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

[PATCH] ASoC: fsl_xcvr: disable all interrupts when suspend happens
by Shengjiu Wang 21 Jun '21

21 Jun '21

There is an unhandled interrupt after suspend, which cause endless interrupt when system resume, so system may hang. Disable all interrupts in runtime suspend callback to avoid above issue. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index df7c189d97dd..37b4fdb236ee 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1233,6 +1233,11 @@ static __maybe_unused int fsl_xcvr_runtime_suspend(struct device *dev) struct fsl_xcvr *xcvr = dev_get_drvdata(dev); int ret; + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_IER0, + FSL_XCVR_IRQ_EARC_ALL, 0); + if (ret < 0) + dev_err(dev, "Failed to clear IER0: %d\n", ret); + /* Assert M0+ reset */ ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, FSL_XCVR_EXT_CTRL_CORE_RESET, -- 2.27.0

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[PATCH -next 1/2] ASoC: qcom: apq8016_sbc: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 21 Jun '21

21 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/qcom/apq8016_sbc.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 270986b2f102..08a05f0ecad7 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -134,7 +134,6 @@ static int apq8016_sbc_platform_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; struct snd_soc_card *card; struct apq8016_sbc_data *data; - struct resource *res; int ret; data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); @@ -151,13 +150,11 @@ static int apq8016_sbc_platform_probe(struct platform_device *pdev) if (ret) return ret; - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mic-iomux"); - data->mic_iomux = devm_ioremap_resource(dev, res); + data->mic_iomux = devm_platform_ioremap_resource_byname(pdev, "mic-iomux"); if (IS_ERR(data->mic_iomux)) return PTR_ERR(data->mic_iomux); - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "spkr-iomux"); - data->spkr_iomux = devm_ioremap_resource(dev, res); + data->spkr_iomux = devm_platform_ioremap_resource_byname(pdev, "spkr-iomux"); if (IS_ERR(data->spkr_iomux)) return PTR_ERR(data->spkr_iomux); -- 2.25.1

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[PATCH -next 0/3] ASoC: sunxi: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 21 Jun '21

21 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Yang Yingliang (3): ASoC: sunxi: sun4i-codec: Use devm_platform_get_and_ioremap_resource() ASoC: sun4i-i2s: Use devm_platform_get_and_ioremap_resource() ASoC: sunxi: sun4i-spdif: Use devm_platform_get_and_ioremap_resource() sound/soc/sunxi/sun4i-codec.c | 3 +-- sound/soc/sunxi/sun4i-i2s.c | 3 +-- sound/soc/sunxi/sun4i-spdif.c | 3 +-- 3 files changed, 3 insertions(+), 6 deletions(-) -- 2.25.1

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[PATCH 1/2] ASoC: atmel-i2s: Set symmetric sample bits
by Codrin Ciubotariu 21 Jun '21

21 Jun '21

The I2S needs to have the same sample bits for both capture and playback streams. Fixes: b543e467d1a9 ("ASoC: atmel-i2s: add driver for the new Atmel I2S controller") Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/atmel/atmel-i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index e8fe1a7a4d83..a95002b0cc8e 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -542,6 +542,7 @@ static struct snd_soc_dai_driver atmel_i2s_dai = { }, .ops = &atmel_i2s_dai_ops, .symmetric_rate = 1, + .symmetric_sample_bits = 1, }; static const struct snd_soc_component_driver atmel_i2s_component = { -- 2.30.2

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[PATCH -next v2 0/9] ASoC: fsl: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 21 Jun '21

21 Jun '21

patch #1 ~ #8: Use devm_platform_get_and_ioremap_resource() patch #9 check return value of platform_get_resource_byname() v2: change error message in patch #9 Yang Yingliang (9): ASoC: fsl_asrc: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_aud2htx: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_easrc: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_esai: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_micfil: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_sai: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_spdif: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_ssi: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_xcvr: check return value after calling platform_get_resource_byname() sound/soc/fsl/fsl_asrc.c | 3 +-- sound/soc/fsl/fsl_aud2htx.c | 3 +-- sound/soc/fsl/fsl_easrc.c | 3 +-- sound/soc/fsl/fsl_esai.c | 3 +-- sound/soc/fsl/fsl_micfil.c | 3 +-- sound/soc/fsl/fsl_sai.c | 3 +-- sound/soc/fsl/fsl_spdif.c | 3 +-- sound/soc/fsl/fsl_ssi.c | 3 +-- sound/soc/fsl/fsl_xcvr.c | 4 ++++ 9 files changed, 12 insertions(+), 16 deletions(-) -- 2.25.1

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[PATCH 2/2] ASoC: rt711: add two jack detection modes
by shumingf＠realtek.com 21 Jun '21

21 Jun '21

From: Shuming Fan <shumingf(a)realtek.com> Some boards use different circuits for jack detection. This patch adds two modes as below 1. JD2/2 ports/external resister 100k 2. JD2/1 port/JD voltage 1.8V Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdw.h | 2 ++ sound/soc/codecs/rt711.c | 30 ++++++++++++++++++++++++++++++ sound/soc/codecs/rt711.h | 29 ++++++++++++++++++++++++++++- 3 files changed, 60 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdw.h b/sound/soc/codecs/rt711-sdw.h index 43b2b984b29c..6acf9858330d 100644 --- a/sound/soc/codecs/rt711-sdw.h +++ b/sound/soc/codecs/rt711-sdw.h @@ -267,7 +267,9 @@ static const struct reg_default rt711_reg_defaults[] = { { 0x8393, 0x00 }, { 0x7319, 0x00 }, { 0x8399, 0x00 }, + { 0x752008, 0xa807 }, { 0x752009, 0x1029 }, + { 0x75200b, 0x7770 }, { 0x752011, 0x007a }, { 0x75201a, 0x8003 }, { 0x752045, 0x5289 }, diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index c3a82b462a93..301c7ab27878 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -389,6 +389,36 @@ static void rt711_jack_init(struct rt711_priv *rt711) RT711_HP_JD_FINAL_RESULT_CTL_JD12, RT711_HP_JD_FINAL_RESULT_CTL_JD12); break; + case RT711_JD2_100K: + rt711_index_update_bits(rt711->regmap, RT711_VENDOR_REG, + RT711_JD_CTL2, RT711_JD2_2PORT_100K_DECODE | RT711_JD2_1PORT_TYPE_DECODE | + RT711_HP_JD_SEL_JD2 | RT711_JD1_2PORT_TYPE_100K_DECODE, + RT711_JD2_2PORT_100K_DECODE_HP | RT711_JD2_1PORT_JD_HP | + RT711_HP_JD_SEL_JD2 | RT711_JD1_2PORT_JD_RESERVED); + rt711_index_update_bits(rt711->regmap, RT711_VENDOR_REG, + RT711_CC_DET1, + RT711_HP_JD_FINAL_RESULT_CTL_JD12, + RT711_HP_JD_FINAL_RESULT_CTL_JD12); + break; + case RT711_JD2_1P8V_1PORT: + rt711_index_update_bits(rt711->regmap, RT711_VENDOR_REG, + RT711_JD_CTL1, RT711_JD2_DIGITAL_JD_MODE_SEL, + RT711_JD2_1_JD_MODE); + rt711_index_update_bits(rt711->regmap, RT711_VENDOR_REG, + RT711_JD_CTL2, RT711_JD2_1PORT_TYPE_DECODE | + RT711_HP_JD_SEL_JD2, + RT711_JD2_1PORT_JD_HP | + RT711_HP_JD_SEL_JD2); + rt711_index_update_bits(rt711->regmap, RT711_VENDOR_REG, + RT711_JD_CTL4, RT711_JD2_PAD_PULL_UP_MASK | + RT711_JD2_MODE_SEL_MASK, + RT711_JD2_PAD_PULL_UP | + RT711_JD2_MODE2_1P8V_1PORT); + rt711_index_update_bits(rt711->regmap, RT711_VENDOR_REG, + RT711_CC_DET1, + RT711_HP_JD_FINAL_RESULT_CTL_JD12, + RT711_HP_JD_FINAL_RESULT_CTL_JD12); + break; default: dev_warn(rt711->component->dev, "Wrong JD source\n"); break; diff --git a/sound/soc/codecs/rt711.h b/sound/soc/codecs/rt711.h index ca0f581feec7..5f2ba1341085 100644 --- a/sound/soc/codecs/rt711.h +++ b/sound/soc/codecs/rt711.h @@ -52,7 +52,9 @@ struct sdw_stream_data { /* Index (NID:20h) */ #define RT711_DAC_DC_CALI_CTL1 0x00 +#define RT711_JD_CTL1 0x08 #define RT711_JD_CTL2 0x09 +#define RT711_JD_CTL4 0x0b #define RT711_CC_DET1 0x11 #define RT711_PARA_VERB_CTL 0x1a #define RT711_COMBO_JACK_AUTO_CTL1 0x45 @@ -171,10 +173,33 @@ struct sdw_stream_data { /* DAC DC offset calibration control-1 (0x00)(NID:20h) */ #define RT711_DAC_DC_CALI_TRIGGER (0x1 << 15) +/* jack detect control 1 (0x08)(NID:20h) */ +#define RT711_JD2_DIGITAL_JD_MODE_SEL (0x1 << 1) +#define RT711_JD2_1_JD_MODE (0x0 << 1) +#define RT711_JD2_2_JD_MODE (0x1 << 1) + /* jack detect control 2 (0x09)(NID:20h) */ #define RT711_JD2_2PORT_200K_DECODE_HP (0x1 << 13) +#define RT711_JD2_2PORT_100K_DECODE (0x1 << 12) +#define RT711_JD2_2PORT_100K_DECODE_HP (0x0 << 12) #define RT711_HP_JD_SEL_JD1 (0x0 << 1) #define RT711_HP_JD_SEL_JD2 (0x1 << 1) +#define RT711_JD2_1PORT_TYPE_DECODE (0x3 << 10) +#define RT711_JD2_1PORT_JD_LINE2 (0x0 << 10) +#define RT711_JD2_1PORT_JD_HP (0x1 << 10) +#define RT711_JD2_1PORT_JD_LINE1 (0x2 << 10) +#define RT711_JD1_2PORT_TYPE_100K_DECODE (0x1 << 0) +#define RT711_JD1_2PORT_JD_RESERVED (0x0 << 0) +#define RT711_JD1_2PORT_JD_LINE1 (0x1 << 0) + +/* jack detect control 4 (0x0b)(NID:20h) */ +#define RT711_JD2_PAD_PULL_UP_MASK (0x1 << 3) +#define RT711_JD2_PAD_NOT_PULL_UP (0x0 << 3) +#define RT711_JD2_PAD_PULL_UP (0x1 << 3) +#define RT711_JD2_MODE_SEL_MASK (0x3 << 0) +#define RT711_JD2_MODE0_2PORT (0x0 << 0) +#define RT711_JD2_MODE1_3P3V_1PORT (0x1 << 0) +#define RT711_JD2_MODE2_1P8V_1PORT (0x2 << 0) /* CC DET1 (0x11)(NID:20h) */ #define RT711_HP_JD_FINAL_RESULT_CTL_JD12 (0x1 << 10) @@ -215,7 +240,9 @@ enum { enum rt711_jd_src { RT711_JD_NULL, RT711_JD1, - RT711_JD2 + RT711_JD2, + RT711_JD2_100K, + RT711_JD2_1P8V_1PORT }; int rt711_io_init(struct device *dev, struct sdw_slave *slave); -- 2.31.1

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[PATCH V2 00/14] Cleanup before adding Scarlett Gen 3 support
by Geoffrey D. Bennett 21 Jun '21

21 Jun '21

Hi Takashi, Here is version 2 of a set of patches which is some cleanup of the Scarlett Gen 2 mixer driver in preparation for adding Scarlett Gen 3 support. One review comment I got (from Hin-Tak) was: > 40+ patches is a lot, for modifying just one file. I would collapse > it all into one and break it up again to under 10, maybe, broadly > into "functionally-equivalent re-org", "small isolated bug fixes", > "additional functions, not yet used", "hooking up those new > functions", etc? I'm not sure that I agree with that comment -- I tried to follow the Documentation/process/submitting-patches.rst advice of "Separate each logical change into a separate patch" for easy review of the individual pieces, but perhaps I went too far in that direction? Please let me know if I should combine some of these patches together. After this set of patches I have a second set of patches almost ready. They add a few features to the Gen 2 support and update the code ready to add Gen 3 support: ALSA: usb-audio: scarlett2: Add usb_tx/rx functions ALSA: usb-audio: scarlett2: Update initialisation sequence ALSA: usb-audio: scarlett2: Fix 6i6 Gen 2 line out descriptions ALSA: usb-audio: scarlett2: Always enable interrupt polling ALSA: usb-audio: scarlett2: Add "Sync Status" control ALSA: usb-audio: scarlett2: Merge common line in capture strings ALSA: usb-audio: scarlett2: Reformat scarlett2_config_items[] ALSA: usb-audio: scarlett2: Improve device info lookup ALSA: usb-audio: scarlett2: Move info lookup out of init function ALSA: usb-audio: scarlett2: Remove repeated device info comments ALSA: usb-audio: scarlett2: Add scarlett2_vol_ctl_write() helper ALSA: usb-audio: scarlett2: Add mute support ALSA: usb-audio: scarlett2: Allow arbitrary ordering of mux entries ALSA: usb-audio: scarlett2: Split struct scarlett2_ports ALSA: usb-audio: scarlett2: Fix Level Meter control The final set of patches adds Gen 3 mixer support and then adds support for the new Gen 3 features: ALSA: usb-audio: scarlett2: Add Gen 3 mixer support ALSA: usb-audio: scarlett2: Add support for "input-other" notify ALSA: usb-audio: scarlett2: Add Gen 3 MSD mode switch ALSA: usb-audio: scarlett2: Move get config above set config ALSA: usb-audio: scarlett2: Allow bit-level access to config ALSA: usb-audio: scarlett2: Add support for Solo and 2i2 Gen 3 ALSA: usb-audio: scarlett2: Add "air" switch support ALSA: usb-audio: scarlett2: Add phantom power switch support ALSA: usb-audio: scarlett2: Add direct monitor support ALSA: usb-audio: scarlett2: Label 18i8 Gen 3 line outputs correctly ALSA: usb-audio: scarlett2: Split up sw_hw_enum_ctl_put() ALSA: usb-audio: scarlett2: Add sw_hw_ctls and mux_ctls ALSA: usb-audio: scarlett2: Update mux controls to allow updates ALSA: usb-audio: scarlett2: Add speaker switching support ALSA: usb-audio: scarlett2: Update get_config to do endian conversion ALSA: usb-audio: scarlett2: Add support for the talkback feature If you'd like to see the commits so far all at once, please check this branch: https://github.com/geoffreybennett/scarlett-gen2/commits/scarlett-gen3-for-… Thanks, Geoffrey. Geoffrey D. Bennett (14): ALSA: usb-audio: scarlett2: Remove incorrect S/PDIF comment ALSA: usb-audio: scarlett2: Fix 18i8 Gen 2 PCM Input count ALSA: usb-audio: scarlett2: Coding style improvements ALSA: usb-audio: scarlett2: Remove unused/useless code ALSA: usb-audio: scarlett2: Remove interrupt debug message ALSA: usb-audio: scarlett2: Remove redundant info->button_count ALSA: usb-audio: scarlett2: Rename buttons/interrupts/vol ALSA: usb-audio: scarlett2: Rename struct scarlett2_mixer_data ALSA: usb-audio: scarlett2: Add temp variable for consistency ALSA: usb-audio: scarlett2: Fix data_mutex lock ALSA: usb-audio: scarlett2: Fix scarlett2_*_ctl_put() return values ALSA: usb-audio: scarlett2: Fix union usage in mixer control callbacks ALSA: usb-audio: scarlett2: Don't copy struct scarlett2_config ALSA: usb-audio: scarlett2: Remove hard-coded USB #defines sound/usb/mixer_scarlett_gen2.c | 425 +++++++++++++++++--------------- 1 file changed, 224 insertions(+), 201 deletions(-) base-commit: f8fbcdfb0665de60997d9746809e1704ed782bbc -- 2.31.1

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[PATCH 1/2] control: Add documentation for snd_ctl_card_* and friends.
by Tanjeff-N. Moos 21 Jun '21

