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[alsa-devel] [PATCH 0/3] ASoC: remove duplicated definitions
by Kuninori Morimoto 24 Aug '17

24 Aug '17

Hi Mark, Lars-Peter This is not a big-deal, but I think these are duplicated definitions. These patches cleanup it. Kuninori Morimoto (3): ASoC: remove duplicate definition of controls/num_controls ASoC: remove duplicate definition of dapm_widgets/num_dapm_widgets ASoC: remove duplicate definition of dapm_routes/num_dapm_routes include/sound/soc.h | 6 ------ sound/soc/soc-core.c | 28 ++++++++++++---------------- 2 files changed, 12 insertions(+), 22 deletions(-) -- 1.9.1

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[alsa-devel] [asoc:topic/rt5514 12/12] sound/soc/codecs/rt5514.c:343:11: error: implicit declaration of function 'rt5514_spi_burst_write'
by kbuild test robot 24 Aug '17

24 Aug '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rt5514 head: d18420b0a0b8f8f51f55fc654b2e0eb86d8b41d7 commit: d18420b0a0b8f8f51f55fc654b2e0eb86d8b41d7 [12/12] ASoC: rt5514: expose Hotword Model control config: sparc64-allmodconfig (attached as .config) compiler: sparc64-linux-gnu-gcc (Debian 6.1.1-9) 6.1.1 20160705 reproduce: wget https://raw.githubusercontent.com/01org/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout d18420b0a0b8f8f51f55fc654b2e0eb86d8b41d7 # save the attached .config to linux build tree make.cross ARCH=sparc64 All errors (new ones prefixed by >>): sound/soc/codecs/rt5514.c: In function 'rt5514_dsp_voice_wake_up_put': >> sound/soc/codecs/rt5514.c:343:11: error: implicit declaration of function 'rt5514_spi_burst_write' [-Werror=implicit-function-declaration] ret = rt5514_spi_burst_write(0x4ff80000, ^~~~~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/rt5514_spi_burst_write +343 sound/soc/codecs/rt5514.c 297 298 static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, 299 struct snd_ctl_elem_value *ucontrol) 300 { 301 struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); 302 struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); 303 struct snd_soc_codec *codec = rt5514->codec; 304 const struct firmware *fw = NULL; 305 int ret = 0; 306 307 if (ucontrol->value.integer.value[0] == rt5514->dsp_enabled) 308 return 0; 309 310 if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { 311 rt5514->dsp_enabled = ucontrol->value.integer.value[0]; 312 313 if (rt5514->dsp_enabled) { 314 rt5514_enable_dsp_prepare(rt5514); 315 316 request_firmware(&fw, RT5514_FIRMWARE1, codec->dev); 317 if (fw) { 318 #if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) 319 rt5514_spi_burst_write(0x4ff60000, fw->data, 320 ((fw->size/8)+1)*8); 321 #else 322 dev_err(codec->dev, "There is no SPI driver for" 323 " loading the firmware\n"); 324 #endif 325 release_firmware(fw); 326 fw = NULL; 327 } 328 329 request_firmware(&fw, RT5514_FIRMWARE2, codec->dev); 330 if (fw) { 331 #if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) 332 rt5514_spi_burst_write(0x4ffc0000, fw->data, 333 ((fw->size/8)+1)*8); 334 #else 335 dev_err(codec->dev, "There is no SPI driver for" 336 " loading the firmware\n"); 337 #endif 338 release_firmware(fw); 339 fw = NULL; 340 } 341 342 if (rt5514->model_buf && rt5514->model_len) { > 343 ret = rt5514_spi_burst_write(0x4ff80000, 344 rt5514->model_buf, 345 ((rt5514->model_len / 8) + 1) * 8); 346 if (ret) { 347 dev_err(codec->dev, 348 "Model load failed %d\n", ret); 349 return ret; 350 } 351 } else { 352 request_firmware(&fw, RT5514_FIRMWARE3, 353 codec->dev); 354 if (fw) { 355 rt5514_spi_burst_write(0x4ff80000, 356 fw->data, 357 ((fw->size/8)+1)*8); 358 release_firmware(fw); 359 fw = NULL; 360 } 361 } 362 363 /* DSP run */ 364 regmap_write(rt5514->i2c_regmap, 0x18002f00, 365 0x00055148); 366 } else { 367 regmap_multi_reg_write(rt5514->i2c_regmap, 368 rt5514_i2c_patch, ARRAY_SIZE(rt5514_i2c_patch)); 369 regcache_mark_dirty(rt5514->regmap); 370 regcache_sync(rt5514->regmap); 371 } 372 } 373 374 return 0; 375 } 376 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: codecs: msm8916-wcd-analog: always true condition
by Dan Carpenter 24 Aug '17

24 Aug '17

Static checkers complain that unsigned int val is always >= 0. The context is that snd_soc_read() returns -1U on error. This is harmless because we're checking that CDC_A_MBHC_RESULT_1_BTN_RESULT_MASK is not set, and it will always be set for -1U. I could have just removed the check against -1 but I preserved it because I thought it helped with readability. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 4aa878d29098..549c269acc7d 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -949,7 +949,7 @@ static irqreturn_t mbhc_btn_release_irq_handler(int irq, void *arg) u32 val = snd_soc_read(codec, CDC_A_MBHC_RESULT_1); /* check if its BTN0 thats released */ - if ((val >= 0) && !(val & CDC_A_MBHC_RESULT_1_BTN_RESULT_MASK)) + if ((val != -1) && !(val & CDC_A_MBHC_RESULT_1_BTN_RESULT_MASK)) priv->mbhc_btn0_pressed = false; } else {

3 2

[alsa-devel] [PATCH] ASoC: Add a sanity check before using dai driver name
by Jeffy Chen 24 Aug '17

