupts = <7 IRQ_TYPE_LEVEL_LOW>; + pinctrl-names = "default"; + pinctrl-0 = <&pmic_int>, <&i2s_8ch_mclk>; + #sound-dai-cells = <0>; + clocks = <&cru SCLK_I2S_8CH_OUT>; + clock-names = "mclk"; + + ........ + + rk817_codec: codec { + mic-in-differential; + status = "okay"; + }; + + ........ + +}; -- 2.25.1

2 2

[v6 1/3] mfd: Add Rockchip rk817 audio CODEC support
by Chris Morgan 20 Apr '21

20 Apr '21

Add rk817 codec support cell to rk808 mfd driver. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- Changes in v6: - Included additional project maintainers for correct subsystems. - Removed unneeded compatible from DT documentation. - Removed binding update for Odroid Go Advance (will do in seperate series). Changes in v5: - Move register definitions from rk817_codec.h to main rk808.h register definitions. - Add volatile register for codec bits. - Add default values for codec bits. - Removed of_compatible from mtd driver (not necessary). - Switched to using parent regmap instead of private regmap for codec. Changes in v4: - Created set_pll() call. - Created user visible gain control in mic. - Check for return value of clk_prepare_enable(). - Removed duplicate clk_prepare_enable(). - Split DT documentation to separate commit. Changes in v3: - Use DAPM macros to set audio path. - Updated devicetree binding (as every rk817 has this codec chip). - Changed documentation to yaml format. - Split MFD changes to separate commit. Changes in v2: - Fixed audio path registers to solve some bugs. drivers/mfd/rk808.c | 89 +++++++++++++++++++++++++++++++++++++++ include/linux/mfd/rk808.h | 81 +++++++++++++++++++++++++++++++++++ 2 files changed, 170 insertions(+) diff --git a/drivers/mfd/rk808.c b/drivers/mfd/rk808.c index ad923dd4e007..b92f4c6f503e 100644 --- a/drivers/mfd/rk808.c +++ b/drivers/mfd/rk808.c @@ -65,6 +65,9 @@ static bool rk817_is_volatile_reg(struct device *dev, unsigned int reg) switch (reg) { case RK817_SECONDS_REG ... RK817_WEEKS_REG: case RK817_RTC_STATUS_REG: +#ifdef CONFIG_SND_SOC_RK817 + case RK817_CODEC_DTOP_LPT_SRST: +#endif case RK817_INT_STS_REG0: case RK817_INT_STS_REG1: case RK817_INT_STS_REG2: @@ -163,6 +166,11 @@ static const struct mfd_cell rk817s[] = { .num_resources = ARRAY_SIZE(rk817_rtc_resources), .resources = &rk817_rtc_resources[0], }, +#ifdef CONFIG_SND_SOC_RK817 + { + .name = "rk817-codec", + }, +#endif }; static const struct mfd_cell rk818s[] = { @@ -201,6 +209,87 @@ static const struct rk808_reg_data rk808_pre_init_reg[] = { static const struct rk808_reg_data rk817_pre_init_reg[] = { {RK817_RTC_CTRL_REG, RTC_STOP, RTC_STOP}, +#ifdef CONFIG_SND_SOC_RK817 + /* Codec specific registers */ + { RK817_CODEC_DTOP_VUCTL, MASK_ALL, 0x03 }, + { RK817_CODEC_DTOP_VUCTIME, MASK_ALL, 0x00 }, + { RK817_CODEC_DTOP_LPT_SRST, MASK_ALL, 0x00 }, + { RK817_CODEC_DTOP_DIGEN_CLKE, MASK_ALL, 0x00 }, + /* from vendor driver, CODEC_AREF_RTCFG0 not defined in data sheet */ + { RK817_CODEC_AREF_RTCFG0, MASK_ALL, 0x00 }, + { RK817_CODEC_AREF_RTCFG1, MASK_ALL, 0x06 }, + { RK817_CODEC_AADC_CFG0, MASK_ALL, 0xc8 }, + /* from vendor driver, CODEC_AADC_CFG1 not defined in data sheet */ + { RK817_CODEC_AADC_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_VOLL, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_VOLR, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_SR_ACL0, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_ALC1, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_ALC2, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_NG, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_HPF, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_RVOLL, MASK_ALL, 0xff }, + { RK817_CODEC_DADC_RVOLR, MASK_ALL, 0xff }, + { RK817_CODEC_AMIC_CFG0, MASK_ALL, 0x70 }, + { RK817_CODEC_AMIC_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_PGA_GAIN, MASK_ALL, 0x66 }, + { RK817_CODEC_DMIC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG2, MASK_ALL, 0x00 }, + /* from vendor driver, CODEC_ADAC_CFG0 not defined in data sheet */ + { RK817_CODEC_ADAC_CFG0, MASK_ALL, 0x00 }, + { RK817_CODEC_ADAC_CFG1, MASK_ALL, 0x07 }, + { RK817_CODEC_DDAC_POPD_DACST, MASK_ALL, 0x82 }, + { RK817_CODEC_DDAC_VOLL, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_VOLR, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_SR_LMT0, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_MUTE_MIXCTL, MASK_ALL, 0xa0 }, + { RK817_CODEC_DDAC_RVOLL, MASK_ALL, 0xff }, + { RK817_CODEC_DADC_RVOLR, MASK_ALL, 0xff }, + { RK817_CODEC_AMIC_CFG0, MASK_ALL, 0x70 }, + { RK817_CODEC_AMIC_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_PGA_GAIN, MASK_ALL, 0x66 }, + { RK817_CODEC_DMIC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG2, MASK_ALL, 0x00 }, + /* from vendor driver, CODEC_ADAC_CFG0 not defined in data sheet */ + { RK817_CODEC_ADAC_CFG0, MASK_ALL, 0x00 }, + { RK817_CODEC_ADAC_CFG1, MASK_ALL, 0x07 }, + { RK817_CODEC_DDAC_POPD_DACST, MASK_ALL, 0x82 }, + { RK817_CODEC_DDAC_VOLL, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_VOLR, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_SR_LMT0, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_MUTE_MIXCTL, MASK_ALL, 0xa0 }, + { RK817_CODEC_DDAC_RVOLL, MASK_ALL, 0xff }, + { RK817_CODEC_DDAC_RVOLR, MASK_ALL, 0xff }, + { RK817_CODEC_AHP_ANTI0, MASK_ALL, 0x00 }, + { RK817_CODEC_AHP_ANTI1, MASK_ALL, 0x00 }, + { RK817_CODEC_AHP_CFG0, MASK_ALL, 0xe0 }, + { RK817_CODEC_AHP_CFG1, MASK_ALL, 0x1f }, + { RK817_CODEC_AHP_CP, MASK_ALL, 0x09 }, + { RK817_CODEC_ACLASSD_CFG1, MASK_ALL, 0x69 }, + { RK817_CODEC_ACLASSD_CFG2, MASK_ALL, 0x44 }, + { RK817_CODEC_APLL_CFG0, MASK_ALL, 0x04 }, + { RK817_CODEC_APLL_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_APLL_CFG2, MASK_ALL, 0x30 }, + { RK817_CODEC_APLL_CFG3, MASK_ALL, 0x19 }, + { RK817_CODEC_APLL_CFG4, MASK_ALL, 0x65 }, + { RK817_CODEC_APLL_CFG5, MASK_ALL, 0x01 }, + { RK817_CODEC_DI2S_CKM, MASK_ALL, 0x01 }, + { RK817_CODEC_DI2S_RSD, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_RXCR1, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_RXCR2, MASK_ALL, 0x17 }, + { RK817_CODEC_DI2S_RXCMD_TSD, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_TXCR1, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_TXCR2, MASK_ALL, 0x17 }, + { RK817_CODEC_DI2S_TXCR3_TXCMD, MASK_ALL, 0x00 }, +#endif {RK817_GPIO_INT_CFG, RK817_INT_POL_MSK, RK817_INT_POL_L}, {RK817_SYS_CFG(1), RK817_HOTDIE_TEMP_MSK | RK817_TSD_TEMP_MSK, RK817_HOTDIE_105 | RK817_TSD_140}, diff --git a/include/linux/mfd/rk808.h b/include/linux/mfd/rk808.h index e07f6e61cd38..a96e6d43ca06 100644 --- a/include/linux/mfd/rk808.h +++ b/include/linux/mfd/rk808.h @@ -437,6 +437,87 @@ enum rk809_reg_id { #define RK817_RTC_COMP_LSB_REG 0x10 #define RK817_RTC_COMP_MSB_REG 0x11 +/* RK817 Codec Registers */ +#define RK817_CODEC_DTOP_VUCTL 0x12 +#define RK817_CODEC_DTOP_VUCTIME 0x13 +#define RK817_CODEC_DTOP_LPT_SRST 0x14 +#define RK817_CODEC_DTOP_DIGEN_CLKE 0x15 +#define RK817_CODEC_AREF_RTCFG0 0x16 +#define RK817_CODEC_AREF_RTCFG1 0x17 +#define RK817_CODEC_AADC_CFG0 0x18 +#define RK817_CODEC_AADC_CFG1 0x19 +#define RK817_CODEC_DADC_VOLL 0x1a +#define RK817_CODEC_DADC_VOLR 0x1b +#define RK817_CODEC_DADC_SR_ACL0 0x1e +#define RK817_CODEC_DADC_ALC1 0x1f +#define RK817_CODEC_DADC_ALC2 0x20 +#define RK817_CODEC_DADC_NG 0x21 +#define RK817_CODEC_DADC_HPF 0x22 +#define RK817_CODEC_DADC_RVOLL 0x23 +#define RK817_CODEC_DADC_RVOLR 0x24 +#define RK817_CODEC_AMIC_CFG0 0x27 +#define RK817_CODEC_AMIC_CFG1 0x28 +#define RK817_CODEC_DMIC_PGA_GAIN 0x29 +#define RK817_CODEC_DMIC_LMT1 0x2a +#define RK817_CODEC_DMIC_LMT2 0x2b +#define RK817_CODEC_DMIC_NG1 0x2c +#define RK817_CODEC_DMIC_NG2 0x2d +#define RK817_CODEC_ADAC_CFG0 0x2e +#define RK817_CODEC_ADAC_CFG1 0x2f +#define RK817_CODEC_DDAC_POPD_DACST 0x30 +#define RK817_CODEC_DDAC_VOLL 0x31 +#define RK817_CODEC_DDAC_VOLR 0x32 +#define RK817_CODEC_DDAC_SR_LMT0 0x35 +#define RK817_CODEC_DDAC_LMT1 0x36 +#define RK817_CODEC_DDAC_LMT2 0x37 +#define RK817_CODEC_DDAC_MUTE_MIXCTL 0x38 +#define RK817_CODEC_DDAC_RVOLL 0x39 +#define RK817_CODEC_DDAC_RVOLR 0x3a +#define RK817_CODEC_AHP_ANTI0 0x3b +#define RK817_CODEC_AHP_ANTI1 0x3c +#define RK817_CODEC_AHP_CFG0 0x3d +#define RK817_CODEC_AHP_CFG1 0x3e +#define RK817_CODEC_AHP_CP 0x3f +#define RK817_CODEC_ACLASSD_CFG1 0x40 +#define RK817_CODEC_ACLASSD_CFG2 0x41 +#define RK817_CODEC_APLL_CFG0 0x42 +#define RK817_CODEC_APLL_CFG1 0x43 +#define RK817_CODEC_APLL_CFG2 0x44 +#define RK817_CODEC_APLL_CFG3 0x45 +#define RK817_CODEC_APLL_CFG4 0x46 +#define RK817_CODEC_APLL_CFG5 0x47 +#define RK817_CODEC_DI2S_CKM 0x48 +#define RK817_CODEC_DI2S_RSD 0x49 +#define RK817_CODEC_DI2S_RXCR1 0x4a +#define RK817_CODEC_DI2S_RXCR2 0x4b +#define RK817_CODEC_DI2S_RXCMD_TSD 0x4c +#define RK817_CODEC_DI2S_TXCR1 0x4d +#define RK817_CODEC_DI2S_TXCR2 0x4e +#define RK817_CODEC_DI2S_TXCR3_TXCMD 0x4f + +/* RK817_CODEC_DI2S_CKM */ +#define RK817_I2S_MODE_MASK (0x1 << 0) +#define RK817_I2S_MODE_MST (0x1 << 0) +#define RK817_I2S_MODE_SLV (0x0 << 0) + +/* RK817_CODEC_DDAC_MUTE_MIXCTL */ +#define DACMT_MASK (0x1 << 0) +#define DACMT_ENABLE (0x1 << 0) +#define DACMT_DISABLE (0x0 << 0) + +/* RK817_CODEC_DI2S_RXCR2 */ +#define VDW_RX_24BITS (0x17) +#define VDW_RX_16BITS (0x0f) + +/* RK817_CODEC_DI2S_TXCR2 */ +#define VDW_TX_24BITS (0x17) +#define VDW_TX_16BITS (0x0f) + +/* RK817_CODEC_AMIC_CFG0 */ +#define MIC_DIFF_MASK (0x1 << 7) +#define MIC_DIFF_DIS (0x0 << 7) +#define MIC_DIFF_EN (0x1 << 7) + #define RK817_POWER_EN_REG(i) (0xb1 + (i)) #define RK817_POWER_SLP_EN_REG(i) (0xb5 + (i)) -- 2.25.1