21 Jun '21

I sent this patch before, but it didn't make it into the Git repo. If I could improve something, please tell me. In this patch series, I added a description about control interface handling and how control interfaces are identified. In addition, I added/improved Doxygen documentation for the snd_ctl_card_info_t type and related corresponding functions, e.g. snd_ctl_card_info(). I also documented other card-related like snd_card_next(). Along the way I did minor documentation improvements. Signed-off-by: Tanjeff-N. Moos <tanjeff(a)cccmz.de> --- include/control.h | 40 +++++++++- src/control/cards.c | 56 +++++++++----- src/control/control.c | 175 +++++++++++++++++++++++++++++++----------- 3 files changed, 202 insertions(+), 69 deletions(-) diff --git a/include/control.h b/include/control.h index ccf906e2..9ebb4905 100644 --- a/include/control.h +++ b/include/control.h @@ -50,7 +50,32 @@ typedef struct snd_aes_iec958 { unsigned char dig_subframe[4]; /**< AES/IEC958 subframe bits */ } snd_aes_iec958_t; -/** CTL card info container */ +/** \brief CTL card info container. + * + * This type contains meta information about a sound card, such as the index, + * name, longname, etc. + * + * \par Memory management + * + * Before using a snd_ctl_card_info_t object, it must be allocated using + * snd_ctl_card_info_alloca() or snd_ctl_card_info_malloc(). When using the + * latter, it must be freed again using snd_ctl_card_info_free(). + * + * A card info object can be zeroed out using snd_ctl_card_info_clear(). + * + * A card info object can be copied to another one using + * snd_ctl_card_info_copy(). + * + * \par Obtaining the Information + * + * To obtain the card information, it must first be opened using + * snd_ctl_open(), and a snd_ctl_card_info_t container must be + * allocated. Then, the information can be read using + * snd_ctl_card_info_get_card(). + * + * Thereafter, the card properties can be read using the + * snd_ctl_card_info_get_*() functions. + */ typedef struct _snd_ctl_card_info snd_ctl_card_info_t; /** CTL element identifier container */ @@ -442,11 +467,20 @@ void snd_ctl_elem_id_set_name(snd_ctl_elem_id_t *obj, const char *val); void snd_ctl_elem_id_set_index(snd_ctl_elem_id_t *obj, unsigned int val); size_t snd_ctl_card_info_sizeof(void); + /** \hideinitializer - * \brief allocate an invalid #snd_ctl_card_info_t using standard alloca - * \param ptr returned pointer + * \brief Allocate an invalid #snd_ctl_card_info_t on the stack. + * + * Allocate space for a card info object on the stack. The allocated + * memory need not be freed, because it is on the stack. + * + * See snd_ctl_card_info_t for details. + * + * \param ptr Pointer to a snd_ctl_elem_value_t pointer. The address + * of the allocated space will returned here. */ #define snd_ctl_card_info_alloca(ptr) __snd_alloca(ptr, snd_ctl_card_info) + int snd_ctl_card_info_malloc(snd_ctl_card_info_t **ptr); void snd_ctl_card_info_free(snd_ctl_card_info_t *obj); void snd_ctl_card_info_clear(snd_ctl_card_info_t *obj); diff --git a/src/control/cards.c b/src/control/cards.c index e57457c2..8226c42d 100644 --- a/src/control/cards.c +++ b/src/control/cards.c @@ -77,8 +77,8 @@ static int snd_card_load1(int card) /** * \brief Try to load the driver for a card. - * \param card Card number. - * \return 1 if driver is present, zero if driver is not present + * \param card Card index. + * \return 1 if driver is present, zero if driver is not present. */ int snd_card_load(int card) { @@ -86,14 +86,24 @@ int snd_card_load(int card) } /** - * \brief Try to determine the next card. - * \param rcard pointer to card number - * \result zero if success, otherwise a negative error code + * \brief Iterate over physical sound cards. + * + * This function takes the index of a physical sound card and sets it to the + * index of the next card. If index is -1, it is set to the index of the first + * card. After the last card, the index is set to -1. + * + * For example, if you have 2 sound cards (with index 0 and 1), the index will + * be modified as follows: + * + * - -1 --> 0 + * - 0 --> 1 + * - 1 --> -1 + * + * This does not work for virtual sound cards. * - * Tries to determine the next card from given card number. - * If card number is -1, then the first available card is - * returned. If the result card number is -1, no more cards - * are available. + * \param rcard Index of current card. The index of the next card is stored + * here. + * \result zero if success, otherwise a negative error code. */ int snd_card_next(int *rcard) { @@ -114,13 +124,18 @@ int snd_card_next(int *rcard) } /** - * \brief Convert card string to an integer value. - * \param string String containing card identifier - * \return zero if success, otherwise a negative error code + * \brief Convert a card string to the card index. * - * The accepted format is an integer value in ASCII representation - * or the card identifier (the id parameter for sound-card drivers). - * The control device name like /dev/snd/controlC0 is accepted, too. + * This works only for physical sound cards, not for virtual cards. + * + * \param string A string identifying the card. + * \return The index of the card. On error, a a negative error code + * is returned. + * + * The accepted formats for "string" are: + * - The index of the card (as listed in /proc/asound/cards), given as string + * - The ID of the card (as listed in /proc/asound/cards) + * - The control device name (like /dev/snd/controlC0) */ int snd_card_get_index(const char *string) { @@ -163,8 +178,9 @@ int snd_card_get_index(const char *string) /** * \brief Obtain the card name. - * \param card Card number - * \param name Result - card name corresponding to card number + * + * \param card The index of the card. + * \param name Result - card name corresponding to card index. * \result zero if success, otherwise a negative error code * * The value returned in name is allocated with strdup and should be @@ -193,9 +209,9 @@ int snd_card_get_name(int card, char **name) /** * \brief Obtain the card long name. - * \param card Card number - * \param name Result - card long name corresponding to card number - * \result zero if success, otherwise a negative error code + * \param card Index of the card. + * \param name Result - card long name corresponding to card index. + * \result Zero if success, otherwise a negative error code. * * The value returned in name is allocated with strdup and should be * freed when no longer used. diff --git a/src/control/control.c b/src/control/control.c index ed986e54..7cf4decb 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -33,8 +33,50 @@ <P>Control interface is designed to access primitive controls. There is also an interface for notifying about control and structure changes. + \section control_general_overview General overview +In Alsa, there are physical sound cards, such as USB headsets, and +virtual sound cards, such as "pulse", which represents the PulseAudio +Sound system. Each sound card offers a control interface, making its +settings (e.g. volume knobs) available. The complete list of available +control interfaces can be obtained using snd_device_name_hint(), +giving -1 as card index and "ctl" as interface type. Each returned +NAME hint identifies a control interface. + +Sound cards have an ID (a string), an index (an int, sometimes called +the "card number"), a name, a longname, a mixername and a "components" +property. The file /proc/asound/cards lists most of these properties +for physical sound cards. Virtual sound cards are not listed in that +file. The format is: + +\verbatim +index [ID ] Driver - name + longname +\endverbatim + +Note that the mixername and components are not listed. + + +\subsection control_cards_id Identifying and Opening Control Interfaces + +To work with a control interface, is must be opened first, using +snd_ctl_open(). This function takes the interface name. + +For physical sound cards, the control interface can be identified +using the string "hw:<index>" (e.g. `hw:2`). The NAME hint - which is +"hw:CARD=<ID>" - can also be used. Further, its device file (something +like `/dev/snd/controlC0`) is also acceptable. Either of them can be +given to snd_ctl_open(). + +For virtual sound cards, the NAME hint is given to snd_ctl_open(). + +The functions snd_card_get_index(), snd_card_get_name() and +snd_card_get_longname() can be used to find an identifying property +when another one is already known. + +\section control_elements Elements + In ALSA control feature, each sound card can have control elements. The elements are managed according to below model. @@ -65,7 +107,7 @@ are managed according to below model. of userspace applications and drivers in kernel. -\section identifying_elements Identifying Elements +\subsection identifying_elements Identifying Elements Each element has the following identifying properties: @@ -84,7 +126,7 @@ but in practice this is rare). The numid can change on each boot. In case of an USB sound card, the numid can also change when it is reconnected. The short numid is used to reduce the lookup time. -\section element_lists Element Lists +\subsection element_lists Element Lists An element list can be used to obtain a list of all elements of the sound card. The list contains generic information (e.g. how many @@ -93,7 +135,7 @@ elements the card has), and the identifying properties of the elements element lists. -\section working_with_elements Working with Elements +\subsection working_with_elements Working with Elements It is possible to obtain information about an element using the snd_ctl_elem_info_*() functions. For enums, the allowed values can be @@ -108,7 +150,7 @@ actual values or settings. It is also possible to get and set the ID values (such as the numid or the name). -\section element_sets Element Sets +\subsection element_sets Element Sets The type of element set is one of integer, integer64, boolean, enumerators, bytes and IEC958 structure. This indicates the type of value for each member in @@ -329,10 +371,15 @@ int snd_ctl_subscribe_events(snd_ctl_t *ctl, int subscribe) /** - * \brief Get card related information - * \param ctl CTL handle - * \param info Card info pointer - * \return 0 on success otherwise a negative error code + * \brief Get information about the sound card. + * + * Obtain information about the sound card previously opened using + * snd_ctl_open(). The object "info" must be allocated prior to calling this + * function. See snd_ctl_card_info_t for details. + * + * \param ctl The CTL handle. + * \param info The card information is stored here. + * \return 0 on success, otherwise a negative error code. */ int snd_ctl_card_info(snd_ctl_t *ctl, snd_ctl_card_info_t *info) { @@ -1508,11 +1555,13 @@ int _snd_ctl_open_named_child(snd_ctl_t **pctl, const char *name, #endif /** - * \brief Opens a CTL - * \param ctlp Returned CTL handle - * \param name ASCII identifier of the CTL handle - * \param mode Open mode (see #SND_CTL_NONBLOCK, #SND_CTL_ASYNC) - * \return 0 on success otherwise a negative error code + * \brief Opens a sound card. + * + * \param ctlp Returned CTL handle. + * \param name A string identifying the card (See \ref control_cards_id). + * \param mode Open mode (see #SND_CTL_NONBLOCK, #SND_CTL_ASYNC). + * + * \return 0 on success otherwise a negative error code. */ int snd_ctl_open(snd_ctl_t **ctlp, const char *name, int mode) { @@ -2027,8 +2076,8 @@ void snd_ctl_elem_id_set_index(snd_ctl_elem_id_t *obj, unsigned int val) } /** - * \brief get size of #snd_ctl_card_info_t - * \return size in bytes + * \brief get size of #snd_ctl_card_info_t. + * \return Size in bytes. */ size_t snd_ctl_card_info_sizeof() { @@ -2036,9 +2085,16 @@ size_t snd_ctl_card_info_sizeof() } /** - * \brief allocate an invalid #snd_ctl_card_info_t using standard malloc - * \param ptr returned pointer - * \return 0 on success otherwise negative error code + * \brief Allocate an invalid #snd_ctl_card_info_t on the heap. + * + * Allocate space for a card info object on the heap. The allocated memory + * must be freed using snd_ctl_card_info_free(). + * + * See snd_ctl_card_info_t for details. + * + * \param ptr Pointer to a snd_ctl_card_info_t pointer. The address + * of the allocated space will be returned here. + * \return 0 on success, otherwise a negative error code. */ int snd_ctl_card_info_malloc(snd_ctl_card_info_t **ptr) { @@ -2050,8 +2106,10 @@ int snd_ctl_card_info_malloc(snd_ctl_card_info_t **ptr) } /** - * \brief frees a previously allocated #snd_ctl_card_info_t - * \param obj pointer to object to free + * \brief Free an #snd_ctl_card_info_t previously allocated using + * snd_ctl_card_info_malloc(). + * + * \param obj Pointer to the snd_ctl_card_info_t. */ void snd_ctl_card_info_free(snd_ctl_card_info_t *obj) { @@ -2059,8 +2117,11 @@ void snd_ctl_card_info_free(snd_ctl_card_info_t *obj) } /** - * \brief clear given #snd_ctl_card_info_t object - * \param obj pointer to object to clear + * \brief Clear given card info object. + * + * See snd_ctl_elem_value_t for details. + * + * \param obj Card info object. */ void snd_ctl_card_info_clear(snd_ctl_card_info_t *obj) { @@ -2068,9 +2129,10 @@ void snd_ctl_card_info_clear(snd_ctl_card_info_t *obj) } /** - * \brief copy one #snd_ctl_card_info_t to another - * \param dst pointer to destination - * \param src pointer to source + * \brief Bitwise copy of a #snd_ctl_card_info_t object. + * + * \param dst Pointer to destination. + * \param src Pointer to source. */ void snd_ctl_card_info_copy(snd_ctl_card_info_t *dst, const snd_ctl_card_info_t *src) { @@ -2079,9 +2141,12 @@ void snd_ctl_card_info_copy(snd_ctl_card_info_t *dst, const snd_ctl_card_info_t } /** - * \brief Get card number from a CTL card info - * \param obj CTL card info - * \return card number + * \brief Get the sound card index from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound card index. */ int snd_ctl_card_info_get_card(const snd_ctl_card_info_t *obj) { @@ -2090,9 +2155,12 @@ int snd_ctl_card_info_get_card(const snd_ctl_card_info_t *obj) } /** - * \brief Get card identifier from a CTL card info - * \param obj CTL card info - * \return card identifier + * \brief Get the sound card ID from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound card ID. */ const char *snd_ctl_card_info_get_id(const snd_ctl_card_info_t *obj) { @@ -2101,9 +2169,12 @@ const char *snd_ctl_card_info_get_id(const snd_ctl_card_info_t *obj) } /** - * \brief Get card driver name from a CTL card info - * \param obj CTL card info - * \return card driver name + * \brief Get the sound card driver from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return The sound card driver. */ const char *snd_ctl_card_info_get_driver(const snd_ctl_card_info_t *obj) { @@ -2112,9 +2183,12 @@ const char *snd_ctl_card_info_get_driver(const snd_ctl_card_info_t *obj) } /** - * \brief Get card name from a CTL card info - * \param obj CTL card info - * \return card name + * \brief Get the sound card name from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound card name. */ const char *snd_ctl_card_info_get_name(const snd_ctl_card_info_t *obj) { @@ -2123,9 +2197,12 @@ const char *snd_ctl_card_info_get_name(const snd_ctl_card_info_t *obj) } /** - * \brief Get card long name from a CTL card info - * \param obj CTL card info - * \return card long name + * \brief Get the sound cards long name from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound cards long name. */ const char *snd_ctl_card_info_get_longname(const snd_ctl_card_info_t *obj) { @@ -2134,9 +2211,12 @@ const char *snd_ctl_card_info_get_longname(const snd_ctl_card_info_t *obj) } /** - * \brief Get card mixer name from a CTL card info - * \param obj CTL card info - * \return card mixer name + * \brief Get the sound card mixer name from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound card mixer name. */ const char *snd_ctl_card_info_get_mixername(const snd_ctl_card_info_t *obj) { @@ -2145,9 +2225,12 @@ const char *snd_ctl_card_info_get_mixername(const snd_ctl_card_info_t *obj) } /** - * \brief Get card component list from a CTL card info - * \param obj CTL card info - * \return card mixer identifier + * \brief Get the sound cards "components" property from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound cards "components" property. */ const char *snd_ctl_card_info_get_components(const snd_ctl_card_info_t *obj) { -- 2.25.1

2 3

A52 documentation in need of updates.
by GitHub issues - opened 21 Jun '21

21 Jun '21

alsa-project/alsa-plugins issue #29 was opened from quequotion: [a52.txt](https://github.com/alsa-project/alsa-plugins/blob/4219f4b3fded74d7… hasn't been changed in fifteen years. In that time, [a preconfiguration has been supplied that ends up getting merged into a dysfunctional mess if users add a manual configuration to their asound.conf without prepending an exclamation point to override it](https://github.com/alsa-project/alsa-lib/issues/154#issuecomment-859799…, and more recently [more sample formats have been added](https://github.com/alsa-project/alsa-plugins/commit/f11e7a8994111f97…. This file needs some attention. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/29 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v3] ASoC: qcom: Fix for DMA interrupt clear reg overwriting
by Srinivasa Rao Mandadapu 21 Jun '21

21 Jun '21

The DMA interrupt clear register overwritten during simultaneous playback and capture in lpass platform interrupt handler. It's causing playback or capture stuck in similtaneous plaback on speaker and capture on dmic test. Update appropriate reg fields of corresponding channel instead of entire register write. Fixes: commit c5c8635a04711 ("ASoC: qcom: Add LPASS platform driver") Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes since v2: -- Removed redundant variables. Changes since v1: -- Subject lines changed. sound/soc/qcom/lpass-platform.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 0df9481ea4c6..f9df76d37858 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -526,7 +526,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, return -EINVAL; } - ret = regmap_write(map, reg_irqclr, val_irqclr); + ret = regmap_update_bits(map, reg_irqclr, val_irqclr, val_irqclr); if (ret) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", ret); return ret; @@ -650,10 +650,11 @@ static irqreturn_t lpass_dma_interrupt_handler( struct lpass_variant *v = drvdata->variant; irqreturn_t ret = IRQ_NONE; int rv; - unsigned int reg = 0, val = 0; + unsigned int reg, val, mask; struct regmap *map; unsigned int dai_id = cpu_dai->driver->id; + mask = LPAIF_IRQ_ALL(chan); switch (dai_id) { case LPASS_DP_RX: map = drvdata->hdmiif_map; @@ -676,8 +677,7 @@ static irqreturn_t lpass_dma_interrupt_handler( return -EINVAL; } if (interrupts & LPAIF_IRQ_PER(chan)) { - - rv = regmap_write(map, reg, LPAIF_IRQ_PER(chan) | val); + rv = regmap_update_bits(map, reg, mask, (LPAIF_IRQ_PER(chan) | val)); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -688,7 +688,7 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_XRUN(chan)) { - rv = regmap_write(map, reg, LPAIF_IRQ_XRUN(chan) | val); + rv = regmap_update_bits(map, reg, mask, (LPAIF_IRQ_XRUN(chan) | val)); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -700,7 +700,7 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_ERR(chan)) { - rv = regmap_write(map, reg, LPAIF_IRQ_ERR(chan) | val); + rv = regmap_update_bits(map, reg, mask, (LPAIF_IRQ_ERR(chan) | val)); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 6

Lenovo Legion ALC287/ALC3306 issue
by GitHub issues - opened 21 Jun '21

21 Jun '21

alsa-project/alsa-lib issue #160 was opened from camberkenpas: Hello, I was able to do a little digging into this issue by running a Windows 10 virtual machine and passing the sound card device to the virtual machine I was able to get audio to play through the laptop's virtual machine using the virtual machine. More importantly, I was able to capture traces that might help us determine what this sound card needs. I've attached traces at the ticket for this bug: https://bugzilla.kernel.org/show_bug.cgi?id=208555 Perhaps this is enough to deduce at least some of what verbs the codec wants in order to work. Is this the right place to share this? Or is the kernel bug tracker the only appropriate place for this? Issue URL : https://github.com/alsa-project/alsa-lib/issues/160 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 14/14] ALSA: usb-audio: scarlett2: Remove hard-coded USB #defines
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Remove the hard-coded interface number and related constants for the vendor-specific interface and look them up from the USB endpoint descriptor. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 74 ++++++++++++++++++++++++--------- 1 file changed, 55 insertions(+), 19 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 45fd540920b9..2e1937b072ee 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -230,6 +230,10 @@ struct scarlett2_data { struct mutex data_mutex; /* lock access to this data */ struct delayed_work work; const struct scarlett2_device_info *info; + __u8 bInterfaceNumber; + __u8 bEndpointAddress; + __u16 wMaxPacketSize; + __u8 bInterval; int num_mux_srcs; int num_mux_dsts; u16 scarlett2_seq; @@ -444,12 +448,6 @@ static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, /*** USB Interactions ***/ -/* Vendor-Specific Interface, Endpoint, MaxPacketSize, Interval */ -#define SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE 5 -#define SCARLETT2_USB_INTERRUPT_ENDPOINT 4 -#define SCARLETT2_USB_INTERRUPT_MAX_DATA 64 -#define SCARLETT2_USB_INTERRUPT_INTERVAL 3 - /* Interrupt flags for dim/mute button and monitor changes */ #define SCARLETT2_USB_NOTIFY_DIM_MUTE 0x00200000 #define SCARLETT2_USB_NOTIFY_MONITOR 0x00400000 @@ -615,7 +613,7 @@ static int scarlett2_usb( SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, 0, - SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + private->bInterfaceNumber, req, req_buf_size); @@ -635,7 +633,7 @@ static int scarlett2_usb( SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, 0, - SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + private->bInterfaceNumber, resp, resp_buf_size); @@ -1886,12 +1884,45 @@ static void scarlett2_count_mux_io(struct scarlett2_data *private) private->num_mux_dsts = dsts; } -/* Initialise private data and sequence number */ +/* Look through the interface descriptors for the Focusrite Control + * interface (bInterfaceClass = 255 Vendor Specific Class) and set + * bInterfaceNumber, bEndpointAddress, wMaxPacketSize, and bInterval + * in private + */ +static int scarlett2_find_fc_interface(struct usb_device *dev, + struct scarlett2_data *private) +{ + struct usb_host_config *config = dev->actconfig; + int i; + + for (i = 0; i < config->desc.bNumInterfaces; i++) { + struct usb_interface *intf = config->interface[i]; + struct usb_interface_descriptor *desc = + &intf->altsetting[0].desc; + struct usb_endpoint_descriptor *epd; + + if (desc->bInterfaceClass != 255) + continue; + + epd = get_endpoint(intf->altsetting, 0); + private->bInterfaceNumber = desc->bInterfaceNumber; + private->bEndpointAddress = epd->bEndpointAddress & + USB_ENDPOINT_NUMBER_MASK; + private->wMaxPacketSize = le16_to_cpu(epd->wMaxPacketSize); + private->bInterval = epd->bInterval; + return 0; + } + + return -EINVAL; +} + +/* Initialise private data, sequence number, and get the USB data */ static int scarlett2_init_private(struct usb_mixer_interface *mixer, const struct scarlett2_device_info *info) { struct scarlett2_data *private = kzalloc(sizeof(struct scarlett2_data), GFP_KERNEL); + int err; if (!private) return -ENOMEM; @@ -1899,13 +1930,19 @@ static int scarlett2_init_private(struct usb_mixer_interface *mixer, mutex_init(&private->usb_mutex); mutex_init(&private->data_mutex); INIT_DELAYED_WORK(&private->work, scarlett2_config_save_work); + + mixer->private_data = private; + mixer->private_free = scarlett2_private_free; + mixer->private_suspend = scarlett2_private_suspend; + private->info = info; scarlett2_count_mux_io(private); private->scarlett2_seq = 0; private->mixer = mixer; - mixer->private_data = private; - mixer->private_free = scarlett2_private_free; - mixer->private_suspend = scarlett2_private_suspend; + + err = scarlett2_find_fc_interface(mixer->chip->dev, private); + if (err < 0) + return err; /* Initialise the sequence number used for the proprietary commands */ return scarlett2_usb(mixer, SCARLETT2_USB_INIT_SEQ, NULL, 0, NULL, 0); @@ -2050,8 +2087,8 @@ static void scarlett2_notify(struct urb *urb) static int scarlett2_init_notify(struct usb_mixer_interface *mixer) { struct usb_device *dev = mixer->chip->dev; - unsigned int pipe = usb_rcvintpipe(dev, - SCARLETT2_USB_INTERRUPT_ENDPOINT); + struct scarlett2_data *private = mixer->private_data; + unsigned int pipe = usb_rcvintpipe(dev, private->bEndpointAddress); void *transfer_buffer; if (mixer->urb) { @@ -2067,14 +2104,13 @@ static int scarlett2_init_notify(struct usb_mixer_interface *mixer) if (!mixer->urb) return -ENOMEM; - transfer_buffer = kmalloc(SCARLETT2_USB_INTERRUPT_MAX_DATA, GFP_KERNEL); + transfer_buffer = kmalloc(private->wMaxPacketSize, GFP_KERNEL); if (!transfer_buffer) return -ENOMEM; usb_fill_int_urb(mixer->urb, dev, pipe, - transfer_buffer, SCARLETT2_USB_INTERRUPT_MAX_DATA, - scarlett2_notify, mixer, - SCARLETT2_USB_INTERRUPT_INTERVAL); + transfer_buffer, private->wMaxPacketSize, + scarlett2_notify, mixer, private->bInterval); return usb_submit_urb(mixer->urb, GFP_KERNEL); } @@ -2084,7 +2120,7 @@ static int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer, { int err; - /* Initialise private data and sequence number */ + /* Initialise private data, sequence number, and get the USB data */ err = scarlett2_init_private(mixer, info); if (err < 0) return err; -- 2.31.1

1 0

[PATCH 13/14] ALSA: usb-audio: scarlett2: Don't copy struct scarlett2_config
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

scarlett2_usb_set_config() and scarlett2_usb_get_config() were copying struct scarlett2_config. Use a pointer instead. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 49c4662531a3..45fd540920b9 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -706,8 +706,8 @@ static int scarlett2_usb_set_config( struct usb_mixer_interface *mixer, int config_item_num, int index, int value) { - const struct scarlett2_config config_item = - scarlett2_config_items[config_item_num]; + const struct scarlett2_config *config_item = + &scarlett2_config_items[config_item_num]; struct { __le32 offset; __le32 bytes; @@ -721,17 +721,17 @@ static int scarlett2_usb_set_config( cancel_delayed_work_sync(&private->work); /* Send the configuration parameter data */ - req.offset = cpu_to_le32(config_item.offset + index * config_item.size); - req.bytes = cpu_to_le32(config_item.size); + req.offset = cpu_to_le32(config_item->offset + index * config_item->size); + req.bytes = cpu_to_le32(config_item->size); req.value = cpu_to_le32(value); err = scarlett2_usb(mixer, SCARLETT2_USB_SET_DATA, - &req, sizeof(u32) * 2 + config_item.size, + &req, sizeof(u32) * 2 + config_item->size, NULL, 0); if (err < 0) return err; /* Activate the change */ - req2 = cpu_to_le32(config_item.activate); + req2 = cpu_to_le32(config_item->activate); err = scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, &req2, sizeof(req2), NULL, 0); if (err < 0) @@ -764,11 +764,11 @@ static int scarlett2_usb_get_config( struct usb_mixer_interface *mixer, int config_item_num, int count, void *buf) { - const struct scarlett2_config config_item = - scarlett2_config_items[config_item_num]; - int size = config_item.size * count; + const struct scarlett2_config *config_item = + &scarlett2_config_items[config_item_num]; + int size = config_item->size * count; - return scarlett2_usb_get(mixer, config_item.offset, buf, size); + return scarlett2_usb_get(mixer, config_item->offset, buf, size); } /* Send a USB message to get volume status; result placed in *buf */ -- 2.31.1

1 0

[PATCH 12/14] ALSA: usb-audio: scarlett2: Fix union usage in mixer control callbacks
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Fix mixer control callbacks to use the correct members of the struct snd_ctl_elem_value. The use of value.integer and value.enumerated were swapped in a few places. Update scarlett2_mux_src_enum_ctl_put() to use min() instead of clamp() as value.enumerated.item is unsigned. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index fe142b7ddb32..49c4662531a3 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -1270,7 +1270,7 @@ static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); oval = private->vol_sw_hw_switch[index]; - val = !!ucontrol->value.integer.value[0]; + val = !!ucontrol->value.enumerated.item[0]; if (oval == val) goto unlock; @@ -1356,7 +1356,7 @@ static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); oval = private->level_switch[index]; - val = !!ucontrol->value.integer.value[0]; + val = !!ucontrol->value.enumerated.item[0]; if (oval == val) goto unlock; @@ -1390,7 +1390,7 @@ static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_elem_info *elem = kctl->private_data; struct scarlett2_data *private = elem->head.mixer->private_data; - ucontrol->value.enumerated.item[0] = + ucontrol->value.integer.value[0] = private->pad_switch[elem->control]; return 0; } @@ -1448,7 +1448,7 @@ static int scarlett2_dim_mute_ctl_get(struct snd_kcontrol *kctl, scarlett2_update_volumes(mixer); mutex_unlock(&private->data_mutex); - ucontrol->value.enumerated.item[0] = private->dim_mute[elem->control]; + ucontrol->value.integer.value[0] = private->dim_mute[elem->control]; return 0; } @@ -1748,8 +1748,8 @@ static int scarlett2_mux_src_enum_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); oval = private->mux[index]; - val = clamp(ucontrol->value.integer.value[0], - 0L, private->num_mux_srcs - 1L); + val = min(ucontrol->value.enumerated.item[0], + private->num_mux_srcs - 1U); if (oval == val) goto unlock; -- 2.31.1