24 Aug '17

The dai driver's name is allowed to be NULL. So add a sanity check for that. Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> Reported-by: Donglin Peng <dolinux.peng(a)gmail.com> --- sound/soc/soc-core.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 77e7e2a11af0..8ab9ee2b460a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -978,11 +978,13 @@ struct snd_soc_dai *snd_soc_find_dai( if (dlc->name && strcmp(component->name, dlc->name)) continue; list_for_each_entry(dai, &component->dai_list, list) { - if (dlc->dai_name && strcmp(dai->name, dlc->dai_name) - && strcmp(dai->driver->name, dlc->dai_name)) - continue; - - return dai; + if (!dlc->dai_name) + return dai; + if (!strcmp(dai->name, dlc->dai_name)) + return dai; + if (dai->driver->name && + !strcmp(dai->driver->name, dlc->dai_name)) + return dai; } } -- 2.11.0

3 6

[alsa-devel] [PATCH][alsa-lib] test: apply optimization for v4.14 kernel about changes for TLV data handling on user-defined element set
by Takashi Sakamoto 24 Aug '17

24 Aug '17

At kernel v4.14, in initial state, elements on user-defined sets have write-only flag. When applications write TLV data, then the elements get readable flag and the TLV data is available. Originally TLV data is shared by elements in the same set, thus events are generated for all of elements in the set by the write operation. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- test/user-ctl-element-set.c | 84 +++++++++++++++++++++++++++++++++++++++------ 1 file changed, 73 insertions(+), 11 deletions(-) diff --git a/test/user-ctl-element-set.c b/test/user-ctl-element-set.c index 8635c156..f3710732 100644 --- a/test/user-ctl-element-set.c +++ b/test/user-ctl-element-set.c @@ -9,6 +9,7 @@ #include "../include/asoundlib.h" #include <sound/tlv.h> +#include <stdbool.h> struct elem_set_trial { snd_ctl_t *handle; @@ -28,6 +29,8 @@ struct elem_set_trial { snd_ctl_elem_value_t *elem_data); int (*allocate_elem_set_tlv)(struct elem_set_trial *trial, unsigned int **tlv); + + bool tlv_readable; }; struct chmap_entry { @@ -420,9 +423,9 @@ static int check_event(struct elem_set_trial *trial, unsigned int mask, continue; /* * I expect each event is generated separately to the same - * element. + * element or several events are generated at once. */ - if (!(snd_ctl_event_elem_get_mask(event) & mask)) + if ((snd_ctl_event_elem_get_mask(event) & mask) != mask) continue; --expected_count; } @@ -561,6 +564,16 @@ static int check_elem_set_props(struct elem_set_trial *trial) if (!snd_ctl_elem_info_is_locked(info)) return -EIO; + /* + * In initial state, any application can register TLV data for + * user-defined element set except for IEC 958 type, thus + * elements in any user-defined set should allow any write + * operation. + */ + if (trial->type != SND_CTL_ELEM_TYPE_IEC958 && + !snd_ctl_elem_info_is_tlv_writable(info)) + return -EIO; + /* Check type-specific properties. */ if (trial->check_elem_props != NULL) { err = trial->check_elem_props(trial, info); @@ -572,6 +585,25 @@ static int check_elem_set_props(struct elem_set_trial *trial) err = snd_ctl_elem_unlock(trial->handle, id); if (err < 0) return err; + + /* + * Till kernel v4.14, ALSA control core allows elements in any + * user-defined set to have TLV_READ flag even if they have no + * TLV data in their initial state. In this case, any read + * operation for TLV data should return -ENXIO. + */ + if (snd_ctl_elem_info_is_tlv_readable(info)) { + unsigned int data[32]; + err = snd_ctl_elem_tlv_read(trial->handle, trial->id, + data, sizeof(data)); + if (err >= 0) + return -EIO; + if (err != -ENXIO) + return err; + + trial->tlv_readable = true; + } + } return 0; @@ -619,6 +651,8 @@ static int check_elems(struct elem_set_trial *trial) static int check_tlv(struct elem_set_trial *trial) { unsigned int *tlv; + int mask; + unsigned int count; unsigned int len; unsigned int *curr; int err; @@ -644,6 +678,36 @@ static int check_tlv(struct elem_set_trial *trial) if (err < 0) goto end; + /* + * Since kernel v4.14, any write operation to an element in user-defined + * set can change state of the other elements in the same set. In this + * case, any TLV data is firstly available after the operation. + */ + if (!trial->tlv_readable) { + mask = SND_CTL_EVENT_MASK_INFO | SND_CTL_EVENT_MASK_TLV; + count = trial->element_count; + } else { + mask = SND_CTL_EVENT_MASK_TLV; + count = 1; + } + err = check_event(trial, mask, count); + if (err < 0) + goto end; + if (!trial->tlv_readable) { + snd_ctl_elem_info_t *info; + snd_ctl_elem_info_alloca(&info); + + snd_ctl_elem_info_set_id(info, trial->id); + err = snd_ctl_elem_info(trial->handle, info); + if (err < 0) + return err; + if (!snd_ctl_elem_info_is_tlv_readable(info)) + return -EIO; + + /* Now TLV data is available for this element set. */ + trial->tlv_readable = true; + } + err = snd_ctl_elem_tlv_read(trial->handle, trial->id, curr, len); if (err < 0) goto end; @@ -690,6 +754,7 @@ int main(void) trial.change_elem_members = change_bool_elem_members; trial.allocate_elem_set_tlv = allocate_bool_elem_set_tlv; + trial.tlv_readable = false; break; case SND_CTL_ELEM_TYPE_INTEGER: trial.element_count = 900; @@ -703,6 +768,7 @@ int main(void) trial.change_elem_members = change_int_elem_members; trial.allocate_elem_set_tlv = allocate_int_elem_set_tlv; + trial.tlv_readable = false; break; case SND_CTL_ELEM_TYPE_ENUMERATED: trial.element_count = 900; @@ -715,6 +781,7 @@ int main(void) trial.check_elem_props = check_enum_elem_props; trial.change_elem_members = change_enum_elem_members; trial.allocate_elem_set_tlv = NULL; + trial.tlv_readable = false; break; case SND_CTL_ELEM_TYPE_BYTES: trial.element_count = 900; @@ -728,6 +795,7 @@ int main(void) trial.change_elem_members = change_bytes_elem_members; trial.allocate_elem_set_tlv = allocate_bytes_elem_set_tlv; + trial.tlv_readable = false; break; case SND_CTL_ELEM_TYPE_IEC958: trial.element_count = 1; @@ -740,6 +808,7 @@ int main(void) trial.check_elem_props = NULL; trial.change_elem_members = change_iec958_elem_members; trial.allocate_elem_set_tlv = NULL; + trial.tlv_readable = false; break; case SND_CTL_ELEM_TYPE_INTEGER64: default: @@ -754,6 +823,7 @@ int main(void) trial.change_elem_members = change_int64_elem_members; trial.allocate_elem_set_tlv = allocate_int64_elem_set_tlv; + trial.tlv_readable = false; break; } @@ -784,7 +854,7 @@ int main(void) /* Check properties of each element in this element set. */ err = check_elem_set_props(&trial); if (err < 0) { - printf("Fail to check propetries of each element with " + printf("Fail to check properties of each element with " "%s type.\n", snd_ctl_elem_type_name(trial.type)); break; @@ -822,14 +892,6 @@ int main(void) snd_ctl_elem_type_name(trial.type)); break; } - err = check_event(&trial, SND_CTL_EVENT_MASK_TLV, 1); - if (err < 0) { - printf("Fail to check an event to change TLV" - "data of an an element set with %s " - "type.\n", - snd_ctl_elem_type_name(trial.type)); - break; - } } /* Test an operation to remove elements in this element set. */ -- 2.11.0