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[PATCH] ALSA: hda/realtek: Enable mute/micmute LEDs and limit mic boost on EliteBook 845 G8
by Kai-Heng Feng 20 Apr '21

20 Apr '21

On HP EliteBook 845 G8, the audio LEDs can be enabled by ALC285_FIXUP_HP_MUTE_LED. So use it accordingly. In addition to that, the mic captures lots of noises, so also limits the mic boost. The quality of capture audio becomes crystal clear after limiting the mic boost. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7544b77d3f7..c71f1f2b6a8c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6427,6 +6427,7 @@ enum { ALC282_FIXUP_ACER_DISABLE_LINEOUT, ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, ALC256_FIXUP_ACER_HEADSET_MIC, + ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -7901,6 +7902,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC285_FIXUP_HP_MUTE_LED, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8079,6 +8086,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.30.2

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[PATCH v6] sound: rawmidi: Add framing mode
by David Henningsson 20 Apr '21

20 Apr '21

This commit adds a new framing mode that frames all MIDI data into 32-byte frames with a timestamp. The main benefit is that we can get accurate timestamps even if userspace wakeup and processing is not immediate. Testing on a Celeron N3150 with this mode has a max jitter of 2.8 ms, compared to the in-kernel seq implementation which has a max jitter of 5 ms during idle and much worse when running scheduler stress tests in parallel. Signed-off-by: David Henningsson <coding(a)diwic.se> --- Changes since v5: Added realtime clock and changed params struct according to Jaroslav's wishes. This version of the patch has been compile tested only. include/sound/rawmidi.h | 2 + include/uapi/sound/asound.h | 30 ++++++++++++- sound/core/rawmidi.c | 86 ++++++++++++++++++++++++++++++++++++- sound/core/rawmidi_compat.c | 4 +- 4 files changed, 118 insertions(+), 4 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 334842daa904..989e1517332d 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -81,6 +81,8 @@ struct snd_rawmidi_substream { bool opened; /* open flag */ bool append; /* append flag (merge more streams) */ bool active_sensing; /* send active sensing when close */ + unsigned int framing; /* whether to frame input data */ + unsigned int clock_type; /* clock source to use for input framing */ int use_count; /* use counter (for output) */ size_t bytes; struct snd_rawmidi *rmidi; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 535a7229e1d9..773a00c0a1d8 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -710,7 +710,7 @@ enum { * Raw MIDI section - /dev/snd/midi?? */ -#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 1) +#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 2) enum { SNDRV_RAWMIDI_STREAM_OUTPUT = 0, @@ -736,12 +736,38 @@ struct snd_rawmidi_info { unsigned char reserved[64]; /* reserved for future use */ }; +#define SNDRV_RAWMIDI_MODE_FRAMING_MASK (7<<0) +#define SNDRV_RAWMIDI_MODE_FRAMING_SHIFT 0 +#define SNDRV_RAWMIDI_MODE_FRAMING_NONE (0<<0) +#define SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP (1<<0) +#define SNDRV_RAWMIDI_MODE_CLOCK_MASK (7<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_SHIFT 3 +#define SNDRV_RAWMIDI_MODE_CLOCK_NONE (0<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_REALTIME (1<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC (2<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC_RAW (3<<3) + +#define SNDRV_RAWMIDI_FRAMING_DATA_LENGTH 16 + +struct snd_rawmidi_framing_tstamp { + /* For now, frame_type is always 0. Midi 2.0 is expected to add new + * types here. Applications are expected to skip unknown frame types. + */ + u8 frame_type; + u8 length; /* number of valid bytes in data field */ + u8 reserved[2]; + u32 tv_nsec; /* nanoseconds */ + u64 tv_sec; /* seconds */ + u8 data[SNDRV_RAWMIDI_FRAMING_DATA_LENGTH]; +} __attribute__((packed)); + struct snd_rawmidi_params { int stream; size_t buffer_size; /* queue size in bytes */ size_t avail_min; /* minimum avail bytes for wakeup */ unsigned int no_active_sensing: 1; /* do not send active sensing byte in close() */ - unsigned char reserved[16]; /* reserved for future use */ + unsigned int mode; /* For input data only, frame incoming data */ + unsigned char reserved[12]; /* reserved for future use */ }; #ifndef __KERNEL__ diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index aca00af93afe..5d5f4363e887 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -680,9 +680,12 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, bool is_input) { char *newbuf, *oldbuf; + unsigned int framing = params->mode & SNDRV_RAWMIDI_MODE_FRAMING_MASK; if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) return -EINVAL; + if (framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP && (params->buffer_size & 0x1f) != 0) + return -EINVAL; if (params->avail_min < 1 || params->avail_min > params->buffer_size) return -EINVAL; if (params->buffer_size != runtime->buffer_size) { @@ -720,7 +723,18 @@ EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params *params) { + unsigned int framing = params->mode & SNDRV_RAWMIDI_MODE_FRAMING_MASK; + unsigned int clock_type = params->mode & SNDRV_RAWMIDI_MODE_CLOCK_MASK; + + if (framing == SNDRV_RAWMIDI_MODE_FRAMING_NONE && clock_type != SNDRV_RAWMIDI_MODE_CLOCK_NONE) + return -EINVAL; + else if (clock_type > SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC_RAW) + return -EINVAL; + if (framing > SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) + return -EINVAL; snd_rawmidi_drain_input(substream); + substream->framing = framing; + substream->clock_type = clock_type; return resize_runtime_buffer(substream->runtime, params, true); } EXPORT_SYMBOL(snd_rawmidi_input_params); @@ -963,6 +977,61 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, return -ENOIOCTLCMD; } +static int receive_with_tstamp_framing(struct snd_rawmidi_substream *substream, + const unsigned char *buffer, int src_count, const struct timespec64 *tstamp) +{ + struct snd_rawmidi_runtime *runtime = substream->runtime; + struct snd_rawmidi_framing_tstamp *dest_ptr; + struct snd_rawmidi_framing_tstamp frame = { .tv_sec = tstamp->tv_sec, .tv_nsec = tstamp->tv_nsec }; + int dest_frames = 0; + int frame_size = sizeof(struct snd_rawmidi_framing_tstamp); + + BUILD_BUG_ON(frame_size != 0x20); + if (snd_BUG_ON((runtime->hw_ptr & 0x1f) != 0)) + return -EINVAL; + + while (src_count > 0) { + if ((int)(runtime->buffer_size - runtime->avail) < frame_size) { + runtime->xruns += src_count; + break; + } + if (src_count >= SNDRV_RAWMIDI_FRAMING_DATA_LENGTH) + frame.length = SNDRV_RAWMIDI_FRAMING_DATA_LENGTH; + else { + frame.length = src_count; + memset(frame.data, 0, SNDRV_RAWMIDI_FRAMING_DATA_LENGTH); + } + memcpy(frame.data, buffer, frame.length); + buffer += frame.length; + src_count -= frame.length; + dest_ptr = (struct