1 0

[PATCH 11/14] ALSA: usb-audio: scarlett2: Fix scarlett2_*_ctl_put() return values
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Mixer control put callbacks should return 1 if the value is changed. Fix the sw_hw, level, pad, and button controls accordingly. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index d6298ab1de56..fe142b7ddb32 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -1304,6 +1304,8 @@ static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, /* Send SW/HW switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); @@ -1364,6 +1366,8 @@ static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, /* Send switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); @@ -1414,6 +1418,8 @@ static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl, /* Send switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_PAD_SWITCH, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); @@ -1469,6 +1475,8 @@ static int scarlett2_dim_mute_ctl_put(struct snd_kcontrol *kctl, /* Send switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_DIM_MUTE, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); -- 2.31.1

1 0

[PATCH 10/14] ALSA: usb-audio: scarlett2: Fix data_mutex lock
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

The private->vol_updated flag was being checked outside of the mutex_lock/unlock() of private->data_mutex leading to the volume data being fetched twice from the device unnecessarily or old volume data being returned. Update scarlett2_*_ctl_get() and include the private->vol_updated flag check inside the critical region. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 21 +++++++++------------ 1 file changed, 9 insertions(+), 12 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index cb8483d3c58a..d6298ab1de56 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -1153,11 +1153,10 @@ static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - if (private->vol_updated) { - mutex_lock(&private->data_mutex); + mutex_lock(&private->data_mutex); + if (private->vol_updated) scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); - } + mutex_unlock(&private->data_mutex); ucontrol->value.integer.value[0] = private->master_vol; return 0; @@ -1171,11 +1170,10 @@ static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, struct scarlett2_data *private = mixer->private_data; int index = elem->control; - if (private->vol_updated) { - mutex_lock(&private->data_mutex); + mutex_lock(&private->data_mutex); + if (private->vol_updated) scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); - } + mutex_unlock(&private->data_mutex); ucontrol->value.integer.value[0] = private->vol[index]; return 0; @@ -1439,11 +1437,10 @@ static int scarlett2_dim_mute_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - if (private->vol_updated) { - mutex_lock(&private->data_mutex); + mutex_lock(&private->data_mutex); + if (private->vol_updated) scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); - } + mutex_unlock(&private->data_mutex); ucontrol->value.enumerated.item[0] = private->dim_mute[elem->control]; return 0; -- 2.31.1

1 0

[PATCH 09/14] ALSA: usb-audio: scarlett2: Add temp variable for consistency
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Add index temporary variable to scarlett2_mixer_ctl_put() for consistency with the other *_ctl_put() functions. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index c1e74918425a..cb8483d3c58a 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -1622,18 +1622,19 @@ static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, const struct scarlett2_device_info *info = private->info; const struct scarlett2_ports *ports = info->ports; int oval, val, num_mixer_in, mix_num, err = 0; + int index = elem->control; mutex_lock(&private->data_mutex); - oval = private->mix[elem->control]; + oval = private->mix[index]; val = ucontrol->value.integer.value[0]; num_mixer_in = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; - mix_num = elem->control / num_mixer_in; + mix_num = index / num_mixer_in; if (oval == val) goto unlock; - private->mix[elem->control] = val; + private->mix[index] = val; err = scarlett2_usb_set_mix(mixer, mix_num); if (err == 0) err = 1; -- 2.31.1

1 0

[PATCH 08/14] ALSA: usb-audio: scarlett2: Rename struct scarlett2_mixer_data
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Rename struct scarlett2_mixer_data to struct scarlett2_data. A less-wordy name is better because it is used everywhere, and although this is a mixer driver, it also controls other vendor-specific features. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 80 ++++++++++++++++----------------- 1 file changed, 40 insertions(+), 40 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 4e2ee979b9bd..c1e74918425a 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -224,7 +224,7 @@ struct scarlett2_device_info { struct scarlett2_ports ports[SCARLETT2_PORT_TYPE_COUNT]; }; -struct scarlett2_mixer_data { +struct scarlett2_data { struct usb_mixer_interface *mixer; struct mutex usb_mutex; /* prevent sending concurrent USB requests */ struct mutex data_mutex; /* lock access to this data */ @@ -564,7 +564,7 @@ struct scarlett2_usb_packet { u8 data[]; }; -static void scarlett2_fill_request_header(struct scarlett2_mixer_data *private, +static void scarlett2_fill_request_header(struct scarlett2_data *private, struct scarlett2_usb_packet *req, u32 cmd, u16 req_size) { @@ -583,7 +583,7 @@ static int scarlett2_usb( struct usb_mixer_interface *mixer, u32 cmd, void *req_data, u16 req_size, void *resp_data, u16 resp_size) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; u16 req_buf_size = sizeof(struct scarlett2_usb_packet) + req_size; u16 resp_buf_size = sizeof(struct scarlett2_usb_packet) + resp_size; struct scarlett2_usb_packet *req, *resp = NULL; @@ -693,8 +693,8 @@ static void scarlett2_config_save(struct usb_mixer_interface *mixer) /* Delayed work to save config */ static void scarlett2_config_save_work(struct work_struct *work) { - struct scarlett2_mixer_data *private = - container_of(work, struct scarlett2_mixer_data, work.work); + struct scarlett2_data *private = + container_of(work, struct scarlett2_data, work.work); scarlett2_config_save(private->mixer); } @@ -715,7 +715,7 @@ static int scarlett2_usb_set_config( } __packed req; __le32 req2; int err; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; /* Cancel any pending NVRAM save */ cancel_delayed_work_sync(&private->work); @@ -786,7 +786,7 @@ static int scarlett2_usb_get_volume_status( static int scarlett2_usb_get_mix(struct usb_mixer_interface *mixer, int mix_num) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; int num_mixer_in = @@ -829,7 +829,7 @@ static int scarlett2_usb_get_mix(struct usb_mixer_interface *mixer, static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer, int mix_num) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; struct { @@ -895,7 +895,7 @@ static u32 scarlett2_mux_id_to_num(const struct scarlett2_ports *ports, } /* Convert one mux entry from the interface and load into private->mux[] */ -static void scarlett2_usb_populate_mux(struct scarlett2_mixer_data *private, +static void scarlett2_usb_populate_mux(struct scarlett2_data *private, u32 mux_entry) { const struct scarlett2_device_info *info = private->info; @@ -933,7 +933,7 @@ static void scarlett2_usb_populate_mux(struct scarlett2_mixer_data *private, /* Send USB message to get mux inputs and then populate private->mux[] */ static int scarlett2_usb_get_mux(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int count = private->num_mux_dsts; int err, i; @@ -962,7 +962,7 @@ static int scarlett2_usb_get_mux(struct usb_mixer_interface *mixer) /* Send USB messages to set mux inputs */ static int scarlett2_usb_set_mux(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; const struct scarlett2_ports *ports = info->ports; int rate, port_dir_rate; @@ -1103,7 +1103,7 @@ static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer, */ static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; const struct scarlett2_ports *ports = info->ports; struct scarlett2_usb_volume_status volume_status; @@ -1151,7 +1151,7 @@ static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; if (private->vol_updated) { mutex_lock(&private->data_mutex); @@ -1168,7 +1168,7 @@ static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int index = elem->control; if (private->vol_updated) { @@ -1186,7 +1186,7 @@ static int scarlett2_volume_ctl_put(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int index = elem->control; int oval, val, err = 0; @@ -1252,7 +1252,7 @@ static int scarlett2_sw_hw_enum_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; ucontrol->value.enumerated.item[0] = private->vol_sw_hw_switch[elem->control]; @@ -1264,7 +1264,7 @@ static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int index = elem->control; int oval, val, err = 0; @@ -1336,7 +1336,7 @@ static int scarlett2_level_enum_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; ucontrol->value.enumerated.item[0] = private->level_switch[elem->control]; @@ -1348,7 +1348,7 @@ static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int index = elem->control; int oval, val, err = 0; @@ -1386,7 +1386,7 @@ static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; ucontrol->value.enumerated.item[0] = private->pad_switch[elem->control]; @@ -1398,7 +1398,7 @@ static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int index = elem->control; int oval, val, err = 0; @@ -1437,7 +1437,7 @@ static int scarlett2_dim_mute_ctl_get(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; if (private->vol_updated) { mutex_lock(&private->data_mutex); @@ -1454,7 +1454,7 @@ static int scarlett2_dim_mute_ctl_put(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int index = elem->control; int oval, val, err = 0; @@ -1490,7 +1490,7 @@ static const struct snd_kcontrol_new scarlett2_dim_mute_ctl = { static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; const struct scarlett2_ports *ports = info->ports; int num_line_out = @@ -1562,7 +1562,7 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; int err, i; char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; @@ -1607,7 +1607,7 @@ static int scarlett2_mixer_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; ucontrol->value.integer.value[0] = private->mix[elem->control]; return 0; @@ -1618,7 +1618,7 @@ static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; const struct scarlett2_ports *ports = info->ports; int oval, val, num_mixer_in, mix_num, err = 0; @@ -1663,7 +1663,7 @@ static const struct snd_kcontrol_new scarlett2_mixer_ctl = { static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_ports *ports = private->info->ports; int err, i, j; int index; @@ -1692,7 +1692,7 @@ static int scarlett2_mux_src_enum_ctl_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; const struct scarlett2_ports *ports = private->info->ports; unsigned int item = uinfo->value.enumerated.item; int items = private->num_mux_srcs; @@ -1724,7 +1724,7 @@ static int scarlett2_mux_src_enum_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; ucontrol->value.enumerated.item[0] = private->mux[elem->control]; return 0; @@ -1735,7 +1735,7 @@ static int scarlett2_mux_src_enum_ctl_put(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int index = elem->control; int oval, val, err = 0; @@ -1768,7 +1768,7 @@ static const struct snd_kcontrol_new scarlett2_mux_src_enum_ctl = { static int scarlett2_add_mux_enums(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_ports *ports = private->info->ports; int port_type, channel, i; @@ -1847,7 +1847,7 @@ static int scarlett2_add_meter_ctl(struct usb_mixer_interface *mixer) static void scarlett2_private_free(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; cancel_delayed_work_sync(&private->work); kfree(private); @@ -1856,7 +1856,7 @@ static void scarlett2_private_free(struct usb_mixer_interface *mixer) static void scarlett2_private_suspend(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; if (cancel_delayed_work_sync(&private->work)) scarlett2_config_save(private->mixer); @@ -1864,7 +1864,7 @@ static void scarlett2_private_suspend(struct usb_mixer_interface *mixer) /*** Initialisation ***/ -static void scarlett2_count_mux_io(struct scarlett2_mixer_data *private) +static void scarlett2_count_mux_io(struct scarlett2_data *private) { const struct scarlett2_ports *ports = private->info->ports; int port_type, srcs = 0, dsts = 0; @@ -1884,8 +1884,8 @@ static void scarlett2_count_mux_io(struct scarlett2_mixer_data *private) static int scarlett2_init_private(struct usb_mixer_interface *mixer, const struct scarlett2_device_info *info) { - struct scarlett2_mixer_data *private = - kzalloc(sizeof(struct scarlett2_mixer_data), GFP_KERNEL); + struct scarlett2_data *private = + kzalloc(sizeof(struct scarlett2_data), GFP_KERNEL); if (!private) return -ENOMEM; @@ -1908,7 +1908,7 @@ static int scarlett2_init_private(struct usb_mixer_interface *mixer, /* Read configuration from the interface on start */ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; const struct scarlett2_ports *ports = info->ports; int num_line_out = @@ -1978,7 +1978,7 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) static void scarlett2_notify_monitor( struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_ports *ports = private->info->ports; int num_line_out = ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; @@ -2001,7 +2001,7 @@ static void scarlett2_notify_monitor( static void scarlett2_notify_dim_mute( struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; int i; -- 2.31.1

1 0

[PATCH 07/14] ALSA: usb-audio: scarlett2: Rename buttons/interrupts/vol
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

To match the vendor's terminology, change #defines, identifiers, and comments: - mute/dim/hardware buttons are now called dim/mute - mixer status/interrupt is now notify - vol is now monitor Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 94 ++++++++++++++++----------------- 1 file changed, 47 insertions(+), 47 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 043b0929ad9d..4e2ee979b9bd 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -194,10 +194,10 @@ enum { SCARLETT2_PORT_DIRNS = 5, }; -/* Hardware buttons on the 18i20 */ -#define SCARLETT2_BUTTON_MAX 2 +/* Dim/Mute buttons on the 18i20 */ +#define SCARLETT2_DIM_MUTE_COUNT 2 -static const char *const scarlett2_button_names[SCARLETT2_BUTTON_MAX] = { +static const char *const scarlett2_dim_mute_names[SCARLETT2_DIM_MUTE_COUNT] = { "Mute", "Dim" }; @@ -239,10 +239,10 @@ struct scarlett2_mixer_data { u8 vol_sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; u8 level_switch[SCARLETT2_LEVEL_SWITCH_MAX]; u8 pad_switch[SCARLETT2_PAD_SWITCH_MAX]; - u8 buttons[SCARLETT2_BUTTON_MAX]; + u8 dim_mute[SCARLETT2_DIM_MUTE_COUNT]; struct snd_kcontrol *master_vol_ctl; struct snd_kcontrol *vol_ctls[SCARLETT2_ANALOGUE_MAX]; - struct snd_kcontrol *button_ctls[SCARLETT2_BUTTON_MAX]; + struct snd_kcontrol *dim_mute_ctls[SCARLETT2_DIM_MUTE_COUNT]; u8 mux[SCARLETT2_MUX_MAX]; u8 mix[SCARLETT2_INPUT_MIX_MAX * SCARLETT2_OUTPUT_MIX_MAX]; }; @@ -450,9 +450,9 @@ static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, #define SCARLETT2_USB_INTERRUPT_MAX_DATA 64 #define SCARLETT2_USB_INTERRUPT_INTERVAL 3 -/* Interrupt flags for volume and mute/dim button changes */ -#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x00400000 -#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x00200000 +/* Interrupt flags for dim/mute button and monitor changes */ +#define SCARLETT2_USB_NOTIFY_DIM_MUTE 0x00200000 +#define SCARLETT2_USB_NOTIFY_MONITOR 0x00400000 /* Commands for sending/receiving requests/responses */ #define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 @@ -475,8 +475,8 @@ static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, /* volume status is read together (matches scarlett2_config_items[]) */ struct scarlett2_usb_volume_status { - /* mute & dim buttons */ - u8 buttons[SCARLETT2_BUTTON_MAX]; + /* dim/mute buttons */ + u8 dim_mute[SCARLETT2_DIM_MUTE_COUNT]; u8 pad1; @@ -499,7 +499,7 @@ struct scarlett2_usb_volume_status { /* Configuration parameters that can be read and written */ enum { - SCARLETT2_CONFIG_BUTTONS = 0, + SCARLETT2_CONFIG_DIM_MUTE = 0, SCARLETT2_CONFIG_LINE_OUT_VOLUME = 1, SCARLETT2_CONFIG_SW_HW_SWITCH = 2, SCARLETT2_CONFIG_LEVEL_SWITCH = 3, @@ -518,7 +518,7 @@ struct scarlett2_config { static const struct scarlett2_config scarlett2_config_items[SCARLETT2_CONFIG_COUNT] = { - /* Mute/Dim Buttons */ + /* Dim/Mute Buttons */ { .offset = 0x31, .size = 1, @@ -1127,8 +1127,8 @@ static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) } if (info->line_out_hw_vol) - for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) - private->buttons[i] = !!volume_status.buttons[i]; + for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) + private->dim_mute[i] = !!volume_status.dim_mute[i]; return 0; } @@ -1430,10 +1430,10 @@ static const struct snd_kcontrol_new scarlett2_pad_ctl = { .put = scarlett2_pad_ctl_put, }; -/*** Mute/Dim Controls ***/ +/*** Dim/Mute Controls ***/ -static int scarlett2_button_ctl_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int scarlett2_dim_mute_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; @@ -1445,12 +1445,12 @@ static int scarlett2_button_ctl_get(struct snd_kcontrol *kctl, mutex_unlock(&private->data_mutex); } - ucontrol->value.enumerated.item[0] = private->buttons[elem->control]; + ucontrol->value.enumerated.item[0] = private->dim_mute[elem->control]; return 0; } -static int scarlett2_button_ctl_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int scarlett2_dim_mute_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; @@ -1461,16 +1461,16 @@ static int scarlett2_button_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); - oval = private->buttons[index]; + oval = private->dim_mute[index]; val = !!ucontrol->value.integer.value[0]; if (oval == val) goto unlock; - private->buttons[index] = val; + private->dim_mute[index] = val; /* Send switch change to the device */ - err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_BUTTONS, + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_DIM_MUTE, index, val); unlock: @@ -1478,12 +1478,12 @@ static int scarlett2_button_ctl_put(struct snd_kcontrol *kctl, return err; } -static const struct snd_kcontrol_new scarlett2_button_ctl = { +static const struct snd_kcontrol_new scarlett2_dim_mute_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", .info = snd_ctl_boolean_mono_info, - .get = scarlett2_button_ctl_get, - .put = scarlett2_button_ctl_put + .get = scarlett2_dim_mute_ctl_get, + .put = scarlett2_dim_mute_ctl_put }; /*** Create the analogue output controls ***/ @@ -1544,13 +1544,13 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) } } - /* Add HW button controls */ + /* Add dim/mute controls */ if (info->line_out_hw_vol) - for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) { + for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) { err = scarlett2_add_new_ctl( - mixer, &scarlett2_button_ctl, - i, 1, scarlett2_button_names[i], - &private->button_ctls[i]); + mixer, &scarlett2_dim_mute_ctl, + i, 1, scarlett2_dim_mute_names[i], + &private->dim_mute_ctls[i]); if (err < 0) return err; } @@ -1962,8 +1962,8 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) } if (info->line_out_hw_vol) - for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) - private->buttons[i] = !!volume_status.buttons[i]; + for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) + private->dim_mute[i] = !!volume_status.dim_mute[i]; for (i = 0; i < num_mixer_out; i++) { err = scarlett2_usb_get_mix(mixer, i); @@ -1974,8 +1974,8 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) return scarlett2_usb_get_mux(mixer); } -/* Notify on volume change */ -static void scarlett2_mixer_interrupt_vol_change( +/* Notify on monitor change */ +static void scarlett2_notify_monitor( struct usb_mixer_interface *mixer) { struct scarlett2_mixer_data *private = mixer->private_data; @@ -1997,8 +1997,8 @@ static void scarlett2_mixer_interrupt_vol_change( } } -/* Notify on button change */ -static void scarlett2_mixer_interrupt_button_change( +/* Notify on dim/mute change */ +static void scarlett2_notify_dim_mute( struct usb_mixer_interface *mixer) { struct scarlett2_mixer_data *private = mixer->private_data; @@ -2010,13 +2010,13 @@ static void scarlett2_mixer_interrupt_button_change( if (!info->line_out_hw_vol) return; - for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) + for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->button_ctls[i]->id); + &private->dim_mute_ctls[i]->id); } /* Interrupt callback */ -static void scarlett2_mixer_interrupt(struct urb *urb) +static void scarlett2_notify(struct urb *urb) { struct usb_mixer_interface *mixer = urb->context; int len = urb->actual_length; @@ -2027,10 +2027,10 @@ static void scarlett2_mixer_interrupt(struct urb *urb) goto requeue; data = le32_to_cpu(*(__le32 *)urb->transfer_buffer); - if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) - scarlett2_mixer_interrupt_vol_change(mixer); - if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) - scarlett2_mixer_interrupt_button_change(mixer); + if (data & SCARLETT2_USB_NOTIFY_MONITOR) + scarlett2_notify_monitor(mixer); + if (data & SCARLETT2_USB_NOTIFY_DIM_MUTE) + scarlett2_notify_dim_mute(mixer); requeue: if (ustatus != -ENOENT && @@ -2041,7 +2041,7 @@ static void scarlett2_mixer_interrupt(struct urb *urb) } } -static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer) +static int scarlett2_init_notify(struct usb_mixer_interface *mixer) { struct usb_device *dev = mixer->chip->dev; unsigned int pipe = usb_rcvintpipe(dev, @@ -2067,7 +2067,7 @@ static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer) usb_fill_int_urb(mixer->urb, dev, pipe, transfer_buffer, SCARLETT2_USB_INTERRUPT_MAX_DATA, - scarlett2_mixer_interrupt, mixer, + scarlett2_notify, mixer, SCARLETT2_USB_INTERRUPT_INTERVAL); return usb_submit_urb(mixer->urb, GFP_KERNEL); @@ -2115,7 +2115,7 @@ static int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer, /* Set up the interrupt polling if there are hardware buttons */ if (info->line_out_hw_vol) { - err = scarlett2_mixer_status_create(mixer); + err = scarlett2_init_notify(mixer); if (err < 0) return err; } -- 2.31.1

1 0

[PATCH 06/14] ALSA: usb-audio: scarlett2: Remove redundant info->button_count
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