2 1

[alsa-devel] [PATCH 0/3] ALSA: control: delegate TLV eventing to each driver
by Takashi Sakamoto 24 Aug '17

24 Aug '17

Hi, This patchse is the same one which I posted and for merge to upstream. [alsa-devel] [RFC][PATCH 0/3] ALSA: control: delegate TLV eventing to each driver http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/124382.html Nothing changed from the original one. Takashi Sakamoto (3): ALSA: control: delegate TLV eventing to each driver ALSA: control: queue TLV event for a set of user-defined element ALSA: control: TLV data is unavailable at initial state of user-defined element set sound/core/control.c | 34 ++++++++++++++++++++++------------ 1 file changed, 22 insertions(+), 12 deletions(-) -- 2.11.0

2 4

[alsa-devel] [PATCH] ALSA: nm256: constify snd_ac97_res_table
by Arvind Yadav 24 Aug '17

24 Aug '17

snd_ac97_res_table are not supposed to change at runtime. All functions working with snd_ac97_res_table provided by <sound/ac97_codec.h> work with const snd_ac97_res_table. So mark the non-const structs as const. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/pci/nm256/nm256.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 0ef8054..b97f4ea 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1271,7 +1271,7 @@ static int nm256_ac97_idx(unsigned short reg) } /* static resolution table */ -static struct snd_ac97_res_table nm256_res_table[] = { +static const struct snd_ac97_res_table nm256_res_table[] = { { AC97_MASTER, 0x1f1f }, { AC97_HEADPHONE, 0x1f1f }, { AC97_MASTER_MONO, 0x001f }, -- 1.9.1

2 1

[alsa-devel] [PATCH] sound/oss/pss: fix spelling mistake: "compoents" -> "components"
by Colin King 24 Aug '17

24 Aug '17

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in MODULE_PARM_DESC text Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/oss/pss.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/pss.c b/sound/oss/pss.c index 33c3a442e162..951b14dec73a 100644 --- a/sound/oss/pss.c +++ b/sound/oss/pss.c @@ -1156,7 +1156,7 @@ MODULE_PARM_DESC(pss_cdrom_port, "Set the PSS CDROM port i/o base (0x340 or othe module_param(pss_enable_joystick, bool, 0); MODULE_PARM_DESC(pss_enable_joystick, "Enables the PSS joystick port (1 to enable, 0 to disable)"); module_param(pss_no_sound, bool, 0); -MODULE_PARM_DESC(pss_no_sound, "Configure sound compoents (0 - no, 1 - yes)"); +MODULE_PARM_DESC(pss_no_sound, "Configure sound components (0 - no, 1 - yes)"); module_param(pss_keep_settings, bool, 0); MODULE_PARM_DESC(pss_keep_settings, "Keep hardware setting at driver unloading (0 - no, 1 - yes)"); module_param(pss_firmware, charp, 0); -- 2.14.1

2 1

[alsa-devel] [RFC][PATCH 0/3] ALSA: control: delegate TLV eventing to each driver
by Takashi Sakamoto 24 Aug '17