snd_rawmidi_framing_tstamp *) (runtime->buffer + runtime->hw_ptr); + *dest_ptr = frame; + runtime->avail += frame_size; + runtime->hw_ptr += frame_size; + runtime->hw_ptr %= runtime->buffer_size; + dest_frames++; + } + return dest_frames * frame_size; +} + +static struct timespec64 get_framing_tstamp(struct snd_rawmidi_substream *substream) +{ + struct timespec64 ts64 = {0, 0}; + + switch (substream->clock_type) { + case SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC_RAW: + ktime_get_raw_ts64(&ts64); + break; + case SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC: + ktime_get_ts64(&ts64); + break; + case SNDRV_RAWMIDI_MODE_CLOCK_REALTIME: + ktime_get_real_ts64(&ts64); + break; + } + return ts64; +} + /** * snd_rawmidi_receive - receive the input data from the device * @substream: the rawmidi substream @@ -977,6 +1046,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count) { unsigned long flags; + struct timespec64 ts64 = get_framing_tstamp(substream); int result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -987,8 +1057,11 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, "snd_rawmidi_receive: input is not active!!!\n"); return -EINVAL; } + spin_lock_irqsave(&runtime->lock, flags); - if (count == 1) { /* special case, faster code */ + if (substream->framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) { + result = receive_with_tstamp_framing(substream, buffer, count, &ts64); + } else if (count == 1) { /* special case, faster code */ substream->bytes++; if (runtime->avail < runtime->buffer_size) { runtime->buffer[runtime->hw_ptr++] = buffer[0]; @@ -1534,6 +1607,7 @@ static __poll_t snd_rawmidi_poll(struct file *file, poll_table *wait) /* */ + static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { @@ -1541,6 +1615,8 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, struct snd_rawmidi_substream *substream; struct snd_rawmidi_runtime *runtime; unsigned long buffer_size, avail, xruns; + unsigned int clock_type; + static const char *clock_names[4] = { "none", "realtime", "monotonic", "monotonic raw" }; rmidi = entry->private_data; snd_iprintf(buffer, "%s\n\n", rmidi->name); @@ -1596,6 +1672,14 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Avail : %lu\n" " Overruns : %lu\n", buffer_size, avail, xruns); + if (substream->framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) { + clock_type = substream->clock_type >> SNDRV_RAWMIDI_MODE_CLOCK_SHIFT; + if (!snd_BUG_ON(clock_type >= sizeof(clock_names))) + snd_iprintf(buffer, + " Framing : tstamp\n" + " Clock type : %s\n", + clock_names[clock_type]); + } } } } diff --git a/sound/core/rawmidi_compat.c b/sound/core/rawmidi_compat.c index 7397130976d0..68a93443583c 100644 --- a/sound/core/rawmidi_compat.c +++ b/sound/core/rawmidi_compat.c @@ -13,7 +13,8 @@ struct snd_rawmidi_params32 { u32 buffer_size; u32 avail_min; unsigned int no_active_sensing; /* avoid bit-field */ - unsigned char reserved[16]; + unsigned int mode; + unsigned char reserved[12]; } __attribute__((packed)); static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile, @@ -25,6 +26,7 @@ static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile, if (get_user(params.stream, &src->stream) || get_user(params.buffer_size, &src->buffer_size) || get_user(params.avail_min, &src->avail_min) || + get_user(params.mode, &src->mode) || get_user(val, &src->no_active_sensing)) return -EFAULT; params.no_active_sensing = val; -- 2.25.1

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