The per-model button_count value was used to determine whether dim/mute controls should be added, but these are present iff line_out_hw_vol is true. Remove button_count and replace with SCARLETT2_BUTTON_MAX and a check for line_out_hw_vol true. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 41 ++++++++++++++++++--------------- 1 file changed, 23 insertions(+), 18 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 3225ec709d98..043b0929ad9d 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -218,7 +218,6 @@ struct scarlett2_ports { struct scarlett2_device_info { u8 line_out_hw_vol; /* line out hw volume is sw controlled */ - u8 button_count; /* number of buttons */ u8 level_input_count; /* inputs with level selectable */ u8 pad_input_count; /* inputs with pad selectable */ const char * const line_out_descrs[SCARLETT2_ANALOGUE_MAX]; @@ -370,9 +369,6 @@ static const struct scarlett2_device_info s18i20_gen2_info = { */ .line_out_hw_vol = 1, - /* Mute and dim buttons */ - .button_count = 2, - .line_out_descrs = { "Monitor L", "Monitor R", @@ -1108,7 +1104,8 @@ static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer, static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) { struct scarlett2_mixer_data *private = mixer->private_data; - const struct scarlett2_ports *ports = private->info->ports; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; struct scarlett2_usb_volume_status volume_status; int num_line_out = ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; @@ -1129,8 +1126,9 @@ static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) private->vol[i] = private->master_vol; } - for (i = 0; i < private->info->button_count; i++) - private->buttons[i] = !!volume_status.buttons[i]; + if (info->line_out_hw_vol) + for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) + private->buttons[i] = !!volume_status.buttons[i]; return 0; } @@ -1547,13 +1545,15 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) } /* Add HW button controls */ - for (i = 0; i < private->info->button_count; i++) { - err = scarlett2_add_new_ctl(mixer, &scarlett2_button_ctl, - i, 1, scarlett2_button_names[i], - &private->button_ctls[i]); - if (err < 0) - return err; - } + if (info->line_out_hw_vol) + for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) { + err = scarlett2_add_new_ctl( + mixer, &scarlett2_button_ctl, + i, 1, scarlett2_button_names[i], + &private->button_ctls[i]); + if (err < 0) + return err; + } return 0; } @@ -1961,8 +1961,9 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) private->vol[i] = volume; } - for (i = 0; i < info->button_count; i++) - private->buttons[i] = !!volume_status.buttons[i]; + if (info->line_out_hw_vol) + for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) + private->buttons[i] = !!volume_status.buttons[i]; for (i = 0; i < num_mixer_out; i++) { err = scarlett2_usb_get_mix(mixer, i); @@ -2001,11 +2002,15 @@ static void scarlett2_mixer_interrupt_button_change( struct usb_mixer_interface *mixer) { struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; int i; private->vol_updated = 1; - for (i = 0; i < private->info->button_count; i++) + if (!info->line_out_hw_vol) + return; + + for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &private->button_ctls[i]->id); } @@ -2109,7 +2114,7 @@ static int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer, return err; /* Set up the interrupt polling if there are hardware buttons */ - if (info->button_count) { + if (info->line_out_hw_vol) { err = scarlett2_mixer_status_create(mixer); if (err < 0) return err; -- 2.31.1

1 0

[PATCH 05/14] ALSA: usb-audio: scarlett2: Remove interrupt debug message
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Just ignore instead of printing an error if the interrupt data is not the expected length. This check was for development and the condition has not been observed. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index a5f5c537e344..3225ec709d98 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -2018,19 +2018,14 @@ static void scarlett2_mixer_interrupt(struct urb *urb) int ustatus = urb->status; u32 data; - if (ustatus != 0) + if (ustatus != 0 || len != 8) goto requeue; - if (len == 8) { - data = le32_to_cpu(*(__le32 *)urb->transfer_buffer); - if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) - scarlett2_mixer_interrupt_vol_change(mixer); - if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) - scarlett2_mixer_interrupt_button_change(mixer); - } else { - usb_audio_err(mixer->chip, - "scarlett mixer interrupt length %d\n", len); - } + data = le32_to_cpu(*(__le32 *)urb->transfer_buffer); + if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) + scarlett2_mixer_interrupt_vol_change(mixer); + if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) + scarlett2_mixer_interrupt_button_change(mixer); requeue: if (ustatus != -ENOENT && -- 2.31.1

1 0

[PATCH 04/14] ALSA: usb-audio: scarlett2: Remove unused/useless code
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Remove #define not used. Remove useless assignments and copies. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 26 +++++--------------------- 1 file changed, 5 insertions(+), 21 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 8026847d27b6..a5f5c537e344 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -271,7 +271,6 @@ static const struct scarlett2_device_info s6i6_gen2_info = { .id = 0x000, .num = { 1, 0, 8, 8, 8 }, .src_descr = "Off", - .src_num_offset = 0, }, [SCARLETT2_PORT_TYPE_ANALOGUE] = { .id = 0x080, @@ -327,7 +326,6 @@ static const struct scarlett2_device_info s18i8_gen2_info = { .id = 0x000, .num = { 1, 0, 8, 8, 4 }, .src_descr = "Off", - .src_num_offset = 0, }, [SCARLETT2_PORT_TYPE_ANALOGUE] = { .id = 0x080, @@ -393,7 +391,6 @@ static const struct scarlett2_device_info s18i20_gen2_info = { .id = 0x000, .num = { 1, 0, 8, 8, 6 }, .src_descr = "Off", - .src_num_offset = 0, }, [SCARLETT2_PORT_TYPE_ANALOGUE] = { .id = 0x080, @@ -571,8 +568,6 @@ struct scarlett2_usb_packet { u8 data[]; }; -#define SCARLETT2_USB_PACKET_LEN (sizeof(struct scarlett2_usb_packet)) - static void scarlett2_fill_request_header(struct scarlett2_mixer_data *private, struct scarlett2_usb_packet *req, u32 cmd, u16 req_size) @@ -595,8 +590,8 @@ static int scarlett2_usb( struct scarlett2_mixer_data *private = mixer->private_data; u16 req_buf_size = sizeof(struct scarlett2_usb_packet) + req_size; u16 resp_buf_size = sizeof(struct scarlett2_usb_packet) + resp_size; - struct scarlett2_usb_packet *req = NULL, *resp = NULL; - int err = 0; + struct scarlett2_usb_packet *req, *resp = NULL; + int err; req = kmalloc(req_buf_size, GFP_KERNEL); if (!req) { @@ -1026,10 +1021,6 @@ static int scarlett2_usb_set_mux(struct usb_mixer_interface *mixer) ports, private->mux[j] ) << 12 ); - - /* skip private->mux[j] entries not output */ - j += ports[port_type].num[SCARLETT2_PORT_OUT] - - ports[port_type].num[port_dir_rate]; } err = scarlett2_usb(mixer, SCARLETT2_USB_SET_MUX, @@ -1681,7 +1672,7 @@ static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) int num_inputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; int num_outputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_IN]; - for (i = 0, index = 0; i < num_outputs; i++) { + for (i = 0, index = 0; i < num_outputs; i++) for (j = 0; j < num_inputs; j++, index++) { snprintf(s, sizeof(s), "Mix %c Input %02d Playback Volume", @@ -1691,7 +1682,6 @@ static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) if (err < 0) return err; } - } return 0; } @@ -1925,8 +1915,6 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; int num_mixer_out = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_IN]; - u8 level_switches[SCARLETT2_LEVEL_SWITCH_MAX]; - u8 pad_switches[SCARLETT2_PAD_SWITCH_MAX]; struct scarlett2_usb_volume_status volume_status; int err, i; @@ -1935,11 +1923,9 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, info->level_input_count, - level_switches); + private->level_switch); if (err < 0) return err; - for (i = 0; i < info->level_input_count; i++) - private->level_switch[i] = level_switches[i]; } if (info->pad_input_count) { @@ -1947,11 +1933,9 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) mixer, SCARLETT2_CONFIG_PAD_SWITCH, info->pad_input_count, - pad_switches); + private->pad_switch); if (err < 0) return err; - for (i = 0; i < info->pad_input_count; i++) - private->pad_switch[i] = pad_switches[i]; } err = scarlett2_usb_get_volume_status(mixer, &volume_status); -- 2.31.1

1 0

[PATCH 03/14] ALSA: usb-audio: scarlett2: Coding style improvements
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Improve alignment and readability with: - Whitespace fixes - Add leading zeros to 32-bit flag values - Rename SCARLETT2_USB_GET_METER_LEVELS to SCARLETT2_USB_GET_METER - Rename SCARLETT2_PORT_DIRECTIONS to SCARLETT2_PORT_DIRNS Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 53 +++++++++++++++++---------------- 1 file changed, 27 insertions(+), 26 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index e327f15b89ce..8026847d27b6 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -175,23 +175,23 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = { * - PCM I/O */ enum { - SCARLETT2_PORT_TYPE_NONE = 0, + SCARLETT2_PORT_TYPE_NONE = 0, SCARLETT2_PORT_TYPE_ANALOGUE = 1, - SCARLETT2_PORT_TYPE_SPDIF = 2, - SCARLETT2_PORT_TYPE_ADAT = 3, - SCARLETT2_PORT_TYPE_MIX = 4, - SCARLETT2_PORT_TYPE_PCM = 5, - SCARLETT2_PORT_TYPE_COUNT = 6, + SCARLETT2_PORT_TYPE_SPDIF = 2, + SCARLETT2_PORT_TYPE_ADAT = 3, + SCARLETT2_PORT_TYPE_MIX = 4, + SCARLETT2_PORT_TYPE_PCM = 5, + SCARLETT2_PORT_TYPE_COUNT = 6, }; /* Count of total I/O and number available at each sample rate */ enum { - SCARLETT2_PORT_IN = 0, - SCARLETT2_PORT_OUT = 1, - SCARLETT2_PORT_OUT_44 = 2, - SCARLETT2_PORT_OUT_88 = 3, + SCARLETT2_PORT_IN = 0, + SCARLETT2_PORT_OUT = 1, + SCARLETT2_PORT_OUT_44 = 2, + SCARLETT2_PORT_OUT_88 = 3, SCARLETT2_PORT_OUT_176 = 4, - SCARLETT2_PORT_DIRECTIONS = 5, + SCARLETT2_PORT_DIRNS = 5, }; /* Hardware buttons on the 18i20 */ @@ -210,7 +210,7 @@ static const char *const scarlett2_button_names[SCARLETT2_BUTTON_MAX] = { */ struct scarlett2_ports { u16 id; - int num[SCARLETT2_PORT_DIRECTIONS]; + int num[SCARLETT2_PORT_DIRNS]; const char * const src_descr; int src_num_offset; const char * const dst_descr; @@ -458,22 +458,23 @@ static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, #define SCARLETT2_USB_INTERRUPT_INTERVAL 3 /* Interrupt flags for volume and mute/dim button changes */ -#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x400000 -#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x200000 +#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x00400000 +#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x00200000 /* Commands for sending/receiving requests/responses */ -#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 #define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP 3 -#define SCARLETT2_USB_INIT_SEQ 0x00000000 -#define SCARLETT2_USB_GET_METER_LEVELS 0x00001001 -#define SCARLETT2_USB_GET_MIX 0x00002001 -#define SCARLETT2_USB_SET_MIX 0x00002002 -#define SCARLETT2_USB_GET_MUX 0x00003001 -#define SCARLETT2_USB_SET_MUX 0x00003002 -#define SCARLETT2_USB_GET_DATA 0x00800000 -#define SCARLETT2_USB_SET_DATA 0x00800001 -#define SCARLETT2_USB_DATA_CMD 0x00800002 +#define SCARLETT2_USB_INIT_SEQ 0x00000000 +#define SCARLETT2_USB_GET_METER 0x00001001 +#define SCARLETT2_USB_GET_MIX 0x00002001 +#define SCARLETT2_USB_SET_MIX 0x00002002 +#define SCARLETT2_USB_GET_MUX 0x00003001 +#define SCARLETT2_USB_SET_MUX 0x00003002 +#define SCARLETT2_USB_GET_DATA 0x00800000 +#define SCARLETT2_USB_SET_DATA 0x00800001 +#define SCARLETT2_USB_DATA_CMD 0x00800002 + #define SCARLETT2_USB_CONFIG_SAVE 6 #define SCARLETT2_USB_VOLUME_STATUS_OFFSET 0x31 @@ -835,7 +836,7 @@ static int scarlett2_usb_get_mix(struct usb_mixer_interface *mixer, * (values obtained from private->mix[]) */ static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer, - int mix_num) + int mix_num) { struct scarlett2_mixer_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; @@ -1056,7 +1057,7 @@ static int scarlett2_usb_get_meter_levels(struct usb_mixer_interface *mixer, req.pad = 0; req.num_meters = cpu_to_le16(SCARLETT2_NUM_METERS); req.magic = cpu_to_le32(SCARLETT2_USB_METER_LEVELS_GET_MAGIC); - err = scarlett2_usb(mixer, SCARLETT2_USB_GET_METER_LEVELS, + err = scarlett2_usb(mixer, SCARLETT2_USB_GET_METER, &req, sizeof(req), resp, sizeof(resp)); if (err < 0) return err; -- 2.31.1

1 0

[PATCH 02/14] ALSA: usb-audio: scarlett2: Fix 18i8 Gen 2 PCM Input count
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

The 18i8 Gen 2 has 8 PCM Inputs, not 20. Fix the ports entry in s18i8_gen2_info. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 962050436a9f..e327f15b89ce 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -358,7 +358,7 @@ static const struct scarlett2_device_info s18i8_gen2_info = { }, [SCARLETT2_PORT_TYPE_PCM] = { .id = 0x600, - .num = { 20, 18, 18, 14, 10 }, + .num = { 8, 18, 18, 14, 10 }, .src_descr = "PCM %d", .src_num_offset = 1, .dst_descr = "PCM %02d Capture" -- 2.31.1

1 0

[PATCH 01/14] ALSA: usb-audio: scarlett2: Remove incorrect S/PDIF comment
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

The 18i8 Gen 2 S/PDIF outputs are available at 192kHz, unlike the 18i20 Gen 2. Remove the comment that says otherwise. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index a461317dc8c6..962050436a9f 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -338,10 +338,6 @@ static const struct scarlett2_device_info s18i8_gen2_info = { }, [SCARLETT2_PORT_TYPE_SPDIF] = { .id = 0x180, - /* S/PDIF outputs aren't available at 192kHz - * but are included in the USB mux I/O - * assignment message anyway - */ .num = { 2, 2, 2, 2, 2 }, .src_descr = "S/PDIF %d", .src_num_offset = 1, -- 2.31.1

1 0

[PATCH AUTOSEL 5.12 11/33] ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms
by Sasha Levin 20 Jun '21

20 Jun '21

From: Mark Pearson <markpearson(a)lenovo.com> [ Upstream commit 19a0aa9b04c5ab9a063b6ceaf7211ee7d9a9d24d ] More laptops identified where the AMD ACP bridge needs to be blocked or the microphone will not work when connected to HDMI. Use DMI to block the microphone PCM device for these platforms. Suggested-by: Gabriel Craciunescu <nix.or.die(a)gmail.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> Link: https://lore.kernel.org/r/20210531145502.6079-1-markpearson@lenovo.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 +++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 050a61fe9693..47a4dfd81a46 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -199,6 +199,41 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, + { + /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20Y1"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UH"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 Campus */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UJ"), + } + }, + { + /* Lenovo ThinkPad T14 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UD"), + } + }, + { + /* Lenovo ThinkPad X13 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UF"), + } + }, {} }; -- 2.30.2

2 2

[PATCH] soundwire: stream: Fix test for DP prepare complete
by Richard Fitzgerald 20 Jun '21

20 Jun '21

In sdw_prep_deprep_slave_ports(), after the wait_for_completion() the DP prepare status register is read. If this indicates that the port is now prepared, the code should continue with the port setup. It is irrelevant whether the wait_for_completion() timed out if the port is now ready. The previous implementation would always fail if the wait_for_completion() timed out, even if the port was reporting successful prepare. This patch also fixes a minor bug where the return from sdw_read() was not checked for error - any error code with LSBits clear could be misinterpreted as a successful port prepare. Fixes: 79df15b7d37c ("soundwire: Add helpers for ports operations") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/soundwire/stream.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 1eaedaaba094..1a18308f4ef4 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -422,7 +422,6 @@ static int sdw_prep_deprep_slave_ports(struct sdw_bus *bus, struct completion *port_ready; struct sdw_dpn_prop *dpn_prop; struct sdw_prepare_ch prep_ch; - unsigned int time_left; bool intr = false; int ret = 0, val; u32 addr; @@ -479,15 +478,15 @@ static int sdw_prep_deprep_slave_ports(struct sdw_bus *bus, /* Wait for completion on port ready */ port_ready = &s_rt->slave->port_ready[prep_ch.num]; - time_left = wait_for_completion_timeout(port_ready, - msecs_to_jiffies(dpn_prop->ch_prep_timeout)); + wait_for_completion_timeout(port_ready, + msecs_to_jiffies(dpn_prop->ch_prep_timeout)); val = sdw_read(s_rt->slave, SDW_DPN_PREPARESTATUS(p_rt->num)); - val &= p_rt->ch_mask; - if (!time_left || val) { + if ((val < 0) || (val & p_rt->ch_mask)) { + ret = (val < 0) ? val : -ETIMEDOUT; dev_err(&s_rt->slave->dev, - "Chn prep failed for port:%d\n", prep_ch.num); - return -ETIMEDOUT; + "Chn prep failed for port %d: %d\n", prep_ch.num, ret); + return ret; } } -- 2.20.1

3 2

[PATCH v2] soundwire: bus: Make sdw_nwrite() data pointer argument const
by Richard Fitzgerald 20 Jun '21

20 Jun '21

Idiomatically, write functions should take const pointers to the data buffer, as they don't change the data. They are also likely to be called from functions that receive a const data pointer. Internally the pointer is passed to function/structs shared with the read functions, requiring a cast, but this is an implementation detail that should be hidden by the public API. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/soundwire/bus.c | 8 ++++---- include/linux/soundwire/sdw.h | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index a9e0aa72654d..49c66deba106 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -394,13 +394,13 @@ sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) } static int -sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) +sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { struct sdw_msg msg; int ret; ret = sdw_fill_msg(&msg, slave, addr, count, - slave->dev_num, SDW_MSG_FLAG_WRITE, val); + slave->dev_num, SDW_MSG_FLAG_WRITE, (u8 *)val); if (ret < 0) return ret; @@ -535,9 +535,9 @@ EXPORT_SYMBOL(sdw_nread); * @slave: SDW Slave * @addr: Register address * @count: length - * @val: Buffer for values to be read + * @val: Buffer for values to be written */ -int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) +int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { int ret; diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index ced07f8fde87..ee45d2984334 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -1040,7 +1040,7 @@ int sdw_write(struct sdw_slave *slave, u32 addr, u8 value); int sdw_write_no_pm(struct sdw_slave *slave, u32 addr, u8 value); int sdw_read_no_pm(struct sdw_slave *slave, u32 addr); int sdw_nread(struct sdw_slave *slave, u32 addr, size_t count, u8 *val); -int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, u8 *val); +int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val); int sdw_compare_devid(struct sdw_slave *slave, struct sdw_slave_id id); void sdw_extract_slave_id(struct sdw_bus *bus, u64 addr, struct sdw_slave_id *id); -- 2.20.1

2 1

[PATCH] ALSA: hda/realtek: Fix bass speaker DAC mapping for Asus UM431D
by Takashi Iwai 20 Jun '21

20 Jun '21

Asus Zenbook 14 UM431D has two speaker pins and a headphone pin, and the auto-parser ends up assigning the bass to the third DAC 0x06. Although the tone comes out, it's inconvenient because this DAC has no volume control unlike two other DACs. For obtaining the volume control for the bass speaker, this patch enforces the mapping to let both front and bass speaker pins sharing the same DAC. It's not ideal but a little bit of improvement. Since we've already applied the same workaround for another ASUS machine, we just need to hook the chain to the existing quirk. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212547 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 07eabcf22b5f..49f4cac8b05e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7839,6 +7839,8 @@ static const struct hda_fixup alc269_fixups[] = { { 0x20, AC_VERB_SET_PROC_COEF, 0x4e4b }, { } }, + .chained = true, + .chain_id = ALC289_FIXUP_ASUS_GA401, }, [ALC285_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, -- 2.26.2

1 0

[PATCH] ALSA: hda/realtek: Improve fixup for HP Spectre x360 15-df0xxx
by Elia Devito 20 Jun '21

20 Jun '21

On HP Spectre x360 15-df0xxx, after system boot with plugged headset, the headset mic are not detected. Moving pincfg and DAC's config to single fixup function fix this. Signed-off-by: Elia Devito <eliadevito(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 27 ++++++++++++++++++++------- 1 file changed, 20 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 59d0936377eb..07eabcf22b5f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6354,6 +6354,24 @@ static void alc_fixup_no_int_mic(struct hda_codec *codec, } } +static void alc285_fixup_hp_spectre_x360(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t conn[] = { 0x02 }; + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170110 }, /* rear speaker */ + { } + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + /* force front speaker to DAC1 */ + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + break; + } +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -8138,13 +8156,8 @@ static const struct hda_fixup alc269_fixups[] = { .chain_id = ALC269_FIXUP_HP_LINE1_MIC1_LED, }, [ALC285_FIXUP_HP_SPECTRE_X360] = { - .type = HDA_FIXUP_PINS, - .v.pins = (const struct hda_pintbl[]) { - { 0x14, 0x90170110 }, /* enable top speaker */ - {} - }, - .chained = true, - .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1, + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_spectre_x360, }, [ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP] = { .type = HDA_FIXUP_FUNC, -- 2.31.1

2 1

[PATCH] ALSA: bebob: add support for ToneWeal FW66
by Takashi Sakamoto 20 Jun '21

20 Jun '21

A user of FFADO project reported the issue of ToneWeal FW66. As a result, the device is identified as one of applications of BeBoB solution. I note that in the report the device returns contradictory result in plug discovery process for audio subunit. Fortunately ALSA BeBoB driver doesn't perform it thus it's likely to handle the device without issues. I receive no reaction to test request for this patch yet, however it would be worth to add support for it. daniel@gibbonmoon:/sys/bus/firewire/devices/fw1$ grep -r . * Binary file config_rom matches dev:244:1 guid:0x0023270002000000 hardware_version:0x000002 is_local:0 model:0x020002 model_name:FW66 power/runtime_active_time:0 power/runtime_active_kids:0 power/runtime_usage:0 power/runtime_status:unsupported power/async:disabled power/runtime_suspended_time:0 power/runtime_enabled:disabled power/control:auto subsystem/drivers_autoprobe:1 uevent:MAJOR=244 uevent:MINOR=1 uevent:DEVNAME=fw1 units:0x00a02d:0x010001 vendor:0x002327 vendor_name:ToneWeal fw1.0/uevent:MODALIAS=ieee1394:ven00002327mo00020002sp0000A02Dver00010001 fw1.0/power/runtime_active_time:0 fw1.0/power/runtime_active_kids:0 fw1.0/power/runtime_usage:0 fw1.0/power/runtime_status:unsupported fw1.0/power/async:disabled fw1.0/power/runtime_suspended_time:0 fw1.0/power/runtime_enabled:disabled fw1.0/power/control:auto fw1.0/model:0x020002 fw1.0/rom_index:15 fw1.0/specifier_id:0x00a02d fw1.0/model_name:FW66 fw1.0/version:0x010001 fw1.0/modalias:ieee1394:ven00002327mo00020002sp0000A02Dver00010001 Cc: Daniel Jozsef <daniel.jozsef(a)gmail.com> Reference: https://lore.kernel.org/alsa-devel/20200119164335.GA11974@workstation/ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 1 + sound/firewire/bebob/bebob.c | 3 +++ 2 files changed, 4 insertions(+) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 62658e2b111a..fd109bea4c53 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -127,6 +127,7 @@ config SND_BEBOB * M-Audio Ozonic/NRV10/ProfireLightBridge * M-Audio FireWire 1814/ProjectMix IO * Digidesign Mbox 2 Pro + * ToneWeal FW66 To compile this driver as a module, choose M here: the module will be called snd-bebob. diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 7ca452efb133..42980da45fbf 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -56,6 +56,7 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS); #define VEN_FOCUSRITE 0x0000130e #define VEN_MAUDIO 0x00000d6c #define VEN_DIGIDESIGN 0x00a07e +#define OUI_SHOUYO 0x002327 #define MODEL_FOCUSRITE_SAFFIRE_BOTH 0x00000000 #define MODEL_MAUDIO_AUDIOPHILE_BOTH 0x00010060 @@ -469,6 +470,8 @@ static const struct ieee1394_device_id bebob_id_table[] = { &maudio_special_spec), /* Digidesign Mbox 2 Pro */ SND_BEBOB_DEV_ENTRY(VEN_DIGIDESIGN, 0x0000a9, &spec_normal), + // Toneweal FW66. + SND_BEBOB_DEV_ENTRY(OUI_SHOUYO, 0x020002, &spec_normal), /* IDs are unknown but able to be supported */ /* Apogee, Mini-ME Firewire */ /* Apogee, Mini-DAC Firewire */ -- 2.30.2