24 Aug '17

Hi, This patchset is to avoid a trouble on ALSA control core, kernel API of ALSA control component and ALSA control interface/protocol to handle TLV operations from user space. In a design of ALSA control core, a set of elements is represented by 'struct snd_kcontrol' to share common attributes. The set of elements shares TLV (Type-Length-Value) data, too. On the other hand, in ALSA control interface/protocol for applications, a TLV operation is committed to an element. Totally, the operation can have sub-effect to the other elements in the set. For example, TLV_WRITE operation is expected to change TLV data, which returns to applications. Applications attempt to change the TLV data per element, but in the above design, they can effect to elements in the same set. As a default, ALSA control core has no implementation except for TLV_READ operation. Thus, the above design looks to have no issue. However, in kernel APIs of ALSA control component, developers can program a handler for any request of the TLV operation. Therefore, for elements in a set which has the handler, applications can commit TLV_WRITE and TLV_COMMAND requests. For the above scenario, ALSA control core assist notification. When the handler returns positive value, the core queueing an event for a requested element. However, this includes design defects that the event is not queued for the other element in a set. Actually, developers can program the handlers to keep per-element TLV data, but it depends on each driver. In this patchset, I attempts to solve the above issue, by delegating the event notification to each driver of ALSA control component. Additionally, this patchset includes a solution against an issue in below message thread: [alsa-devel] snd_ctl_add_enumerated_elem_set and TLV http://mailman.alsa-project.org/pipermail/alsa-devel/2017-July/123364.html Takashi Sakamoto (3): ALSA: control: delegate TLV eventing to each driver ALSA: control: queue TLV event for all of elements in an user-defined set ALSA: control: disable TLV data at initial state of user-defined element set sound/core/control.c | 34 ++++++++++++++++++++++------------ 1 file changed, 22 insertions(+), 12 deletions(-) -- 2.11.0

2 5

[alsa-devel] [PATCH] ASoC: soc-ac97: replace codec to component
by Kuninori Morimoto 24 Aug '17

24 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> codec will be replaced to component soon. AC97 is codec related feature now. Thus this patch replaces codec to component on AC97. To keep compatibility, previous snd_soc_xxx_ac97_codec() is now defined as macro to convert snd_soc_xxx_ac97_component(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- - I have no AC97 board. Compile test only had done. include/sound/soc.h | 10 +++++-- sound/soc/soc-ac97.c | 84 +++++++++++++++++++++++++++------------------------- 2 files changed, 51 insertions(+), 43 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 180c41b..f8fa9d0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -568,10 +568,14 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value); #ifdef CONFIG_SND_SOC_AC97_BUS -struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec); -struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec, +#define snd_soc_alloc_ac97_codec(codec) snd_soc_alloc_ac97_component(&codec->component) +#define snd_soc_new_ac97_codec(codec, id, id_mask) snd_soc_new_ac97_component(&codec->component, id, id_mask) +#define snd_soc_free_ac97_codec(ac97) snd_soc_free_ac97_component(ac97) + +struct snd_ac97 *snd_soc_alloc_ac97_component(struct snd_soc_component *component); +struct snd_ac97 *snd_soc_new_ac97_component(struct snd_soc_component *component, unsigned int id, unsigned int id_mask); -void snd_soc_free_ac97_codec(struct snd_ac97 *ac97); +void snd_soc_free_ac97_component(struct snd_ac97 *ac97); int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 36dae41..3f424f2 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -44,7 +44,7 @@ struct snd_ac97_gpio_priv { struct gpio_chip gpio_chip; #endif unsigned int gpios_set; - struct snd_soc_codec *codec; + struct snd_soc_component *component; }; static struct snd_ac97_bus soc_ac97_bus = { @@ -57,11 +57,11 @@ static void soc_ac97_device_release(struct device *dev) } #ifdef CONFIG_GPIOLIB -static inline struct snd_soc_codec *gpio_to_codec(struct gpio_chip *chip) +static inline struct snd_soc_component *gpio_to_component(struct gpio_chip *chip) { struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip); - return gpio_priv->codec; + return gpio_priv->component; } static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned offset) @@ -75,20 +75,22 @@ static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned offset) static int snd_soc_ac97_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { - struct snd_soc_codec *codec = gpio_to_codec(chip); + struct snd_soc_component *component = gpio_to_component(chip); - dev_dbg(codec->dev, "set gpio %d to output\n", offset); - return snd_soc_update_bits(codec, AC97_GPIO_CFG, + dev_dbg(component->dev, "set gpio %d to output\n", offset); + return snd_soc_component_update_bits(component, AC97_GPIO_CFG, 1 << offset, 1 << offset); } static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) { - struct snd_soc_codec *codec = gpio_to_codec(chip); + struct snd_soc_component *component = gpio_to_component(chip); int ret; - ret = snd_soc_read(codec, AC97_GPIO_STATUS); - dev_dbg(codec->dev, "get gpio %d : %d\n", offset, + if (snd_soc_component_read(component, AC97_GPIO_STATUS, &ret) < 0) + ret = -1; + + dev_dbg(component->dev, "get gpio %d : %d\n", offset, ret < 0 ? ret : ret & (1 << offset)); return ret < 0 ? ret : !!(ret & (1 << offset)); @@ -98,22 +100,24 @@ static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip); - struct snd_soc_codec *codec = gpio_to_codec(chip); + struct snd_soc_component *component = gpio_to_component(chip); gpio_priv->gpios_set &= ~(1 << offset); gpio_priv->gpios_set |= (!!value) << offset; - snd_soc_write(codec, AC97_GPIO_STATUS, gpio_priv->gpios_set); - dev_dbg(codec->dev, "set gpio %d to %d\n", offset, !!value); + snd_soc_component_write(component, AC97_GPIO_STATUS, + gpio_priv->gpios_set); + dev_dbg(component->dev, "set gpio %d to %d\n", offset, !!value); } static int snd_soc_ac97_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct snd_soc_codec *codec = gpio_to_codec(chip); + struct snd_soc_component *component = gpio_to_component(chip); - dev_dbg(codec->dev, "set gpio %d to output\n", offset); + dev_dbg(component->dev, "set gpio %d to output\n", offset); snd_soc_ac97_gpio_set(chip, offset, value); - return snd_soc_update_bits(codec, AC97_GPIO_CFG, 1 << offset, 0); + return snd_soc_component_update_bits(component, AC97_GPIO_CFG, + 1 << offset, 0); } static const struct gpio_chip snd_soc_ac97_gpio_chip = { @@ -128,24 +132,24 @@ static int snd_soc_ac97_gpio_direction_out(struct gpio_chip *chip, }; static int snd_soc_ac97_init_gpio(struct snd_ac97 *ac97, - struct snd_soc_codec *codec) + struct snd_soc_component *component) { struct snd_ac97_gpio_priv *gpio_priv; int ret; - gpio_priv = devm_kzalloc(codec->dev, sizeof(*gpio_priv), GFP_KERNEL); + gpio_priv = devm_kzalloc(component->dev, sizeof(*gpio_priv), GFP_KERNEL); if (!gpio_priv) return -ENOMEM; ac97->gpio_priv = gpio_priv; - gpio_priv->codec = codec; + gpio_priv->component = component; gpio_priv->gpio_chip = snd_soc_ac97_gpio_chip; gpio_priv->gpio_chip.ngpio = AC97_NUM_GPIOS; - gpio_priv->gpio_chip.parent = codec->dev; + gpio_priv->gpio_chip.parent = component->dev; gpio_priv->gpio_chip.base = -1; ret = gpiochip_add_data(&gpio_priv->gpio_chip, gpio_priv); if (ret != 0) - dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); + dev_err(component->dev, "Failed to add GPIOs: %d\n", ret); return ret; } @@ -155,7 +159,7 @@ static void snd_soc_ac97_free_gpio(struct snd_ac97 *ac97) } #else static int snd_soc_ac97_init_gpio(struct snd_ac97 *ac97, - struct snd_soc_codec *codec) + struct snd_soc_component *component) { return 0; } @@ -166,8 +170,8 @@ static void snd_soc_ac97_free_gpio(struct snd_ac97 *ac97) #endif /** - * snd_soc_alloc_ac97_codec() - Allocate new a AC'97 device - * @codec: The CODEC for which to create the AC'97 device + * snd_soc_alloc_ac97_component() - Allocate new a AC'97 device + * @component: The COMPONENT for which to create the AC'97 device * * Allocated a new snd_ac97 device and intializes it, but does not yet register * it. The caller is responsible to either call device_add(&ac97->dev) to @@ -175,7 +179,7 @@ static void snd_soc_ac97_free_gpio(struct snd_ac97 *ac97) * * Returns: A snd_ac97 device or a PTR_ERR in case of an error. */ -struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec) +struct snd_ac97 *snd_soc_alloc_ac97_component(struct snd_soc_component *component) { struct snd_ac97 *ac97; @@ -187,26 +191,26 @@ struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec) ac97->num = 0; ac97->dev.bus = &ac97_bus_type; - ac97->dev.parent = codec->component.card->dev; + ac97->dev.parent = component->card->dev; ac97->dev.release = soc_ac97_device_release; dev_set_name(&ac97->dev, "%d-%d:%s", - codec->component.card->snd_card->number, 0, - codec->component.name); + component->card->snd_card->number, 0, + component->name); device_initialize(&ac97->dev); return ac97; } -EXPORT_SYMBOL(snd_soc_alloc_ac97_codec); +EXPORT_SYMBOL(snd_soc_alloc_ac97_component); /** - * snd_soc_new_ac97_codec - initailise AC97 device - * @codec: audio codec + * snd_soc_new_ac97_component - initailise AC97 device + * @component: audio component * @id: The expected device ID * @id_mask: Mask that is applied to the device ID before comparing with @id * - * Initialises AC97 codec resources for use by ad-hoc devices only. + * Initialises AC97 component resources for use by ad-hoc devices only. * * If @id is not 0 this function will reset the device, then read the ID from * the device and check if it matches the expected ID. If it doesn't match an @@ -214,20 +218,20 @@ struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec) * * Returns: A PTR_ERR() on failure or a valid snd_ac97 struct on success. */ -struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec, +struct snd_ac97 *snd_soc_new_ac97_component(struct snd_soc_component *component, unsigned int id, unsigned int id_mask) { struct snd_ac97 *ac97; int ret; - ac97 = snd_soc_alloc_ac97_codec(codec); + ac97 = snd_soc_alloc_ac97_component(component); if (IS_ERR(ac97)) return ac97; if (id) { ret = snd_ac97_reset(ac97, false, id, id_mask); if (ret < 0) { - dev_err(codec->dev, "Failed to reset AC97 device: %d\n", + dev_err(component->dev, "Failed to reset AC97 device: %d\n", ret); goto err_put_device; } @@ -237,7 +241,7 @@ struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec, if (ret) goto err_put_device; - ret = snd_soc_ac97_init_gpio(ac97, codec); + ret = snd_soc_ac97_init_gpio(ac97, component); if (ret) goto err_put_device; @@ -247,22 +251,22 @@ struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec, put_device(&ac97->dev); return ERR_PTR(ret); } -EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); +EXPORT_SYMBOL_GPL(snd_soc_new_ac97_component); /** - * snd_soc_free_ac97_codec - free AC97 codec device + * snd_soc_free_ac97_component - free AC97 component device * @ac97: snd_ac97 device to be freed * - * Frees AC97 codec device resources. + * Frees AC97 component device resources. */ -void snd_soc_free_ac97_codec(struct snd_ac97 *ac97) +void snd_soc_free_ac97_component(struct snd_ac97 *ac97) { snd_soc_ac97_free_gpio(ac97); device_del(&ac97->dev); ac97->bus = NULL; put_device(&ac97->dev); } -EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); +EXPORT_SYMBOL_GPL(snd_soc_free_ac97_component); static struct snd_ac97_reset_cfg snd_ac97_rst_cfg; -- 1.9.1

2 2

[alsa-devel] [PATCH 00/11] ASoC: make snd_soc_platform_driver const
by Bhumika Goyal 23 Aug '17