2 1

[PATCH 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 19 Jun '21

19 Jun '21

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 408 ++++++++++++++++++++++++++++++++++ 4 files changed, 424 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index b4094f07214e..1b87225c39b0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..78d512d93974 --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,408 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * ALSA SoC DMIC Audio Layer + * + * Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + * + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMOLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMOLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct platform_device *pdev; + struct clk *dmic_clk; + struct clk *apb_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static void sun50i_snd_rxctrl_enable(struct snd_pcm_substream *substream, + struct sun50i_dmic_dev *host, bool enable) +{ + if (enable) { + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + } else { + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + } +} + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(rtd->cpu_dai); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_write(host->regmap, SUN50I_DMIC_INT_STA, + SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING | + SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct platform_device *pdev = host->pdev; + struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(&pdev->dev, "Invalid format!\n"); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMOLE_RATE_MASK, + SUN50I_DMIC_SR_SAMOLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + if (i == ARRAY_SIZE(dmic_rate_s)) { + dev_err(&pdev->dev, "Invalid rate!\n"); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sun50i_snd_rxctrl_enable(substream, host, true); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sun50i_snd_rxctrl_enable(substream, host, false); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (clk_set_rate(host->dmic_clk, freq)) { + dev_err(dai->dev, "Freq : %u not support\n", freq); + return -EINVAL; + } + + return 0; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, + .set_sysclk = sun50i_dmic_set_sysclk, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->apb_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->apb_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + host->pdev = pdev; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->apb_clk = devm_clk_get(&pdev->dev, "apb"); + if (IS_ERR(host->apb_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->apb_clk), + "failed to get apb clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "dmic"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst) && PTR_ERR(host->rst) == -EPROBE_DEFER) { + ret = -EPROBE_DEFER; + dev_err(&pdev->dev, "Failed to get reset: %d\n", ret); + return ret; + } + if (!IS_ERR(host->rst)) + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.29.0

5 10

[PATCH] ALSA: hda/realtek: Add another ALC236 variant support
by Takashi Iwai 18 Jun '21

18 Jun '21

The codec chip 10ec:0230 is another variant of ALC236, combined with a card reader. Apply the equivalent setup as 10ec:0236. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1184869 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bbbe37f3e724..fac2f15d9e26 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -385,6 +385,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) alc_update_coef_idx(codec, 0x67, 0xf000, 0x3000); fallthrough; case 0x10ec0215: + case 0x10ec0230: case 0x10ec0233: case 0x10ec0235: case 0x10ec0236: @@ -3127,6 +3128,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x48, 0x0); @@ -3154,6 +3156,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x48, 0xd011); @@ -4689,6 +4692,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, coef0256); @@ -4803,6 +4807,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, alc_process_coef_fw(codec, coef0255); snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x45, 0xc489); @@ -4952,6 +4957,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x1b, 0x0e4b); @@ -5050,6 +5056,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, coef0256); @@ -5163,6 +5170,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, coef0256); @@ -5263,6 +5271,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) val = alc_read_coef_idx(codec, 0x46); is_ctia = (val & 0x0070) == 0x0070; break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x1b, 0x0e4b); @@ -5556,6 +5565,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, alc255fw); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, alc256fw); @@ -6156,6 +6166,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4a, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x4a, 0x8000, 0 << 15); break; + case 0x10ec0230: case 0x10ec0235: case 0x10ec0236: case 0x10ec0255: @@ -9130,6 +9141,7 @@ static int patch_alc269(struct hda_codec *codec) spec->shutup = alc256_shutup; spec->init_hook = alc256_init; break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: spec->codec_variant = ALC269_TYPE_ALC256; @@ -10421,6 +10433,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269), HDA_CODEC_ENTRY(0x10ec0222, "ALC222", patch_alc269), HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0230, "ALC236", patch_alc269), HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269), HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269), HDA_CODEC_ENTRY(0x10ec0234, "ALC234", patch_alc269), -- 2.26.2

1 0

[PATCH v3 1/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8
by Andy Chi 18 Jun '21

18 Jun '21

>From fdeab6f32dbdbfd83c9af6efc5a51f91021b6578 Mon Sep 17 00:00:00 2001 From: Andy Chi <andy.chi(a)canonical.com> Date: Fri, 18 Jun 2021 16:17:48 +0800 Subject: [PATCH v3 1/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8 The HP ProBook 450 G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- v2 -> v3: no changes. v1 -> v2: fix typo from HP ProBook 440 to HP ProBook 450 G8 (https://lkml.org/lkml/2021/6/18/325) sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ab5113cccffa..70bc5b11acfd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8336,6 +8336,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f2, "HP ProBook 640 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), -- 2.25.1

2 1

[PATCH 1/1] ASoC: Intel: bdw-rt5677: remove unnecessary oom message
by Zhen Lei 18 Jun '21

18 Jun '21

Fixes scripts/checkpatch.pl warning: WARNING: Possible unnecessary 'out of memory' message Remove it can help us save a bit of memory. Signed-off-by: Zhen Lei <thunder.leizhen(a)huawei.com> --- sound/soc/intel/boards/bdw-rt5677.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 021bc59aac80..e01b7a90ca6c 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -423,10 +423,8 @@ static int bdw_rt5677_probe(struct platform_device *pdev) /* Allocate driver private struct */ bdw_rt5677 = devm_kzalloc(&pdev->dev, sizeof(struct bdw_rt5677_priv), GFP_KERNEL); - if (!bdw_rt5677) { - dev_err(&pdev->dev, "Can't allocate bdw_rt5677\n"); + if (!bdw_rt5677) return -ENOMEM; - } /* override plaform name, if required */ mach = pdev->dev.platform_data; -- 2.25.1

3 2

[PATCH] ASoC: fsi: fix spelling mistake
by Flavio Suligoi 18 Jun '21

18 Jun '21

Fix "thse" --> "these" in struct fsi_stream declaration. Signed-off-by: Flavio Suligoi <f.suligoi(a)asem.it> --- sound/soc/sh/fsi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3c934f87c242..cdf3b7f69ba7 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -219,7 +219,7 @@ struct fsi_stream { u32 bus_option; /* - * thse are initialized by fsi_handler_init() + * these are initialized by fsi_handler_init() */ struct fsi_stream_handler *handler; struct fsi_priv *priv; -- 2.25.1

2 1

[PATCH -next] ASoC: rk817: Constify static struct snd_soc_dai_ops
by Wei Yongjun 18 Jun '21

18 Jun '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/rk817_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 17e672b85ee5..9a896e4326c3 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -382,7 +382,7 @@ static int rk817_digital_mute(struct snd_soc_dai *dai, int mute, int stream) SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops rk817_dai_ops = { +static const struct snd_soc_dai_ops rk817_dai_ops = { .hw_params = rk817_hw_params, .set_fmt = rk817_set_dai_fmt, .set_sysclk = rk817_set_dai_sysclk,

3 2

[PATCH] soundwire: stream: Use polling for DP Prepare completion
by Richard Fitzgerald 18 Jun '21

18 Jun '21

sdw_prep_deprep_slave_ports() cannot use the port interrupt to signal CP_SM completion because it is called while holding the bus lock, which blocks the alert handler from running. Change to polling the PREPARESTATUS register and remove the infrastructure that was implemented for the previous interrupt waiting. A new configuration field 'ch_prep_poll_us' is added to struct sdw_dpn_prop so that the peripheral driver may select a polling interval. If this is left at zero a default interval of 500 usec is used. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/soundwire/bus.c | 2 -- drivers/soundwire/slave.c | 4 ---- drivers/soundwire/stream.c | 24 ++++++++++++++---------- include/linux/soundwire/sdw.h | 8 ++++++-- 4 files changed, 20 insertions(+), 18 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index adcbf3969110..606fc26d407f 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1351,7 +1351,6 @@ static int sdw_handle_dp0_interrupt(struct sdw_slave *slave, u8 *slave_status) */ if (status & SDW_DP0_INT_PORT_READY) { - complete(&slave->port_ready[0]); clear |= SDW_DP0_INT_PORT_READY; } @@ -1429,7 +1428,6 @@ static int sdw_handle_port_interrupt(struct sdw_slave *slave, * for ports implementing CP_SM. */ if (status & SDW_DPN_INT_PORT_READY) { - complete(&slave->port_ready[port]); clear |= SDW_DPN_INT_PORT_READY; } diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 669d7573320b..55ca884ea951 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -26,7 +26,6 @@ int sdw_slave_add(struct sdw_bus *bus, { struct sdw_slave *slave; int ret; - int i; slave = kzalloc(sizeof(*slave), GFP_KERNEL); if (!slave) @@ -62,9 +61,6 @@ int sdw_slave_add(struct sdw_bus *bus, slave->probed = false; slave->first_interrupt_done = false; - for (i = 0; i < SDW_MAX_PORTS; i++) - init_completion(&slave->port_ready[i]); - mutex_lock(&bus->bus_lock); list_add_tail(&slave->node, &bus->slaves); mutex_unlock(&bus->bus_lock); diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 1eaedaaba094..bd6b3b64de90 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -8,11 +8,13 @@ #include <linux/delay.h> #include <linux/device.h> #include <linux/init.h> +#include <linux/iopoll.h> #include <linux/module.h> #include <linux/mod_devicetable.h> #include <linux/slab.h> #include <linux/soundwire/sdw_registers.h> #include <linux/soundwire/sdw.h> +#include <linux/time.h> #include <sound/soc.h> #include "bus.h" @@ -419,11 +421,10 @@ static int sdw_prep_deprep_slave_ports(struct sdw_bus *bus, struct sdw_port_runtime *p_rt, bool prep) { - struct completion *port_ready; struct sdw_dpn_prop *dpn_prop; struct sdw_prepare_ch prep_ch; - unsigned int time_left; bool intr = false; + unsigned long prep_poll_us, prep_timeout_us; int ret = 0, val; u32 addr; @@ -478,16 +479,19 @@ static int sdw_prep_deprep_slave_ports(struct sdw_bus *bus, } /* Wait for completion on port ready */ - port_ready = &s_rt->slave->port_ready[prep_ch.num]; - time_left = wait_for_completion_timeout(port_ready, - msecs_to_jiffies(dpn_prop->ch_prep_timeout)); + prep_timeout_us = dpn_prop->ch_prep_timeout * USEC_PER_MSEC; + if (dpn_prop->ch_prep_poll_us) + prep_poll_us = dpn_prop->ch_prep_poll_us; + else + prep_poll_us = SDW_DEFAULT_DP_PREP_POLL_US; - val = sdw_read(s_rt->slave, SDW_DPN_PREPARESTATUS(p_rt->num)); - val &= p_rt->ch_mask; - if (!time_left || val) { + ret = read_poll_timeout(sdw_read, val, ((val & p_rt->ch_mask) == 0), + prep_poll_us, prep_timeout_us, false, + s_rt->slave, SDW_DPN_PREPARESTATUS(p_rt->num)); + if (ret < 0) { dev_err(&s_rt->slave->dev, - "Chn prep failed for port:%d\n", prep_ch.num); - return -ETIMEDOUT; + "Chn prep failed for port %d: %d\n", prep_ch.num, ret); + return ret; } } diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index ddbeb00799e4..4e5290b083bf 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -67,6 +67,9 @@ enum { #define SDW_BLOCK_PACKG_PER_PORT BIT(0) #define SDW_BLOCK_PACKG_PER_CH BIT(1) +/* Default interval to poll for DP prepare completion */ +#define SDW_DEFAULT_DP_PREP_POLL_US 500 + /** * enum sdw_slave_status - Slave status * @SDW_SLAVE_UNATTACHED: Slave is not attached with the bus. @@ -295,6 +298,8 @@ struct sdw_dpn_audio_mode { * @simple_ch_prep_sm: If the port supports simplified channel prepare state * machine * @ch_prep_timeout: Port-specific timeout value, in milliseconds + * @ch_prep_poll_us: Interval to poll for CP_SM prepare completion. Zero means + * poll at SDW_DEFAULT_DP_PREP_POLL_US intervals. * @imp_def_interrupts: If set, each bit corresponds to support for * implementation-defined interrupts * @max_ch: Maximum channels supported @@ -321,6 +326,7 @@ struct sdw_dpn_prop { u32 max_grouping; bool simple_ch_prep_sm; u32 ch_prep_timeout; + u32 ch_prep_poll_us; u32 imp_def_interrupts; u32 max_ch; u32 min_ch; @@ -641,7 +647,6 @@ struct sdw_slave_ops { * @prop: Slave properties * @debugfs: Slave debugfs * @node: node for bus list - * @port_ready: Port ready completion flag for each Slave port * @m_port_map: static Master port map for each Slave port * @dev_num: Current Device Number, values can be 0 or dev_num_sticky * @dev_num_sticky: one-time static Device Number assigned by Bus @@ -673,7 +678,6 @@ struct sdw_slave { struct dentry *debugfs; #endif struct list_head node; - struct completion port_ready[SDW_MAX_PORTS]; unsigned int m_port_map[SDW_MAX_PORTS]; u16 dev_num; u16 dev_num_sticky; -- 2.20.1

2 3

[PATCH v3 2/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 445 G8
by Andy Chi 18 Jun '21

18 Jun '21

>From f6d2556f8f6bcdd2261b48604b8086fb67ae9a03 Mon Sep 17 00:00:00 2001 From: Andy Chi <andy.chi(a)canonical.com> Date: Fri, 18 Jun 2021 16:23:06 +0800 Subject: [PATCH v3 2/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 445 G8 The HP ProBook 445 G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- v2 -> v3: no changes. v1 -> v2: no changes. sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 70bc5b11acfd..a1e62b2de167 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8344,6 +8344,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero 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, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), -- 2.25.1

1 0

[PATCH v2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8
by Andy Chi 18 Jun '21

18 Jun '21

>From fdeab6f32dbdbfd83c9af6efc5a51f91021b6578 Mon Sep 17 00:00:00 2001 From: Andy Chi <andy.chi(a)canonical.com> Date: Fri, 18 Jun 2021 16:17:48 +0800 Subject: [PATCH v2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8 The HP ProBook 450 G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ab5113cccffa..70bc5b11acfd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8336,6 +8336,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f2, "HP ProBook 640 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), -- 2.25.1

1 0

[PATCH 1/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8
by Andy Chi 18 Jun '21

18 Jun '21

>From fdeab6f32dbdbfd83c9af6efc5a51f91021b6578 Mon Sep 17 00:00:00 2001 From: Andy Chi <andy.chi(a)canonical.com> Date: Fri, 18 Jun 2021 16:17:48 +0800 Subject: [PATCH 1/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8 The HP ProBook 450 G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ab5113cccffa..70bc5b11acfd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8336,6 +8336,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f2, "HP ProBook 640 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), -- 2.25.1

1 1

[bug report] ASoC: codecs: wcd938x: add capture dapm widgets
by Dan Carpenter 18 Jun '21

18 Jun '21

Hello Srinivas Kandagatla, The patch d5add08fcbce: "ASoC: codecs: wcd938x: add capture dapm widgets" from Jun 9, 2021, leads to the following static checker warning: sound/soc/codecs/wcd938x.c:2189 wcd938x_tx_channel_config() error: uninitialized symbol 'reg'. sound/soc/codecs/wcd938x.c 2165 static void wcd938x_tx_channel_config(struct snd_soc_component *component, 2166 int channel, int mode) 2167 { 2168 int reg, mask; 2169 2170 switch (channel) { 2171 case 0: 2172 reg = WCD938X_ANA_TX_CH2; 2173 mask = WCD938X_HPF1_INIT_MASK; 2174 break; 2175 case 1: 2176 reg = WCD938X_ANA_TX_CH2; 2177 mask = WCD938X_HPF2_INIT_MASK; 2178 break; 2179 case 2: 2180 reg = WCD938X_ANA_TX_CH4; 2181 mask = WCD938X_HPF3_INIT_MASK; 2182 break; 2183 case 3: 2184 reg = WCD938X_ANA_TX_CH4; 2185 mask = WCD938X_HPF4_INIT_MASK; 2186 break; We could add a default case? default: return; 2187 } 2188 2189 snd_soc_component_write_field(component, reg, mask, mode); 2190 } regards, dan carpenter

1 0

[PATCH] ALSA: firewire-motu: fix rx packet format at higher rate for MOTU 828 mk3 Hybrid
by Takashi Sakamoto 18 Jun '21

18 Jun '21

I assumed that the combination of packet formats for MOTU 828 mk3 Hybrid is the same as MOTU 828 mk3 FireWire. However at higher sampling rate, it is different. MOTU 828 mk3 Hybrid has additional 4 dummy data chunks for rx packet. This commit fixes the issue to which I address at a commit f2ac3b839540 ("ALSA: firewire-motu: sequence replay for source packet header"). Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-protocol-v3.c | 15 ++++++++++++--- sound/firewire/motu/motu.c | 4 ++-- sound/firewire/motu/motu.h | 3 ++- 3 files changed, 16 insertions(+), 6 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index 4e6b0e449ee4..77e61e89770b 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -185,7 +185,7 @@ int snd_motu_protocol_v3_get_clock_source(struct snd_motu *motu, return err; data = be32_to_cpu(reg) & V3_CLOCK_SOURCE_MASK; - if (motu->spec == &snd_motu_spec_828mk3) + if (motu->spec == &snd_motu_spec_828mk3_fw || motu->spec == &snd_motu_spec_828mk3_hybrid) return detect_clock_source_828mk3(motu, data, src); else return v3_detect_clock_source(motu, data, src); @@ -284,14 +284,14 @@ int snd_motu_protocol_v3_cache_packet_formats(struct snd_motu *motu) motu->spec->rx_fixed_pcm_chunks, sizeof(motu->rx_packet_formats.pcm_chunks)); - if (motu->spec == &snd_motu_spec_828mk3) + if (motu->spec == &snd_motu_spec_828mk3_fw || motu->spec == &snd_motu_spec_828mk3_hybrid) return detect_packet_formats_828mk3(motu, data); else return 0; } -const struct snd_motu_spec snd_motu_spec_828mk3 = { +const struct snd_motu_spec snd_motu_spec_828mk3_fw = { .name = "828mk3", .protocol_version = SND_MOTU_PROTOCOL_V3, .flags = SND_MOTU_SPEC_RX_MIDI_3RD_Q | @@ -300,6 +300,15 @@ const struct snd_motu_spec snd_motu_spec_828mk3 = { .rx_fixed_pcm_chunks = {14, 14, 10}, }; +const struct snd_motu_spec snd_motu_spec_828mk3_hybrid = { + .name = "828mk3", + .protocol_version = SND_MOTU_PROTOCOL_V3, + .flags = SND_MOTU_SPEC_RX_MIDI_3RD_Q | + SND_MOTU_SPEC_TX_MIDI_3RD_Q, + .tx_fixed_pcm_chunks = {18, 18, 14}, + .rx_fixed_pcm_chunks = {14, 14, 14}, // Additional 4 dummy chunks at higher rate. +}; + const struct snd_motu_spec snd_motu_spec_ultralite_mk3 = { .name = "UltraLiteMk3", .protocol_version = SND_MOTU_PROTOCOL_V3, diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 0cae670d711c..543136578c70 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -156,10 +156,10 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x000009, &snd_motu_spec_traveler), SND_MOTU_DEV_ENTRY(0x00000d, &snd_motu_spec_ultralite), SND_MOTU_DEV_ENTRY(0x00000f, &snd_motu_spec_8pre), - SND_MOTU_DEV_ENTRY(0x000015, &snd_motu_spec_828mk3), // FireWire only. + SND_MOTU_DEV_ENTRY(0x000015, &snd_motu_spec_828mk3_fw), // FireWire only. SND_MOTU_DEV_ENTRY(0x000019, &snd_motu_spec_ultralite_mk3), // FireWire only. SND_MOTU_DEV_ENTRY(0x000030, &snd_motu_spec_ultralite_mk3), // Hybrid. - SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3), // Hybrid. + SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3_hybrid), // Hybrid. SND_MOTU_DEV_ENTRY(0x000033, &snd_motu_spec_audio_express), SND_MOTU_DEV_ENTRY(0x000045, &snd_motu_spec_4pre), { } diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index a3deabdf9e34..73f36d1be515 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -130,7 +130,8 @@ extern const struct snd_motu_spec snd_motu_spec_traveler; extern const struct snd_motu_spec snd_motu_spec_ultralite; extern const struct snd_motu_spec snd_motu_spec_8pre; -extern const struct snd_motu_spec snd_motu_spec_828mk3; +extern const struct snd_motu_spec snd_motu_spec_828mk3_fw; +extern const struct snd_motu_spec snd_motu_spec_828mk3_hybrid; extern const struct snd_motu_spec snd_motu_spec_ultralite_mk3; extern const struct snd_motu_spec snd_motu_spec_audio_express; extern const struct snd_motu_spec snd_motu_spec_4pre; -- 2.30.2