23 Aug '17

Make snd_soc_platform_driver const. Bhumika Goyal (11): ASoC: codecs: make snd_soc_platform_driver const ASoC: Intel: make snd_soc_platform_driver const ASoC: fsl: make snd_soc_platform_driver const ASoC: nuc900: make snd_soc_platform_driver const ASoC: omap: make snd_soc_platform_driver const ASoC: pxa: make snd_soc_platform_driver const ASoC: qcom: make snd_soc_platform_driver const ASoC: samsung: make snd_soc_platform_driver const ASoC: sh: make snd_soc_platform_driver const ASoC: txx9: make snd_soc_platform_driver const ASoC: soc-utils: make snd_soc_platform_driver const sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/rt5514-spi.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/fsl/imx-pcm-fiq.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 2 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 2 +- sound/soc/nuc900/nuc900-pcm.c | 2 +- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 2 +- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/samsung/idma.c | 2 +- sound/soc/sh/dma-sh7760.c | 2 +- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/soc-utils.c | 2 +- sound/soc/txx9/txx9aclc.c | 2 +- 21 files changed, 21 insertions(+), 21 deletions(-) -- 1.9.1

8 18

[alsa-devel] [ANNOUNCE RESENT] Audio Mini Summit 2017 at Prague, Oct 27
by Takashi Iwai 23 Aug '17

23 Aug '17

Hi, this is a gentle reminder for people who missed the previous announcement for Audio Mini Summit. Below is the content I posted in a month ago. If anyone is interested, please join up! thanks, Takashi === Hi, this is an announcement of Audio Mini Summit 2017. It's an annual meeting we regularly have along with major Linux conferences, and this year, we'll hold the meeting at Prague (Czech Republic) on Friday, October 27, right after GStreamer Summit, ELCE and Open Source Summit Europe. We planned a whole-day meeting, and will likely have an evening dinner event, too. Please note that the meeting will be held in a conference room of SUSE Prague office, not in Hilton hotel where ELCE/OSSE is held. It's located near Hilton, just a couple of blocks away. The address of SUSE office is: Corso II, Křižíkova 148/34, 18600 Praha 8 https://goo.gl/maps/anET1yh3XXu Also, since the conference room is hosted by SUSE, it's an open event at this time; that is, you can attend only this meeting without registration for other Linux Foundation events in that week, too. If you are interested in the event, please fill up the registration form in Google doc spreadsheet below: https://docs.google.com/spreadsheets/d/1a-RaYtTVVtgh_cyRN5GwATiixppGCMEwqw6… Just add yourself in the attendee list from the web page. And, now we are gathering the topics to discuss, too. If you have any interesting topics to be discussed in the meeting, feel free to post to alsa-devel ML. I'm going to write up later. thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: aaci: constify amba_id
by Arvind Yadav 23 Aug '17

23 Aug '17

amba_id are not supposed to change at runtime. All functions working with const amba_id. So mark the non-const structs as const. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/arm/aaci.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 4140b1b..05562ba 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -1085,7 +1085,7 @@ static int aaci_remove(struct amba_device *dev) return 0; } -static struct amba_id aaci_ids[] = { +static const struct amba_id aaci_ids[] = { { .id = 0x00041041, .mask = 0x000fffff, -- 2.7.4

1 0

[alsa-devel] [GIT PULL]: intel audio firmware updates
by Vinod Koul 23 Aug '17

23 Aug '17

Hi, Please pull to receive updated KBL audio firmware. Along with this time for some cleaning of the exiting files and remove no longer used SKL and BXT fw files. The following changes since commit 7d2c913dcd1be083350d97a8cb1eba24cfacbc8a: ath10k: update year in license (2017-06-22 12:06:02 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git intel for you to fetch changes up to 7b7513e03eb6949f39fba2eb4b661c31f8ea6f5b: linux-firmware: intel: remove old broxton firmware (2017-08-21 10:35:32 +0530) ---------------------------------------------------------------- Subhransu S. Prusty (1): linux-firmware: intel: Update Kabylake audio firmware Vinod Koul (2): linux-firmware: intel: remove old skylake firmware linux-firmware: intel: remove old broxton firmware WHENCE | 26 +++----------------------- intel/dsp_fw_bxtn_v1118.bin | Bin 505608 -> 0 bytes intel/dsp_fw_bxtn_v2010.bin | Bin 505608 -> 0 bytes intel/dsp_fw_bxtn_v430.bin | Bin 505608 -> 0 bytes intel/dsp_fw_bxtn_v702.bin | Bin 505608 -> 0 bytes intel/dsp_fw_kbl.bin | 2 +- intel/dsp_fw_kbl_v2630.bin | Bin 0 -> 243016 bytes intel/dsp_fw_release_v827.bin | Bin 233472 -> 0 bytes intel/dsp_fw_release_v869.bin | Bin 233472 -> 0 bytes intel/dsp_fw_release_v896.bin | Bin 233472 -> 0 bytes intel/dsp_fw_release_v927.bin | Bin 233472 -> 0 bytes intel/dsp_fw_release_v948.bin | Bin 221184 -> 0 bytes intel/dsp_fw_release_v951.bin | Bin 221184 -> 0 bytes intel/dsp_fw_release_v958.bin | Bin 221184 -> 0 bytes 14 files changed, 4 insertions(+), 24 deletions(-) delete mode 100644 intel/dsp_fw_bxtn_v1118.bin delete mode 100644 intel/dsp_fw_bxtn_v2010.bin delete mode 100644 intel/dsp_fw_bxtn_v430.bin delete mode 100644 intel/dsp_fw_bxtn_v702.bin create mode 100644 intel/dsp_fw_kbl_v2630.bin delete mode 100644 intel/dsp_fw_release_v827.bin delete mode 100644 intel/dsp_fw_release_v869.bin delete mode 100644 intel/dsp_fw_release_v896.bin delete mode 100644 intel/dsp_fw_release_v927.bin delete mode 100644 intel/dsp_fw_release_v948.bin delete mode 100644 intel/dsp_fw_release_v951.bin delete mode 100644 intel/dsp_fw_release_v958.bin Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH] ASoC: rt5645: make rt5645_platform_data const
by Bhumika Goyal 23 Aug '17

23 Aug '17

Make these const as they are only used during a copy operation. Done using Coccinelle. @match disable optional_qualifier@ identifier s; @@ static struct rt5645_platform_data s = {...}; @ref@ position p; identifier match.s; @@ s@p @good1@ position ref.p; identifier match.s,f,c; expression e; @@ ( e = s@p | e = s(a)p.f | c(...,s@p.f,...) | c(...,s@p,...) ) @bad depends on !good1@ position ref.p; identifier match.s; @@ s@p @depends on forall !bad disable optional_qualifier@ identifier match.s; @@ static + const struct rt5645_platform_data s; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/codecs/rt5645.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index f14f648..efec452 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3559,7 +3559,7 @@ static int rt5645_resume(struct snd_soc_codec *codec) MODULE_DEVICE_TABLE(acpi, rt5645_acpi_match); #endif -static struct rt5645_platform_data general_platform_data = { +static const struct rt5645_platform_data general_platform_data = { .dmic1_data_pin = RT5645_DMIC1_DISABLE, .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, .jd_mode = 3, @@ -3593,7 +3593,7 @@ static int rt5645_resume(struct snd_soc_codec *codec) { } }; -static struct rt5645_platform_data buddy_platform_data = { +static const struct rt5645_platform_data buddy_platform_data = { .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, .jd_mode = 3, @@ -3610,7 +3610,7 @@ static int rt5645_resume(struct snd_soc_codec *codec) { } }; -static struct rt5645_platform_data gpd_win_platform_data = { +static const struct rt5645_platform_data gpd_win_platform_data = { .jd_mode = 3, .inv_jd1_1 = true, }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: tegra: Remove superfluous 'machine' variable
by Arnd Bergmann 23 Aug '17

23 Aug '17

The last cleanup left behind an unused variable: sound/soc/tegra/tegra_wm8903.c: In function 'tegra_wm8903_remove': sound/soc/tegra/tegra_wm8903.c:206:23: error: unused variable 'machine' [-Werror=unused-variable] This removes it as welremoves it as welll Fixes: 2a8bd83af30e ("ASoC: tegra: Remove superfluous snd_soc_jack_free_gpios() call") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/tegra/tegra_wm8903.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 2384cb9cb95c..18bdae59a4df 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -203,7 +203,6 @@ static int tegra_wm8903_remove(struct snd_soc_card *card) snd_soc_get_pcm_runtime(card, card->dai_link[0].name); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); wm8903_mic_detect(codec, NULL, 0, 0); -- 2.9.0

2 2

[alsa-devel] [PATCH] ALSA: ctxfi: make hw structures const
by Bhumika Goyal 23 Aug '17

23 Aug '17

Make these const as they are only used in a copy operation. Done using Coccinelle: @match disable optional_qualifier@ identifier s; @@ static struct hw s = {...}; @ref@ position p; identifier match.s; @@ s@p @good1@ position ref.p; identifier match.s,f,c; expression e; @@ ( e = s@p | e = s(a)p.f | c(...,s@p.f,...) | c(...,s@p,...) ) @bad depends on !good1@ position ref.p; identifier match.s; @@ s@p @depends on forall !bad disable optional_qualifier@ identifier match.s; @@ static + const struct hw s; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/pci/ctxfi/cthw20k1.c | 2 +- sound/pci/ctxfi/cthw20k2.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 79edd88..8e6eb9d 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -2154,7 +2154,7 @@ static void hw_write_pci(struct hw *hw, u32 reg, u32 data) &container_of(hw, struct hw20k1, hw)->reg_pci_lock, flags); } -static struct hw ct20k1_preset = { +static const struct hw ct20k1_preset = { .irq = -1, .card_init = hw_card_init, diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 18ee776..b866d6b 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -2220,7 +2220,7 @@ static void hw_write_20kx(struct hw *hw, u32 reg, u32 data) writel(data, hw->mem_base + reg); } -static struct hw ct20k2_preset = { +static const struct hw ct20k2_preset = { .irq = -1, .card_init = hw_card_init, -- 1.9.1

2 1

[alsa-devel] [PATCH 0/4] constify ac97_pcm structures
by Arvind Yadav 23 Aug '17

23 Aug '17

ac97_pcm are not supposed to change at runtime. All functions working with ac97_pcm provided by <sound/ac97_codec.h> work with const ac97_pcm. So mark the non-const structs as const. Arvind Yadav (4): [PATCH 1/4] ALSA: aaci: constify ac97_pcm structures [PATCH 2/4] ALSA: ac97c: constify ac97_pcm structures [PATCH 3/4] ALSA: atiixp: constify ac97_pcm structures [PATCH 4/4] ALSA: intel8x0: constify ac97_pcm structures sound/arm/aaci.c | 2 +- sound/atmel/ac97c.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/intel8x0.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH] ALSA: Fireface: Use common error handling code in pcm_open()
by SF Markus Elfring 23 Aug '17

23 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 22 Aug 2017 21:45:42 +0200 Add a jump target so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/firewire/fireface/ff-pcm.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index d12a0e3a4219..40975395a64a 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -138,16 +138,12 @@ static int pcm_open(struct snd_pcm_substream *substream) return err; err = pcm_init_hw_params(ff, substream); - if (err < 0) { - snd_ff_stream_lock_release(ff); - return err; - } + if (err < 0) + goto release_lock; err = ff->spec->protocol->get_clock(ff, &rate, &src); - if (err < 0) { - snd_ff_stream_lock_release(ff); - return err; - } + if (err < 0) + goto release_lock; if (src != SND_FF_CLOCK_SRC_INTERNAL) { for (i = 0; i < CIP_SFC_COUNT; ++i) { @@ -177,6 +173,10 @@ static int pcm_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); return 0; + +release_lock: + snd_ff_stream_lock_release(ff); + return err; } static int pcm_close(struct snd_pcm_substream *substream) -- 2.14.0