2 1

[PATCH] ALSA: bebob: fix rx packet format for Yamaha GO44/GO46, Terratec Phase 24/x24
by Takashi Sakamoto 18 Jun '21

18 Jun '21

Below devices reports zero as the number of channels for external output plug with MIDI type: * Yamaha GO44/GO46 * Terratec Phase 24/X24 As a result, rx packet format is invalid and they generate silent sound. This is a regression added in v5.13. This commit fixes the bug, addressed at a commit 1bd1b3be8655 ("ALSA: bebob: perform sequence replay for media clock recovery"). Cc: <stable(a)vger.kernel.org> Fixes: 5c6ea94f2b7c ("ALSA: bebob: detect the number of available MIDI ports") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_stream.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index e3e23e42add3..8629b14ded76 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -856,6 +856,11 @@ static int detect_midi_ports(struct snd_bebob *bebob, err = avc_bridgeco_get_plug_ch_count(bebob->unit, addr, &ch_count); if (err < 0) break; + // Yamaha GO44, GO46, Terratec Phase 24, Phase x24 reports 0 for the number of + // channels in external output plug 3 (MIDI type) even if it has a pair of physical + // MIDI jacks. As a workaround, assume it as one. + if (ch_count == 0) + ch_count = 1; *midi_ports += ch_count; } @@ -934,12 +939,12 @@ int snd_bebob_stream_discover(struct snd_bebob *bebob) if (err < 0) goto end; - err = detect_midi_ports(bebob, bebob->rx_stream_formations, addr, AVC_BRIDGECO_PLUG_DIR_IN, + err = detect_midi_ports(bebob, bebob->tx_stream_formations, addr, AVC_BRIDGECO_PLUG_DIR_IN, plugs[2], &bebob->midi_input_ports); if (err < 0) goto end; - err = detect_midi_ports(bebob, bebob->tx_stream_formations, addr, AVC_BRIDGECO_PLUG_DIR_OUT, + err = detect_midi_ports(bebob, bebob->rx_stream_formations, addr, AVC_BRIDGECO_PLUG_DIR_OUT, plugs[3], &bebob->midi_output_ports); if (err < 0) goto end; -- 2.30.2

2 1

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP EliteBook x360 830 G8
by Jeremy Szu 18 Jun '21

18 Jun '21

The HP EliteBook x360 830 G8 using ALC285 codec which using 0x04 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dcd9bceace68..ae5963345093 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8330,6 +8330,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), 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, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero 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, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), -- 2.31.1

2 1

[bug report] ASoC: codecs: wcd938x: add capture dapm widgets
by Dan Carpenter 18 Jun '21

18 Jun '21

Hello Srinivas Kandagatla, The patch d5add08fcbce: "ASoC: codecs: wcd938x: add capture dapm widgets" from Jun 9, 2021, leads to the following static checker warning: sound/soc/codecs/wcd938x.c:2092 wcd938x_tx_swr_ctrl() error: uninitialized symbol 'rate'. sound/soc/codecs/wcd938x.c 2064 switch (event) { 2065 case SND_SOC_DAPM_PRE_PMU: 2066 if (strnstr(w->name, "ADC", sizeof("ADC"))) { Use strncmp() here instead of strnstr(). The sizeof() will include the NUL terminator so it's size 4 and you wanted 3. The off by one means that strnstr() will do two memcmp()s. The equivalent of: if (memcmp(w->name, "ADC", 3) == 0) return w->name; if (memcmp(w->name + 1, "ADC", 3) == 0) return w->name + 1; return NULL; 2067 int i = 0, mode = 0; 2068 2069 if (test_bit(WCD_ADC1, &wcd938x->status_mask)) 2070 mode |= tx_mode_bit[wcd938x->tx_mode[WCD_ADC1]]; 2071 if (test_bit(WCD_ADC2, &wcd938x->status_mask)) 2072 mode |= tx_mode_bit[wcd938x->tx_mode[WCD_ADC2]]; 2073 if (test_bit(WCD_ADC3, &wcd938x->status_mask)) 2074 mode |= tx_mode_bit[wcd938x->tx_mode[WCD_ADC3]]; 2075 if (test_bit(WCD_ADC4, &wcd938x->status_mask)) 2076 mode |= tx_mode_bit[wcd938x->tx_mode[WCD_ADC4]]; 2077 2078 if (mode != 0) { 2079 for (i = 0; i < ADC_MODE_ULP2; i++) { 2080 if (mode & (1 << i)) { 2081 i++; 2082 break; 2083 } 2084 } 2085 } 2086 rate = wcd938x_get_clk_rate(i); 2087 wcd938x_set_swr_clk_rate(component, rate, bank); 2088 } 2089 2090 if (strnstr(w->name, "ADC", sizeof("ADC"))) 2091 /* Copy clk settings to active bank */ 2092 wcd938x_set_swr_clk_rate(component, rate, !bank); This a false positive, but you could silence it by combining it with the previous if block. 2093 break; 2094 case SND_SOC_DAPM_POST_PMD: 2095 if (strnstr(w->name, "ADC", sizeof("ADC"))) { 2096 rate = wcd938x_get_clk_rate(ADC_MODE_INVALID); 2097 wcd938x_set_swr_clk_rate(component, rate, !bank); 2098 wcd938x_set_swr_clk_rate(component, rate, bank); 2099 } 2100 break; 2101 } regards, dan carpenter

1 0

[PATCH v3 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 18 Jun '21

18 Jun '21

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- v1->v2: 1.Fix some compilation errors. 2.Modify some code styles. --- v2->v3: None. --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 404 ++++++++++++++++++++++++++++++++++ 4 files changed, 420 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index b4094f07214e..1b87225c39b0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..d0b576528552 --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,404 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMOLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMOLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMOLE_RATE_MASK, + SUN50I_DMIC_SR_SAMOLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->bus_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.29.0

1 0

[PATCH v2 2/2] ASoC: sun50i-dmic: dt-bindings: add DT bindings for DMIC controller
by Ban Tao 17 Jun '21

17 Jun '21

DT binding documentation for this new ASoC driver. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- .../sound/allwinner,sun50i-h6-dmic.yaml | 68 +++++++++++++++++++ 1 file changed, 68 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml new file mode 100644 index 000000000000..d7af66775e86 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml @@ -0,0 +1,68 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner H6 DMIC Device Tree Bindings + +maintainers: + - Ban Tao <fengzheng923(a)gmail.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: allwinner,sun50i-h6-dmic + + reg: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + + dmas: + items: + - description: RX DMA Channel + + dma-names: + items: + - const: rx + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - clocks + - clock-names + - dmas + - dma-names + - resets + +additionalProperties: false + +examples: + - | + dmic: dmic@5095000 { + #sound-dai-cells = <1>; + compatible = "allwinner,sun50i-h6-dmic"; + reg = <0x05095000 0x400>; + clocks = <&ccu 101>, <&ccu 100>; + clock-names = "bus", "mod"; + dmas = <&dma 7>; + dma-names = "rx"; + resets = <&ccu 42>; + }; + +... -- 2.29.0

2 1

[PATCH 1/1] ALSA: aoa: remove unnecessary oom message
by Zhen Lei 17 Jun '21

17 Jun '21

Fixes scripts/checkpatch.pl warning: WARNING: Possible unnecessary 'out of memory' message Remove it can help us save a bit of memory. Signed-off-by: Zhen Lei <thunder.leizhen(a)huawei.com> --- sound/aoa/soundbus/i2sbus/pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 1c8e8131a716..a9e502a6cdeb 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -918,10 +918,8 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, } cii = kzalloc(sizeof(struct codec_info_item), GFP_KERNEL); - if (!cii) { - printk(KERN_DEBUG "i2sbus: failed to allocate cii\n"); + if (!cii) return -ENOMEM; - } /* use the private data to point to the codec info */ cii->sdev = soundbus_dev_get(dev); -- 2.25.1

2 1

[PATCH 1/1] ASoC: soc-core: Fix the error return code in snd_soc_of_parse_audio_routing()
by Zhen Lei 17 Jun '21

17 Jun '21

When devm_kcalloc() fails, the error code -ENOMEM should be returned instead of -EINVAL. Signed-off-by: Zhen Lei <thunder.leizhen(a)huawei.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9e7f4ceffb65..dbf7de925f8d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2957,7 +2957,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, if (!routes) { dev_err(card->dev, "ASoC: Could not allocate DAPM route table\n"); - return -EINVAL; + return -ENOMEM; } for (i = 0; i < num_routes; i++) { -- 2.25.1

2 1

[PATCH -next] ASoC: spear: spdif_out: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 17 Jun '21

17 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/spear/spdif_out.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 38f9fff5be6b..549295a6ed50 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -287,8 +287,7 @@ static int spdif_out_probe(struct platform_device *pdev) if (!host) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - host->io_base = devm_ioremap_resource(&pdev->dev, res); + host->io_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(host->io_base)) return PTR_ERR(host->io_base); -- 2.25.1

2 1

[PATCH -next 0/3] ASoC: stm32: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 17 Jun '21

17 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Yang Yingliang (3): ASoC: stm32: i2s: Use devm_platform_get_and_ioremap_resource() ASoC: stm32: sai: Use devm_platform_get_and_ioremap_resource() ASoC: stm32: spdifrx: Use devm_platform_get_and_ioremap_resource() sound/soc/stm/stm32_i2s.c | 3 +-- sound/soc/stm/stm32_sai_sub.c | 3 +-- sound/soc/stm/stm32_spdifrx.c | 3 +-- 3 files changed, 3 insertions(+), 6 deletions(-) -- 2.25.1

2 4

[PATCH -next] ASoC: sti: sti_uniperif: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 17 Jun '21

17 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sti/sti_uniperif.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index e3561f00ed40..34668fe3909d 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -410,16 +410,8 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node, *dai = sti_uniperiph_dai_template; dai->name = dev_data->dai_names; - /* Get resources */ - uni->mem_region = platform_get_resource(priv->pdev, IORESOURCE_MEM, 0); - - if (!uni->mem_region) { - dev_err(dev, "Failed to get memory resource\n"); - return -ENODEV; - } - - uni->base = devm_ioremap_resource(dev, uni->mem_region); - + /* Get resources and base address */ + uni->base = devm_platform_get_and_ioremap_resource(priv->pdev, 0, &uni->mem_region); if (IS_ERR(uni->base)) return PTR_ERR(uni->base); -- 2.25.1

2 1

[PATCH -next] ASoC: sprd: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 17 Jun '21

17 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sprd/sprd-mcdt.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index 34b2ce733b54..f6a55fa60c1b 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -949,8 +949,7 @@ static int sprd_mcdt_probe(struct platform_device *pdev) if (!mcdt) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - mcdt->base = devm_ioremap_resource(&pdev->dev, res); + mcdt->base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(mcdt->base)) return PTR_ERR(mcdt->base); -- 2.25.1

2 1

[PATCH] ALSA: seq: oss: Fix error check at system port creation
by Takashi Iwai 17 Jun '21

17 Jun '21

The system port creation in ALSA OSS sequencer was wrongly checked against to the port number that can be never negative. The error code should be checked rather against the ioctl call. This patch corrects the error check. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/oss/seq_oss_init.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index a53d81a86af2..0ee4a5081fd6 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -94,11 +94,10 @@ snd_seq_oss_create_client(void) port_callback.event_input = receive_announce; port->kernel = &port_callback; - call_ctl(SNDRV_SEQ_IOCTL_CREATE_PORT, port); - system_port = port->addr.port; - if (system_port >= 0) { + if (call_ctl(SNDRV_SEQ_IOCTL_CREATE_PORT, port) >= 0) { struct snd_seq_port_subscribe subs; + system_port = port->addr.port; memset(&subs, 0, sizeof(subs)); subs.sender.client = SNDRV_SEQ_CLIENT_SYSTEM; subs.sender.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE; -- 2.26.2

1 0

[bug report] ALSA: seq: Fix assignment in if condition
by Dan Carpenter 17 Jun '21

17 Jun '21

Hello Takashi Iwai, The patch f9a6bb841f73: "ALSA: seq: Fix assignment in if condition" from Jun 8, 2021, leads to the following static checker warning: sound/core/seq/oss/seq_oss_init.c:99 snd_seq_oss_create_client() warn: 'port->addr.port' is unsigned sound/core/seq/oss/seq_oss_init.c 82 83 /* create annoucement receiver port */ 84 memset(port, 0, sizeof(*port)); 85 strcpy(port->name, "Receiver"); 86 port->addr.client = system_client; 87 port->capability = SNDRV_SEQ_PORT_CAP_WRITE; /* receive only */ 88 port->type = 0; 89 90 memset(&port_callback, 0, sizeof(port_callback)); 91 /* don't set port_callback.owner here. otherwise the module counter 92 * is incremented and we can no longer release the module.. 93 */ 94 port_callback.event_input = receive_announce; 95 port->kernel = &port_callback; 96 97 call_ctl(SNDRV_SEQ_IOCTL_CREATE_PORT, port); 98 system_port = port->addr.port; 99 if (system_port >= 0) { ^^^^^^^^^^^^^^^^ This was from the old code. It's not clear what is going on. I think the condition can be deleted. 100 struct snd_seq_port_subscribe subs; 101 102 memset(&subs, 0, sizeof(subs)); 103 subs.sender.client = SNDRV_SEQ_CLIENT_SYSTEM; 104 subs.sender.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE; 105 subs.dest.client = system_client; 106 subs.dest.port = system_port; 107 call_ctl(SNDRV_SEQ_IOCTL_SUBSCRIBE_PORT, &subs); 108 } 109 rc = 0; 110 111 /* look up midi devices */ 112 schedule_work(&async_lookup_work); 113 114 __error: 115 kfree(port); 116 return rc; 117 } regards, dan carpenter

2 1

[PATCH v2 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 17 Jun '21

17 Jun '21

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- v1->v2: 1.fix some compilation errors. 2.Modify some code styles. --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 405 ++++++++++++++++++++++++++++++++++ 4 files changed, 421 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index b4094f07214e..1b87225c39b0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..d495ee0f6a2d --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,405 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMOLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMOLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMOLE_RATE_MASK, + SUN50I_DMIC_SR_SAMOLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->bus_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.29.0

1 0

[PATCH v2 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 17 Jun '21

17 Jun '21

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- v1->v2: 1.fix some compilation errors. 2.Modify some code styles. --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 405 ++++++++++++++++++++++++++++++++++ 4 files changed, 421 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index b4094f07214e..1b87225c39b0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..d495ee0f6a2d --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,405 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMOLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMOLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMOLE_RATE_MASK, + SUN50I_DMIC_SR_SAMOLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->bus_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.29.0

1 0

a52 plugin crashing pulseaudio
by GitHub issues - edited 17 Jun '21

17 Jun '21

alsa-project/alsa-plugins issue #28 was edited from quequotion: Ever since f11e7a8994111f971bc881d4069f768472452f4f, the recent commit that brings the a52 plugin up to speed with ffmpeg's requirement for s32le format sampling, sending any audio to plug:a52 from any pulseaudio client crashes pulseaudio. I [posted this log](https://gist.github.com/quequotion/78972bac339f5249cf6b383519cdcedf) earlier in #23, in which audio from mpv crashes pulseaudio. I have tried other clients since and they all crash the same way. This is with a default pulseaudio configuration, as shipped in Archlinux, and the following `/etc/asound.conf`: ``` # Alsa Resampler defaults.pcm.rate_converter "speexrate_best" # Use PulseAudio by default pcm.!default { type pulse fallback "sysdefault" hint { show on description "Default ALSA Output (currently PulseAudio Sound Server)" } } ctl.!default { type pulse fallback "sysdefault" } defaults.namehint.showall on defaults.namehint.extended on ``` To enable the use of plug:a52 in pulseaudio without configuring it manually, I select the "Digital Surround 5.1 (IEC958/AC3) Output" profile in `pavucontrol`. Pulseaudio, as far as I know, is agnostic of how ALSA plugins work internally. This leads me to think there is something not getting done right in the new patchset, even if it is tolerable in an ALSA-only setup. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/28 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

a52 plugin crashing pulseaudio
by GitHub issues - edited 17 Jun '21

17 Jun '21

alsa-project/alsa-plugins issue #28 was edited from quequotion: Ever since f11e7a8994111f971bc881d4069f768472452f4f, the recent commit that brings the a52 plugin up to speed with ffmpeg's requirement for s32le format sampling, sending any audio via plug:a52 from any pulseaudio client crashes pulseaudio. I [posted this log](https://gist.github.com/quequotion/78972bac339f5249cf6b383519cdcedf) earlier in #23, in which audio from mpv crashes pulseaudio. I have tried other clients since and they all crash the same way. This is with a default pulseaudio configuration, as shipped in Archlinux, and the following `/etc/asound.conf`: ``` # Alsa Resampler defaults.pcm.rate_converter "speexrate_best" # Use PulseAudio by default pcm.!default { type pulse fallback "sysdefault" hint { show on description "Default ALSA Output (currently PulseAudio Sound Server)" } } ctl.!default { type pulse fallback "sysdefault" } defaults.namehint.showall on defaults.namehint.extended on ``` To enable the use of plug:a52 in pulseaudio without configuring it manually, I select the "Digital Surround 5.1 (IEC958/AC3) Output" profile in `pavucontrol`. Pulseaudio, as far as I know, is agnostic of how ALSA plugins work internally. This leads me to think there is something not getting done right in the new patchset, even if it is tolerable in an ALSA-only setup. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/28 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v2 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 17 Jun '21

17 Jun '21

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 405 ++++++++++++++++++++++++++++++++++ 4 files changed, 421 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index b4094f07214e..1b87225c39b0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..d495ee0f6a2d --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,405 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMOLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMOLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMOLE_RATE_MASK, + SUN50I_DMIC_SR_SAMOLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->bus_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.29.0

1 0

[PATCH v2 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 17 Jun '21

17 Jun '21

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 405 ++++++++++++++++++++++++++++++++++ 4 files changed, 421 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index b4094f07214e..1b87225c39b0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..d495ee0f6a2d --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,405 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMOLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMOLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMOLE_RATE_MASK, + SUN50I_DMIC_SR_SAMOLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->bus_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.29.0

1 0

[PATCH -next] ASoC: tegra: Fix missing clk_disable_unprepare() on error path
by Zou Wei 17 Jun '21

17 Jun '21

Fix the missing clk_disable_unprepare() before return from tegra_machine_hw_params() in the error handling case. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> --- sound/soc/tegra/tegra_asoc_machine.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_asoc_machine.c b/sound/soc/tegra/tegra_asoc_machine.c index a53aec3..397f326 100644 --- a/sound/soc/tegra/tegra_asoc_machine.c +++ b/sound/soc/tegra/tegra_asoc_machine.c @@ -306,6 +306,7 @@ static int tegra_machine_hw_params(struct snd_pcm_substream *substream, err = snd_soc_dai_set_sysclk(codec_dai, clk_id, mclk, SND_SOC_CLOCK_IN); if (err < 0) { + clk_disable_unprepare(machine->clk_cdev1); dev_err(card->dev, "codec_dai clock not set: %d\n", err); return err; } @@ -523,8 +524,10 @@ int tegra_asoc_machine_probe(struct platform_device *pdev) } err = devm_snd_soc_register_card(dev, card); - if (err) + if (err) { + clk_disable_unprepare(machine->clk_cdev1); return err; + } return 0; } -- 2.6.2

2 1

[PATCH 1/2] ASoC: rt711: remap buttons
by shumingf＠realtek.com 17 Jun '21

17 Jun '21

From: Shuming Fan <shumingf(a)realtek.com> This patch uses the same mapping as the machine driver: BTN_0 : KEY_PLAYPAUSE BTN_1 : KEY_VOICECOMMAND BTN_2 : KEY_VOLUMEUP BTN_3 : KEY_VOLUMEDOWN Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 9f5b2dc16c54..c3a82b462a93 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -154,26 +154,26 @@ static unsigned int rt711_button_detect(struct rt711_priv *rt711) case 0x0200: case 0x0100: case 0x0080: - btn_type |= SND_JACK_BTN_0; + btn_type |= SND_JACK_BTN_2; break; case 0x0001: - btn_type |= SND_JACK_BTN_3; + btn_type |= SND_JACK_BTN_1; break; } switch (val81) { case 0x8000: case 0x4000: case 0x2000: - btn_type |= SND_JACK_BTN_1; + btn_type |= SND_JACK_BTN_0; break; case 0x1000: case 0x0800: case 0x0400: - btn_type |= SND_JACK_BTN_2; + btn_type |= SND_JACK_BTN_3; break; case 0x0200: case 0x0100: - btn_type |= SND_JACK_BTN_3; + btn_type |= SND_JACK_BTN_1; break; } read_error: -- 2.31.1

2 1

a52 plugin crashing pulseaudio
by GitHub issues - opened 17 Jun '21

17 Jun '21

alsa-project/alsa-plugins issue #28 was opened from quequotion: Ever since f11e7a8994111f971bc881d4069f768472452f4f, the recent commit that brings the a52 plugin up to speed with ffmpeg's requirement for s32le format sampling, sending any audio via plug:a52 from any pulseaudio client crashes pulseaudio. I [posted this log](https://gist.github.com/quequotion/78972bac339f5249cf6b383519cdcedf) earlier in #23, in which audio from mpv crashes pulseaudio. I have tried other clients since and they all crash the same way. This is with a default pulseaudio configuration, as shipped in Archlinux, and the following `/etc/asound.conf`: ``` # Alsa Resampler defaults.pcm.rate_converter "speexrate_best" # Use PulseAudio by default pcm.!default { type pulse fallback "sysdefault" hint { show on description "Default ALSA Output (currently PulseAudio Sound Server)" } } ctl.!default { type pulse fallback "sysdefault" } defaults.namehint.showall on defaults.namehint.extended on ``` To enable the use of plug:52 in pulseaudio without configuring it manually, I select the "Digital Surround 5.1 (IEC958/AC3) Output" profile in `pavucontrol`. Pulseaudio, as far as I know, is agnostic of how ALSA plugins work internally. This leads me to think there is something not getting done right in the new patchset, even if it is tolerable in an ALSA-only setup. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/28 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH -next] ASoC: tegra: Add missing of_node_put() in tegra_machine_parse_phandle()
by Zou Wei 17 Jun '21

17 Jun '21

The function is missing a of_node_put on node, fix this by adding the call before returning. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> --- sound/soc/tegra/tegra_asoc_machine.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_asoc_machine.c b/sound/soc/tegra/tegra_asoc_machine.c index 397f326..cba55ca 100644 --- a/sound/soc/tegra/tegra_asoc_machine.c +++ b/sound/soc/tegra/tegra_asoc_machine.c @@ -336,9 +336,12 @@ tegra_machine_parse_phandle(struct device *dev, const char *name) } err = devm_add_action_or_reset(dev, tegra_machine_node_release, np); - if (err) + if (err) { + of_node_put(np); return ERR_PTR(err); + } + of_node_put(np); return np; } -- 2.6.2

2 1

[PATCH] ALSA: firewire-motu: fix stream format for MOTU 8pre FireWire
by Takashi Sakamoto 17 Jun '21

17 Jun '21

My previous refactoring for ALSA firewire-motu driver brought regression to handle MOTU 8pre FireWire. The packet format is not operated correctly. Cc: <stable(a)vger.kernel.org> Fixes: dfbaa4dc11eb ("ALSA: firewire-motu: add model-specific table of chunk count") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-protocol-v2.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c index e59e69ab1538..784073aa1026 100644 --- a/sound/firewire/motu/motu-protocol-v2.c +++ b/sound/firewire/motu/motu-protocol-v2.c @@ -353,6 +353,7 @@ const struct snd_motu_spec snd_motu_spec_8pre = { .protocol_version = SND_MOTU_PROTOCOL_V2, .flags = SND_MOTU_SPEC_RX_MIDI_2ND_Q | SND_MOTU_SPEC_TX_MIDI_2ND_Q, - .tx_fixed_pcm_chunks = {10, 6, 0}, - .rx_fixed_pcm_chunks = {10, 6, 0}, + // Two dummy chunks always in the end of data block. + .tx_fixed_pcm_chunks = {10, 10, 0}, + .rx_fixed_pcm_chunks = {6, 6, 0}, }; -- 2.30.2

2 1

[PATCH alsa-plugins 0/2] Support multiple formats
by Takashi Iwai 17 Jun '21

17 Jun '21

Hi, this is a patch set to add the multi-format support for the existing rate plugins. Takashi === Takashi Iwai (2): rate-lav: Support multiple formats samplerate: Support multiple formats rate-lav/rate_lavrate.c | 115 ++++++++++++++++++++++++++++++++++------ rate/rate_samplerate.c | 99 ++++++++++++++++++++++++++++------ 2 files changed, 183 insertions(+), 31 deletions(-) -- 2.26.2

1 2

[PATCH 0/2] ALSA: firewire-motu: add support for MOTU 828 and 896
by Takashi Sakamoto 17 Jun '21

17 Jun '21

Hi, This patchset is to add support for Mark of the unicorn (MOTU) 828 and 896, which were shipped two decades ago (2001) and already discontinued. (yes, it's 2021). It's reasonable to count them as first generation of MOTU FireWire series since they are based on QuickLogic QuickRAM FPGA apart from Altera or Xilinx FPGA in the latter generation. Unlike the latter generation, they doesn't allow software to configure internal multiplexer. I note that the sequence replay for media clock recovery[1] allows to support them. The device doesn't generate better sound with nominal (ideal) sequence of packets, thus the patches should be applied to development tree for v5.14 kernel. [1] [PATCH 0/3] ALSA: firewire-motu: media clock recovery for sph-aware devices https://lore.kernel.org/alsa-devel/20210602013406.26442-1-o-takashi@sakamoc… Takashi Sakamoto (2): ALSA: firewire-motu: add support for MOTU 828 ALSA: firewire-motu: add support for MOTU 896 sound/firewire/Kconfig | 2 + sound/firewire/motu/Makefile | 3 +- sound/firewire/motu/motu-protocol-v1.c | 440 +++++++++++++++++++++++++ sound/firewire/motu/motu.c | 2 + sound/firewire/motu/motu.h | 24 ++ 5 files changed, 470 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/motu/motu-protocol-v1.c -- 2.30.2

2 4

[PATCH alsa-lib 0/2] Allow multiple formats for PCM rate plugin
by Takashi Iwai 17 Jun '21

17 Jun '21

Hi, this is a patch set for alsa-lib to extend the PCM rate plugin for allowing multiple formats for its converter plugins. This optimizes and improves the quality (skips the unnecessary buffer and conversion) when the converter supports the source and/or the target format natively. The patches for libswresample and libsamplerate will follow. Takashi === Takashi Iwai (2): pcm: rate: Refactoring temporary buffer allocations pcm: rate: Improve the support multiple formats include/pcm_rate.h | 36 +++- src/pcm/pcm_rate.c | 424 +++++++++++++++++++++++++++------------------ 2 files changed, 293 insertions(+), 167 deletions(-) -- 2.26.2

1 2

ALSA: usb-audio: fix rate on Ozone Z90 USB headset
by Daehwan Jung 17 Jun '21

17 Jun '21

It mislabels its 96 kHz altsetting and that's why it causes some noise Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Daehwan Jung <dh10.jung(a)samsung.com> --- sound/usb/format.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 2287f8c..eb216fe 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -223,9 +223,11 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof continue; /* C-Media CM6501 mislabels its 96 kHz altsetting */ /* Terratec Aureon 7.1 USB C-Media 6206, too */ + /* Ozone Z90 USB C-Media, too */ if (rate == 48000 && nr_rates == 1 && (chip->usb_id == USB_ID(0x0d8c, 0x0201) || chip->usb_id == USB_ID(0x0d8c, 0x0102) || + chip->usb_id == USB_ID(0x0d8c, 0x0078) || chip->usb_id == USB_ID(0x0ccd, 0x00b1)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; -- 2.7.4

3 2

HDA-Intel: add Boost volume control for Headset Mic
by GitHub pull_request - opened 17 Jun '21

17 Jun '21

alsa-project/alsa-ucm-conf pull request #101 was opened from jason77-wang: Ubuntu Linux users complain even they set the input volume to maximum, they still can't record sound with a pretty satisfied volume, they need to run commandline commands to adjust "Mic Boost/Headset Mic Boost/Headphone Mic boost" value to increase the input volume. That is because the current ucm only defines "Capture Volume" to control the input volume. Here we add Boost control, then users could adjust both "Capture Volume" and "Boost Volume" through Pulseaudio and Gnome. BugLink: https://bugs.launchpad.net/bugs/1930188 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/101 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/101.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: max98357a: set channels_max to 4
by Judy Hsiao 16 Jun '21

16 Jun '21

Sets channels_max to 4 to support QUAD channel. Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- sound/soc/codecs/max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 918812763884..c3cf1743caad 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -117,7 +117,7 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { .rate_min = 8000, .rate_max = 96000, .channels_min = 1, - .channels_max = 2, + .channels_max = 4, }, .ops = &max98357a_dai_ops, }; -- 2.31.0

5 6

[PATCH -next 1/3] ASoC: rockchip: i2s: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 16 Jun '21

16 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/rockchip/rockchip_i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 0740764e7f71..c7dc3509bceb 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -618,8 +618,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) return PTR_ERR(i2s->mclk); } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - regs = devm_ioremap_resource(&pdev->dev, res); + regs = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(regs)) return PTR_ERR(regs); -- 2.25.1

2 3

[PATCH -next] SoC: codecs: wcd938x: fix boolreturn.cocci warning
by Yang Li 16 Jun '21

16 Jun '21

Return statements in functions returning bool should use true/false instead of 1/0. Fix the following coccicheck warning: ./sound/soc/codecs/wcd938x.c:1190:9-10: WARNING: return of 0/1 in function 'wcd938x_volatile_register' with return type bool. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/codecs/wcd938x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 2cf6145..cfe00536 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -1187,7 +1187,7 @@ static bool wcd938x_writeable_register(struct device *dev, unsigned int reg) static bool wcd938x_volatile_register(struct device *dev, unsigned int reg) { if (reg <= WCD938X_BASE_ADDRESS) - return 0; + return false; if (reg == WCD938X_DIGITAL_SWR_TX_CLK_RATE) return true; -- 1.8.3.1

2 1

[PATCH] ASoC: codecs: Fix duplicate included sound/soc.h
by Jiapeng Chong 16 Jun '21

16 Jun '21

Clean up the following includecheck warnings: ./sound/soc/codecs/wcd938x.c: sound/soc.h is included more than once. ./sound/soc/codecs/wcd938x-sdw.c: sound/soc.h is included more than once. No functional change. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/wcd938x-sdw.c | 1 - sound/soc/codecs/wcd938x.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/codecs/wcd938x-sdw.c b/sound/soc/codecs/wcd938x-sdw.c index d82c40e..88334d3 100644 --- a/sound/soc/codecs/wcd938x-sdw.c +++ b/sound/soc/codecs/wcd938x-sdw.c @@ -7,7 +7,6 @@ #include <linux/device.h> #include <linux/kernel.h> #include <linux/component.h> -#include <sound/soc.h> #include <linux/pm_runtime.h> #include <linux/irqdomain.h> #include <linux/of.h> diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 2cf6145..c0d7452 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -9,7 +9,6 @@ #include <linux/kernel.h> #include <linux/pm_runtime.h> #include <linux/component.h> -#include <sound/soc.h> #include <sound/tlv.h> #include <linux/of_gpio.h> #include <linux/of.h> -- 1.8.3.1

2 1

[PATCH] ASoC: hdmi-codec: make hdmi_codec_controls static
by Jiapeng Chong 16 Jun '21

16 Jun '21

This symbol is not used outside of hdmi-codec.c, so marks it static. Fix the following sparse warning: sound/soc/codecs/hdmi-codec.c:750:25: warning: symbol 'hdmi_codec_controls' was not declared. Should it be static? Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index e0460a9..b61f980 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -747,7 +747,7 @@ static int hdmi_codec_mute(struct snd_soc_dai *dai, int mute, int direction) SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE |\ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) -struct snd_kcontrol_new hdmi_codec_controls[] = { +static struct snd_kcontrol_new hdmi_codec_controls[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, -- 1.8.3.1

2 1

[PATCH] ASoC: cs42l42: Correct definition of CS42L42_ADC_PDN_MASK
by Richard Fitzgerald 16 Jun '21

16 Jun '21

The definition of CS42L42_ADC_PDN_MASK was incorrectly defined as the HP_PDN bit. Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 7bf05ff05f74..206b3c81d3e0 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -79,7 +79,7 @@ #define CS42L42_HP_PDN_SHIFT 3 #define CS42L42_HP_PDN_MASK (1 << CS42L42_HP_PDN_SHIFT) #define CS42L42_ADC_PDN_SHIFT 2 -#define CS42L42_ADC_PDN_MASK (1 << CS42L42_HP_PDN_SHIFT) +#define CS42L42_ADC_PDN_MASK (1 << CS42L42_ADC_PDN_SHIFT) #define CS42L42_PDN_ALL_SHIFT 0 #define CS42L42_PDN_ALL_MASK (1 << CS42L42_PDN_ALL_SHIFT) -- 2.20.1

2 1

[PATCH v2] media: Fix Media Controller API config checks
by Shuah Khan 16 Jun '21

16 Jun '21

Smatch static checker warns that "mdev" can be null: sound/usb/media.c:287 snd_media_device_create() warn: 'mdev' can also be NULL If CONFIG_MEDIA_CONTROLLER is disabled, this file should not be included in the build. The below conditions in the sound/usb/Makefile are in place to ensure that media.c isn't included in the build. sound/usb/Makefile: snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o select SND_USB_AUDIO_USE_MEDIA_CONTROLLER if MEDIA_CONTROLLER && (MEDIA_SUPPORT=y || MEDIA_SUPPORT=SND_USB_AUDIO) The following config check in include/media/media-dev-allocator.h is in place to enable the API only when CONFIG_MEDIA_CONTROLLER and CONFIG_USB are enabled. #if defined(CONFIG_MEDIA_CONTROLLER) && defined(CONFIG_USB) This check doesn't work as intended when CONFIG_USB=m. When CONFIG_USB=m, CONFIG_USB_MODULE is defined and CONFIG_USB is not. The above config check doesn't catch that CONFIG_USB is defined as a module and disables the API. This results in sound/usb enabling Media Controller specific ALSA driver code, while Media disables the Media Controller API. Fix the problem requires two changes: 1. Change the check to use IS_ENABLED to detect when CONFIG_USB is enabled as a module or static. Since CONFIG_MEDIA_CONTROLLER is a bool, leave the check unchanged to be consistent with drivers/media/Makefile. 2. Change the drivers/media/mc/Makefile to include mc-dev-allocator.o in mc-objs when CONFIG_USB is enabled. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/alsa-devel/YLeAvT+R22FQ%2FEyw@mwanda/ Signed-off-by: Shuah Khan <skhan(a)linuxfoundation.org> --- Changes since v1: - Fixed the drivers/media/mc/Makefile incorrect logic to check CONFIG_USB value. Test with CONFIG_USB_SUPPORT enabled/disabled CONFIG_USB y/m and disabled. drivers/media/mc/Makefile | 2 +- include/media/media-dev-allocator.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/media/mc/Makefile b/drivers/media/mc/Makefile index 119037f0e686..2b7af42ba59c 100644 --- a/drivers/media/mc/Makefile +++ b/drivers/media/mc/Makefile @@ -3,7 +3,7 @@ mc-objs := mc-device.o mc-devnode.o mc-entity.o \ mc-request.o -ifeq ($(CONFIG_USB),y) +ifneq ($(CONFIG_USB),) mc-objs += mc-dev-allocator.o endif diff --git a/include/media/media-dev-allocator.h b/include/media/media-dev-allocator.h index b35ea6062596..2ab54d426c64 100644 --- a/include/media/media-dev-allocator.h +++ b/include/media/media-dev-allocator.h @@ -19,7 +19,7 @@ struct usb_device; -#if defined(CONFIG_MEDIA_CONTROLLER) && defined(CONFIG_USB) +#if defined(CONFIG_MEDIA_CONTROLLER) && IS_ENABLED(CONFIG_USB) /** * media_device_usb_allocate() - Allocate and return struct &media device * -- 2.30.2

1 0

HDSPconf don't recognize RME RayDat
by GitHub issues - opened 16 Jun '21

16 Jun '21

alsa-project/alsa-tools issue #6 was opened from profejuanca: Hi, I'm using a RME Raydat (PCI express) many years ago, but HDSPconf don't recognize my card never, hdspmixer and the rest of jack/alsa tools see my card without problems, but hdspconf in various distros never sees my card. The output is: `HDSPConf 1.4 - Copyright (C) 2003 Thomas Charbonnel <thomas(a)undata.org> This program comes WITH ABSOLUTELY NO WARRANTY HDSPConf is free software, see the file copying for details Looking for HDSP cards : Card 0 : RME RayDAT S/N 0x3250cb at 0xfe300000, irq 57 Card 1 : HDA ATI SB at 0xfe800000 irq 16 Card 2 : HDA NVidia at 0xfe080000 irq 60 No Hammerfall DSP card found. ` Card 0 es my RME card... I suposse that raydat is not in the list of the tool. Issue URL : https://github.com/alsa-project/alsa-tools/issues/6 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH 00/10] ASoC: Intel: machine driver corrections
by Pierre-Louis Bossart 16 Jun '21

16 Jun '21

Some of these patches dealing with Sparse warnings were submitted earlier but not merged. I grouped them with a couple of fixes from Kai. Kai Vehmanen (3): ASoC: Intel: sof_sdw: use mach data for ADL RVP DMIC count ASoC: Intel: sof_sdw: remove hdac-hdmi support ASoC: Intel: skl_hda_dsp_generic: Update Kconfig documentation Pierre-Louis Bossart (7): ASoC: Intel: sof_rt5682: shrink platform_id names below 20 characters ASoC: Intel: glk_rt5682_max98357a: shrink platform_id below 20 characters ASoC: Intel: kbl_da7219_max98357a: shrink platform_id below 20 characters ASoC: Intel: sof_da7219_max98373: shrink platform_id below 20 characters ASoC: Intel: sof_sdw: fix signed/unsigned warning ASoC: Intel: soc-acpi: add ull suffix for SoundWire _ADR values ASoC: Intel: use MODULE_DEVICE_TABLE with platform_device_id tables sound/soc/intel/boards/Kconfig | 7 ++-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 4 +- sound/soc/intel/boards/bxt_rt298.c | 3 +- sound/soc/intel/boards/ehl_rt5660.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 4 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 4 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 5 +-- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 3 +- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- .../soc/intel/boards/skl_nau88l25_max98357a.c | 3 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 3 +- sound/soc/intel/boards/skl_rt286.c | 3 +- sound/soc/intel/boards/sof_cs42l42.c | 3 +- sound/soc/intel/boards/sof_da7219_max98373.c | 6 +-- sound/soc/intel/boards/sof_rt5682.c | 20 +++------- sound/soc/intel/boards/sof_sdw.c | 5 +-- sound/soc/intel/boards/sof_sdw_common.h | 1 - sound/soc/intel/boards/sof_sdw_hdmi.c | 37 +----------------- .../intel/common/soc-acpi-intel-adl-match.c | 28 +++++++------- .../intel/common/soc-acpi-intel-cml-match.c | 20 +++++----- .../intel/common/soc-acpi-intel-cnl-match.c | 2 +- .../intel/common/soc-acpi-intel-glk-match.c | 2 +- .../intel/common/soc-acpi-intel-icl-match.c | 12 +++--- .../intel/common/soc-acpi-intel-jsl-match.c | 6 +-- .../intel/common/soc-acpi-intel-kbl-match.c | 2 +- .../intel/common/soc-acpi-intel-tgl-match.c | 38 +++++++++---------- 27 files changed, 83 insertions(+), 144 deletions(-) -- 2.25.1

2 11

alsa-lib errors on boot
by GitHub issues - opened 16 Jun '21

16 Jun '21

alsa-project/alsa-lib issue #159 was opened from miquecg: ``` alsactl[769]: alsa-lib parser.c:242:(error_node) UCM is not supported for this HDA model (HD-Audio Generic at 0xfd3c8000 irq 108) alsactl[769]: alsa-lib main.c:1405:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -6 alsactl[769]: alsa-lib parser.c:242:(error_node) UCM is not supported for this HDA model (HD-Audio Generic at 0xfd3c0000 irq 109) alsactl[769]: alsa-lib main.c:1405:(snd_use_case_mgr_open) error: failed to import hw:1 use case configuration -6 ``` This is happening on a ThinkPad T14 AMD running Arch Linux and `alsa-lib` 1.2.5.1. Previous version also had this issue. Issue URL : https://github.com/alsa-project/alsa-lib/issues/159 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 2/2] ASoC: sun50i-dmic: dt-bindings: add DT bindings for DMIC controller
by Ban Tao 16 Jun '21

16 Jun '21

The Allwinner SoCs feature an I2S controller across multiple SoC generations. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- .../sound/allwinner,sun50i-h6-dmic.yaml | 66 +++++++++++++++++++ 1 file changed, 66 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml new file mode 100644 index 000000000000..81d40c374e44 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner H6 DMIC Controller Device Tree Bindings + +maintainers: + - Ban Tao <fengzheng923(a)gmail.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: allwinner,sun50i-h6-dmic + + reg: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: dmic + + dmas: + maxItems: 1 + + dma-names: + const: rx + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - clocks + - clock-names + - dmas + - dma-names + - resets + +additionalProperties: false + +examples: + - | + spdif: spdif@5095000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun50i-h6-dmic"; + reg = <0x05095000 0x400>; + clocks = <&ccu CLK_BUS_DMIC>, <&ccu CLK_DMIC>; + clock-names = "apb", "dmic"; + dmas = <&dma 7>; + dma-names = "rx"; + resets = <&ccu RST_BUS_DMIC>; + }; + +... -- 2.29.0

4 4

[PATCH] ALSA: oss: Remove redundant assignment to src_access
by Jiapeng Chong 16 Jun '21

16 Jun '21

Variable src_access is set to 'dst_access', but this value is never read as it is not used later on, hence it is a redundant assignment and can be removed. Clean up the following clang-analyzer warning: sound/core/oss/pcm_plugin.c:472:4: warning: Value stored to 'src_access' is never read [clang-analyzer-deadcode.DeadStores]. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/core/oss/pcm_plugin.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 061ba06..a7c0e50 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -469,7 +469,6 @@ int snd_pcm_plug_format_plugins(struct snd_pcm_substream *plug, return err; } srcformat = tmpformat; - src_access = dst_access; } tmpformat.rate = dstformat.rate; err = snd_pcm_plugin_build_rate(plug, -- 1.8.3.1

2 1

ALSA: usb-audio: Fix rate on Ozone Z90 USB headset
by Daehwan Jung 16 Jun '21

16 Jun '21

It mislabels its 96 kHz altsetting and that's why it causes some noise Signed-off-by: Daehwan Jung <dh10.jung(a)samsung.com> --- sound/usb/format.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 2287f8c..eb216fe 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -223,9 +223,11 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof continue; /* C-Media CM6501 mislabels its 96 kHz altsetting */ /* Terratec Aureon 7.1 USB C-Media 6206, too */ + /* Ozone Z90 USB C-Media, too */ if (rate == 48000 && nr_rates == 1 && (chip->usb_id == USB_ID(0x0d8c, 0x0201) || chip->usb_id == USB_ID(0x0d8c, 0x0102) || + chip->usb_id == USB_ID(0x0d8c, 0x0078) || chip->usb_id == USB_ID(0x0ccd, 0x00b1)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; -- 2.7.4

2 1

ThinkPad T14 AMD microphone LED always ON
by GitHub issues - opened 16 Jun '21

16 Jun '21

alsa-project/alsa-ucm-conf issue #100 was opened from miquecg: I'm running Arch Linux with `alsa-ucm-conf` 1.2.5.1. This problem was present in previous versions and it's being discussed on a [forum thread](https://bbs.archlinux.org/viewtopic.php?pid=1977273) among other things. So the problem is that microphone LED is always ON but mute/unmute behaviour still works. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/100 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

rate-lav: Convert to libswresample
by GitHub pull_request - opened 16 Jun '21

16 Jun '21

alsa-project/alsa-plugins pull request #27 was opened from tiwai: The libavresample has been deprecated. Convert to the new API for libswresample. The phase shift and cutoff seem to be either redundant or not-working properly, so those are dropped. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Request URL : https://github.com/alsa-project/alsa-plugins/pull/27 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/27.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH -next] ASoC: hdmi-codec: Make symbol 'hdmi_codec_controls' static
by Wei Yongjun 16 Jun '21

16 Jun '21

The sparse tool complains as follows: sound/soc/codecs/hdmi-codec.c:750:25: warning: symbol 'hdmi_codec_controls' was not declared. Should it be static? This symbol is not used outside of hdmi-codec.c, so marks it static. Fixes: 366b45b97448 ("ASoC: hdmi-codec: Rework to support more controls") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index e0460a9bd17a..b61f980cabdc 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -747,7 +747,7 @@ static const struct snd_soc_dai_ops hdmi_codec_spdif_dai_ops = { SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE |\ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) -struct snd_kcontrol_new hdmi_codec_controls[] = { +static struct snd_kcontrol_new hdmi_codec_controls[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM,

2 1

[PATCH -next] ALSA: fm801: Fix missing pci_release_regions() on error in snd_fm801_create()
by Yang Yingliang 16 Jun '21

16 Jun '21

Fix the missing pci_release_regions() before return from snd_fm801_create() in the error handling case. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/pci/fm801.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index ed9dae87145b..adac9c6a67f0 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1254,6 +1254,7 @@ static int snd_fm801_create(struct snd_card *card, IRQF_SHARED, KBUILD_MODNAME, chip)) { dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); snd_fm801_free(chip); + pci_release_regions(pci); return -EBUSY; } chip->irq = pci->irq; @@ -1266,6 +1267,7 @@ static int snd_fm801_create(struct snd_card *card, err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { snd_fm801_free(chip); + pci_release_regions(pci); return err; } @@ -1273,6 +1275,7 @@ static int snd_fm801_create(struct snd_card *card, err = v4l2_device_register(&pci->dev, &chip->v4l2_dev); if (err < 0) { snd_fm801_free(chip); + pci_release_regions(pci); return err; } chip->tea.v4l2_dev = &chip->v4l2_dev; @@ -1285,6 +1288,7 @@ static int snd_fm801_create(struct snd_card *card, if (snd_tea575x_init(&chip->tea, THIS_MODULE)) { dev_err(card->dev, "TEA575x radio not found\n"); snd_fm801_free(chip); + pci_release_regions(pci); return -ENODEV; } } else if ((chip->tea575x_tuner & TUNER_TYPE_MASK) == 0) { -- 2.25.1

2 2

[PATCH][next] ALSA: bebob: Fix bit flag quirk constants
by Colin King 16 Jun '21

16 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> The quirking bit-flags are currently set as contiguous integer enum values and so currently SND_BEBOB_QUIRK_INITIAL_DISCONTINUOUS_DBC is zero and so he quirking never getting set or tested correctly for this quirk. Fix this by setting the quirking constants as shifted bit values. Addresses-Coverity: ("Bitwise-and with zero") Fixes: 93cd12d6e88a ("ALSA: bebob: code refactoring for model-dependent quirks") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/firewire/bebob/bebob.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index c06579d9380e..4d73ecb30d79 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -76,8 +76,8 @@ struct snd_bebob_spec { }; enum snd_bebob_quirk { - SND_BEBOB_QUIRK_INITIAL_DISCONTINUOUS_DBC, - SND_BEBOB_QUIRK_WRONG_DBC, + SND_BEBOB_QUIRK_INITIAL_DISCONTINUOUS_DBC = (1 << 0), + SND_BEBOB_QUIRK_WRONG_DBC = (1 << 1), }; struct snd_bebob { -- 2.31.1

3 2

[PATCH -next] ALSA: ppc: fix error return code in snd_pmac_probe()
by Yang Yingliang 16 Jun '21

16 Jun '21

If snd_pmac_tumbler_init() or snd_pmac_tumbler_post_init() fails, snd_pmac_probe() need return error code. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/ppc/powermac.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 1d2865c43d4b..db414b61157e 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -79,7 +79,11 @@ static int snd_pmac_probe(struct platform_device *devptr) sprintf(card->shortname, "PowerMac %s", name_ext); sprintf(card->longname, "%s (Dev %d) Sub-frame %d", card->shortname, chip->device_id, chip->subframe); - if ( snd_pmac_tumbler_init(chip) < 0 || snd_pmac_tumbler_post_init() < 0) + err = snd_pmac_tumbler_init(chip); + if (err < 0) + goto __error; + err = snd_pmac_tumbler_post_init(); + if (err < 0) goto __error; break; case PMAC_AWACS: -- 2.25.1

2 1

Fix a52 for the recent avcodec
by GitHub pull_request - opened 16 Jun '21

16 Jun '21

alsa-project/alsa-plugins pull request #26 was opened from tiwai: Request URL : https://github.com/alsa-project/alsa-plugins/pull/26 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/26.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] media: Fix Media Controller API config checks
by Shuah Khan 16 Jun '21

16 Jun '21

Smatch static checker warns that "mdev" can be null: sound/usb/media.c:287 snd_media_device_create() warn: 'mdev' can also be NULL If CONFIG_MEDIA_CONTROLLER is disabled, this file should not be included in the build. The below conditions in the sound/usb/Makefile are in place to ensure that media.c isn't included in the build. sound/usb/Makefile: snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o select SND_USB_AUDIO_USE_MEDIA_CONTROLLER if MEDIA_CONTROLLER && (MEDIA_SUPPORT=y || MEDIA_SUPPORT=SND_USB_AUDIO) The following config check in include/media/media-dev-allocator.h is in place to enable the API only when CONFIG_MEDIA_CONTROLLER and CONFIG_USB are enabled. #if defined(CONFIG_MEDIA_CONTROLLER) && defined(CONFIG_USB) This check doesn't work as intended when CONFIG_USB=m. When CONFIG_USB=m, CONFIG_USB_MODULE is defined and CONFIG_USB is not. The above config check doesn't catch that CONFIG_USB is defined as a module and disables the API. This results in sound/usb enabling Media Controller specific ALSA driver code, while Media disables the Media Controller API. Fix the problem requires two changes: 1. Change the check to use IS_ENABLED to detect when CONFIG_USB is enabled as a module or static. Since CONFIG_MEDIA_CONTROLLER is a bool, leave the check unchanged to be consistent with drivers/media/Makefile. 2. Change the drivers/media/mc/Makefile to include mc-dev-allocator.o in mc-objs when CONFIG_USB is y or m. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/alsa-devel/YLeAvT+R22FQ%2FEyw@mwanda/ Signed-off-by: Shuah Khan <skhan(a)linuxfoundation.org> --- drivers/media/mc/Makefile | 2 +- include/media/media-dev-allocator.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/media/mc/Makefile b/drivers/media/mc/Makefile index 119037f0e686..140f0a78540e 100644 --- a/drivers/media/mc/Makefile +++ b/drivers/media/mc/Makefile @@ -3,7 +3,7 @@ mc-objs := mc-device.o mc-devnode.o mc-entity.o \ mc-request.o -ifeq ($(CONFIG_USB),y) +ifeq ($(CONFIG_USB),$(filter $(CONFIG_USB),y m)) mc-objs += mc-dev-allocator.o endif diff --git a/include/media/media-dev-allocator.h b/include/media/media-dev-allocator.h index b35ea6062596..2ab54d426c64 100644 --- a/include/media/media-dev-allocator.h +++ b/include/media/media-dev-allocator.h @@ -19,7 +19,7 @@ struct usb_device; -#if defined(CONFIG_MEDIA_CONTROLLER) && defined(CONFIG_USB) +#if defined(CONFIG_MEDIA_CONTROLLER) && IS_ENABLED(CONFIG_USB) /** * media_device_usb_allocate() - Allocate and return struct &media device * -- 2.30.2

2 4

conf: refine check for snd_config_is_array
by GitHub pull_request - opened 16 Jun '21

16 Jun '21

alsa-project/alsa-lib pull request #158 was opened from ranj063: Apart from checking the index, also check if the node is not compound to qualify a config as an array. For ex: mixer [ "mixer1" ] is an array but mixer.0 { name "mixer 1" } is not an array even though the id is 0. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/158 Patch URL : https://github.com/alsa-project/alsa-lib/pull/158.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Build fails with "undefined macro AM_GNU_GETTEXT"
by GitHub issues - edited 15 Jun '21

15 Jun '21

alsa-project/alsa-utils issue #99 was edited from edward-a: **./gitcompile** fails with the following output: configure.ac:10: warning: macro 'AM_GNU_GETTEXT' not found in library configure.ac:11: warning: macro 'AM_GNU_GETTEXT_VERSION' not found in library ./gitcompile: line 10: gettextize: command not found configure.ac:15: installing './compile' configure.ac:6: installing './install-sh' configure.ac:6: installing './missing' alsactl/Makefile.am: installing './depcomp' axfer/test/Makefile.am:9: warning: source file '../container.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled automake: warning: possible forward-incompatibility. automake: At least a source file is in a subdirectory, but the 'subdir-objects' automake: automake option hasn't been enabled. For now, the corresponding output automake: object file(s) will be placed in the top-level directory. However, automake: this behaviour will change in future Automake versions: they will automake: unconditionally cause object files to be placed in the same subdirectory automake: of the corresponding sources. automake: You are advised to start using 'subdir-objects' option throughout your automake: project, to avoid future incompatibilities. axfer/test/Makefile.am:9: warning: source file '../container-riff-wave.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:9: warning: source file '../container-au.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:9: warning: source file '../container-voc.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:9: warning: source file '../container-raw.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-riff-wave.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-au.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-voc.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-raw.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../mapper.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../mapper-single.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../mapper-multiple.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled parallel-tests: installing './test-driver' **configure.ac:10: error: possibly undefined macro: AM_GNU_GETTEXT If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. configure.ac:11: error: possibly undefined macro: AM_GNU_GETTEXT_VERSION** CFLAGS=-O2 -Wall -pipe -g ./configure checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for a thread-safe mkdir -p... /usr/bin/mkdir -p checking for gawk... gawk checking whether make sets $(MAKE)... yes checking whether make supports nested variables... yes checking whether to enable maintainer-specific portions of Makefiles... yes ./configure: line 2937: syntax error near unexpected token `external' ./configure: line 2937: `AM_GNU_GETTEXT(external)' ### System info: DISTRIB_ID=Ubuntu DISTRIB_RELEASE=20.04 DISTRIB_CODENAME=focal DISTRIB_DESCRIPTION="Ubuntu 20.04.2 LTS" NAME="Ubuntu" VERSION="20.04.2 LTS (Focal Fossa)" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 20.04.2 LTS" VERSION_ID="20.04" HOME_URL="https://www.ubuntu.com/" SUPPORT_URL="https://help.ubuntu.com/" BUG_REPORT_URL="https://bugs.launchpad.net/ubuntu/" PRIVACY_POLICY_URL="https://www.ubuntu.com/legal/terms-and-policies/privacy-policy" VERSION_CODENAME=focal UBUNTU_CODENAME=focal Issue URL : https://github.com/alsa-project/alsa-utils/issues/99 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Build failes with "undefined macro AM_GNU_GETTEXT"
by GitHub issues - opened 15 Jun '21

15 Jun '21

alsa-project/alsa-utils issue #99 was opened from edward-a: **./gitcompile** fails with the following output: configure.ac:10: warning: macro 'AM_GNU_GETTEXT' not found in library configure.ac:11: warning: macro 'AM_GNU_GETTEXT_VERSION' not found in library ./gitcompile: line 10: gettextize: command not found configure.ac:15: installing './compile' configure.ac:6: installing './install-sh' configure.ac:6: installing './missing' alsactl/Makefile.am: installing './depcomp' axfer/test/Makefile.am:9: warning: source file '../container.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled automake: warning: possible forward-incompatibility. automake: At least a source file is in a subdirectory, but the 'subdir-objects' automake: automake option hasn't been enabled. For now, the corresponding output automake: object file(s) will be placed in the top-level directory. However, automake: this behaviour will change in future Automake versions: they will automake: unconditionally cause object files to be placed in the same subdirectory automake: of the corresponding sources. automake: You are advised to start using 'subdir-objects' option throughout your automake: project, to avoid future incompatibilities. axfer/test/Makefile.am:9: warning: source file '../container-riff-wave.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:9: warning: source file '../container-au.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:9: warning: source file '../container-voc.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:9: warning: source file '../container-raw.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-riff-wave.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-au.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-voc.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-raw.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../mapper.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../mapper-single.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../mapper-multiple.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled parallel-tests: installing './test-driver' **configure.ac:10: error: possibly undefined macro: AM_GNU_GETTEXT If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. configure.ac:11: error: possibly undefined macro: AM_GNU_GETTEXT_VERSION** CFLAGS=-O2 -Wall -pipe -g ./configure checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for a thread-safe mkdir -p... /usr/bin/mkdir -p checking for gawk... gawk checking whether make sets $(MAKE)... yes checking whether make supports nested variables... yes checking whether to enable maintainer-specific portions of Makefiles... yes ./configure: line 2937: syntax error near unexpected token `external' ./configure: line 2937: `AM_GNU_GETTEXT(external)' ### System info: DISTRIB_ID=Ubuntu DISTRIB_RELEASE=20.04 DISTRIB_CODENAME=focal DISTRIB_DESCRIPTION="Ubuntu 20.04.2 LTS" NAME="Ubuntu" VERSION="20.04.2 LTS (Focal Fossa)" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 20.04.2 LTS" VERSION_ID="20.04" HOME_URL="https://www.ubuntu.com/" SUPPORT_URL="https://help.ubuntu.com/" BUG_REPORT_URL="https://bugs.launchpad.net/ubuntu/" PRIVACY_POLICY_URL="https://www.ubuntu.com/legal/terms-and-policies/privacy-policy" VERSION_CODENAME=focal UBUNTU_CODENAME=focal Issue URL : https://github.com/alsa-project/alsa-utils/issues/99 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH -next] ASoC: dwc: dwc-i2s: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 15 Jun '21

15 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/dwc/dwc-i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index fd4160289fac..8ebf76e04702 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -636,8 +636,7 @@ static int dw_i2s_probe(struct platform_device *pdev) dw_i2s_dai->ops = &dw_i2s_dai_ops; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - dev->i2s_base = devm_ioremap_resource(&pdev->dev, res); + dev->i2s_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(dev->i2s_base)) return PTR_ERR(dev->i2s_base); -- 2.25.1

2 1

[PATCH] ASoC: codecs: wcd938x: remove incorrect module interdependency
by Srinivas Kandagatla 15 Jun '21

15 Jun '21

For some reason we ended up with cyclic dependency between snd_soc_wcd938x and snd_soc_wcd938x_sdw modules. Remove this cyclic dependency by handling them in respective modules. Without this below error is reported during make modules_install depmod: ERROR: Cycle detected: snd_soc_wcd938x -> snd_soc_wcd938x_sdw -> snd_soc_wcd938x depmod: ERROR: Found 2 modules in dependency cycles! Fixes: 045442228868 ("ASoC: codecs: wcd938x: add audio routing and Kconfig") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wcd938x-sdw.c | 19 ++++++++++++------- sound/soc/codecs/wcd938x.c | 27 +++++---------------------- sound/soc/codecs/wcd938x.h | 4 +--- 3 files changed, 18 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/wcd938x-sdw.c b/sound/soc/codecs/wcd938x-sdw.c index d82c40ec6898..d1cabf300c36 100644 --- a/sound/soc/codecs/wcd938x-sdw.c +++ b/sound/soc/codecs/wcd938x-sdw.c @@ -176,8 +176,19 @@ static int wcd9380_interrupt_callback(struct sdw_slave *slave, struct sdw_slave_intr_status *status) { struct wcd938x_sdw_priv *wcd = dev_get_drvdata(&slave->dev); + struct irq_domain *slave_irq = wcd->slave_irq; + struct regmap *regmap = dev_get_regmap(&slave->dev, NULL); + u32 sts1, sts2, sts3; - return wcd938x_handle_sdw_irq(wcd); + do { + handle_nested_irq(irq_find_mapping(slave_irq, 0)); + regmap_read(regmap, WCD938X_DIGITAL_INTR_STATUS_0, &sts1); + regmap_read(regmap, WCD938X_DIGITAL_INTR_STATUS_1, &sts2); + regmap_read(regmap, WCD938X_DIGITAL_INTR_STATUS_2, &sts3); + + } while (sts1 || sts2 || sts3); + + return IRQ_HANDLED; } static struct sdw_slave_ops wcd9380_slave_ops = { @@ -239,16 +250,10 @@ static int wcd9380_probe(struct sdw_slave *pdev, SDW_SCP_INT1_PARITY; pdev->prop.lane_control_support = true; if (wcd->is_tx) { - struct regmap *rm; - pdev->prop.source_ports = GENMASK(WCD938X_MAX_SWR_PORTS, 0); pdev->prop.src_dpn_prop = wcd938x_dpn_prop; wcd->ch_info = &wcd938x_sdw_tx_ch_info[0]; pdev->prop.wake_capable = true; - - rm = devm_regmap_init_sdw(pdev, &wcd938x_regmap_config); - if (IS_ERR(rm)) - return PTR_ERR(rm); } else { pdev->prop.sink_ports = GENMASK(WCD938X_MAX_SWR_PORTS, 0); pdev->prop.sink_dpn_prop = wcd938x_dpn_prop; diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index a2c76dc8fd89..24afe93830ce 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -1198,7 +1198,7 @@ static bool wcd938x_volatile_register(struct device *dev, unsigned int reg) return false; } -struct regmap_config wcd938x_regmap_config = { +static struct regmap_config wcd938x_regmap_config = { .name = "wcd938x_csr", .reg_bits = 32, .val_bits = 8, @@ -1211,7 +1211,6 @@ struct regmap_config wcd938x_regmap_config = { .volatile_reg = wcd938x_volatile_register, .can_multi_write = true, }; -EXPORT_SYMBOL_GPL(wcd938x_regmap_config); static const struct regmap_irq wcd938x_irqs[WCD938X_NUM_IRQS] = { REGMAP_IRQ_REG(WCD938X_IRQ_MBHC_BUTTON_PRESS_DET, 0, 0x01), @@ -3472,24 +3471,6 @@ static int wcd938x_reset(struct wcd938x_priv *wcd938x) return 0; } -int wcd938x_handle_sdw_irq(struct wcd938x_sdw_priv *wcd) -{ - struct wcd938x_priv *wcd938x = wcd->wcd938x; - struct irq_domain *slave_irq = wcd938x->virq; - u32 sts1, sts2, sts3; - - do { - handle_nested_irq(irq_find_mapping(slave_irq, 0)); - regmap_read(wcd938x->regmap, WCD938X_DIGITAL_INTR_STATUS_0, &sts1); - regmap_read(wcd938x->regmap, WCD938X_DIGITAL_INTR_STATUS_1, &sts2); - regmap_read(wcd938x->regmap, WCD938X_DIGITAL_INTR_STATUS_2, &sts3); - - } while (sts1 || sts2 || sts3); - - return IRQ_HANDLED; -} -EXPORT_SYMBOL_GPL(wcd938x_handle_sdw_irq); - static int wcd938x_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -3573,6 +3554,7 @@ static int wcd938x_bind(struct device *dev) } wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev); wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x; + wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq; wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode); if (!wcd938x->txdev) { @@ -3581,6 +3563,7 @@ static int wcd938x_bind(struct device *dev) } wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev); wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x; + wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq; wcd938x->tx_sdw_dev = dev_to_sdw_dev(wcd938x->txdev); if (!wcd938x->tx_sdw_dev) { dev_err(dev, "could not get txslave with matching of dev\n"); @@ -3607,8 +3590,8 @@ static int wcd938x_bind(struct device *dev) return -EINVAL; } - wcd938x->regmap = dev_get_regmap(wcd938x->txdev, NULL); - if (!wcd938x->regmap) { + wcd938x->regmap = devm_regmap_init_sdw(wcd938x->tx_sdw_dev, &wcd938x_regmap_config); + if (IS_ERR(wcd938x->regmap)) { dev_err(dev, "%s: tx csr regmap not found\n", __func__); return PTR_ERR(wcd938x->regmap); } diff --git a/sound/soc/codecs/wcd938x.h b/sound/soc/codecs/wcd938x.h index 9db3ab6e47a6..07b08de4cebf 100644 --- a/sound/soc/codecs/wcd938x.h +++ b/sound/soc/codecs/wcd938x.h @@ -663,11 +663,9 @@ struct wcd938x_sdw_priv { int num_ports; bool is_tx; struct wcd938x_priv *wcd938x; + struct irq_domain *slave_irq; }; -extern struct regmap_config wcd938x_regmap_config; -int wcd938x_handle_sdw_irq(struct wcd938x_sdw_priv *priv); - #if IS_ENABLED(CONFIG_SND_SOC_WCD938X_SDW) int wcd938x_sdw_free(struct wcd938x_sdw_priv *wcd, struct snd_pcm_substream *substream, -- 2.21.0

2 2

[PATCH -next] ASoC: jz4740-i2s: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 15 Jun '21

15 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/jz4740/jz4740-i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 65d0bf939134..7ad5d9a924d8 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -525,8 +525,7 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) i2s->soc_info = device_get_match_data(dev); - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - i2s->base = devm_ioremap_resource(dev, mem); + i2s->base = devm_platform_get_and_ioremap_resource(pdev, 0, &mem); if (IS_ERR(i2s->base)) return PTR_ERR(i2s->base); -- 2.25.1

2 1

[PATCH 0/4] ASoC: qdsp6: Add Quinary MI2S ports support
by Gabriel David 15 Jun '21

15 Jun '21

Various devices(such as the Motorola Moto G7 Power, codename ocean) use the Quinary MI2S ports for reproducing audio. Add support to them in kernel. Signed-off-by: Gabriel David <ultracoolguy(a)disroot.org> Gabriel David (4): ASoC: q6afe: dt-bindings: Add QUIN_MI2S_RX/TX ASoC: qdsp6: q6afe: Add Quinary MI2S ports ASoC: qdsp6: q6afe-dai: Add Quinary MI2S ports ASoC: qdsp6: q6routing: Add Quinary MI2S ports include/dt-bindings/sound/qcom,q6afe.h | 2 ++ sound/soc/qcom/qdsp6/q6afe-dai.c | 41 ++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.c | 8 +++++ sound/soc/qcom/qdsp6/q6afe.h | 2 +- sound/soc/qcom/qdsp6/q6routing.c | 11 +++++++ 5 files changed, 63 insertions(+), 1 deletion(-) --- As a warning, I'm currently the only tester of these patches. If that's gonna be a problem then I understand. -- 2.31.1

3 6