3 4

[alsa-devel] Fixed latency between multiple channels
by cuhl＠eit.uni-kl.de 23 Aug '17

23 Aug '17

Hello Alsa-Developers, currently I try to decide, if ALSA is useable for a certain multi-channel signal processing task. I've read, that even with a realtime kernel the input-output latency varies with time - especially if USB is inbetween. Even so this is not what I actually want, I think the application might be sufficiently robust against this. However I expect to require a fixed latency between multiple ( I2S ) channels. This shall include stereo, but extend to all channels ( between 6 to 12 mono / 3 to 6 srereo channels probably ). So every channel's signal shall be exactly determined to its relative position with respect to the other channels, e. g. it shall be known inside the system which events occured simultanously. Likewise, the processed or generated signals shall be transferred to the outputs in a way, that they occur there with intended and modifiable relative latency to each other. Is this ensured by the ALSA architecture or is it at least possible to ensure it, if it is used? Any comments or answers are appreciated. Kind regards Christian

2 1

[alsa-devel] [asoc:topic/intel 52/54] sound/soc/intel/skylake/skl-messages.c:320:9: warning: 'ret' may be used uninitialized in this function
by kbuild test robot 23 Aug '17

23 Aug '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 7b992c24de78a206b3abd07192686d2d5db5012c commit: f77d443c4c299aff5ad9c74811dd063f4d8bebcf [52/54] ASoC: Intel: Skylake: Fix to free resources for dsp_init failure config: x86_64-randconfig-ne0-08232016 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout f77d443c4c299aff5ad9c74811dd063f4d8bebcf # save the attached .config to linux build tree make ARCH=x86_64 Note: it may well be a FALSE warning. FWIW you are at least aware of it now. http://gcc.gnu.org/wiki/Better_Uninitialized_Warnings All warnings (new ones prefixed by >>): sound/soc/intel/skylake/skl-messages.c: In function 'skl_init_dsp': >> sound/soc/intel/skylake/skl-messages.c:320:9: warning: 'ret' may be used uninitialized in this function [-Wmaybe-uninitialized] return ret; ^~~ vim +/ret +320 sound/soc/intel/skylake/skl-messages.c bc23ca35c Jeeja KP 2016-03-11 256 d255b0955 Jeeja KP 2015-07-21 257 int skl_init_dsp(struct skl *skl) d255b0955 Jeeja KP 2015-07-21 258 { d255b0955 Jeeja KP 2015-07-21 259 void __iomem *mmio_base; d255b0955 Jeeja KP 2015-07-21 260 struct hdac_ext_bus *ebus = &skl->ebus; d255b0955 Jeeja KP 2015-07-21 261 struct hdac_bus *bus = ebus_to_hbus(ebus); d255b0955 Jeeja KP 2015-07-21 262 struct skl_dsp_loader_ops loader_ops; bc23ca35c Jeeja KP 2016-03-11 263 int irq = bus->irq; 73a675816 Vinod Koul 2016-07-26 264 const struct skl_dsp_ops *ops; f0a550a88 Guneshwor Singh 2017-08-02 265 struct skl_dsp_cores *cores; 73a675816 Vinod Koul 2016-07-26 266 int ret; d255b0955 Jeeja KP 2015-07-21 267 d255b0955 Jeeja KP 2015-07-21 268 /* enable ppcap interrupt */ d255b0955 Jeeja KP 2015-07-21 269 snd_hdac_ext_bus_ppcap_enable(&skl->ebus, true); d255b0955 Jeeja KP 2015-07-21 270 snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, true); d255b0955 Jeeja KP 2015-07-21 271 d255b0955 Jeeja KP 2015-07-21 272 /* read the BAR of the ADSP MMIO */ d255b0955 Jeeja KP 2015-07-21 273 mmio_base = pci_ioremap_bar(skl->pci, 4); d255b0955 Jeeja KP 2015-07-21 274 if (mmio_base == NULL) { d255b0955 Jeeja KP 2015-07-21 275 dev_err(bus->dev, "ioremap error\n"); d255b0955 Jeeja KP 2015-07-21 276 return -ENXIO; d255b0955 Jeeja KP 2015-07-21 277 } d255b0955 Jeeja KP 2015-07-21 278 73a675816 Vinod Koul 2016-07-26 279 ops = skl_get_dsp_ops(skl->pci->device); f77d443c4 Subhransu S. Prusty 2017-08-22 280 if (!ops) { f77d443c4 Subhransu S. Prusty 2017-08-22 281 goto unmap_mmio; f77d443c4 Subhransu S. Prusty 2017-08-22 282 ret = -EIO; f77d443c4 Subhransu S. Prusty 2017-08-22 283 } bc23ca35c Jeeja KP 2016-03-11 284 73a675816 Vinod Koul 2016-07-26 285 loader_ops = ops->loader_ops(); 73a675816 Vinod Koul 2016-07-26 286 ret = ops->init(bus->dev, mmio_base, irq, 73a675816 Vinod Koul 2016-07-26 287 skl->fw_name, loader_ops, 73a675816 Vinod Koul 2016-07-26 288 &skl->skl_sst); bc23ca35c Jeeja KP 2016-03-11 289 2ac454ffe Jeeja KP 2015-10-22 290 if (ret < 0) f77d443c4 Subhransu S. Prusty 2017-08-22 291 goto unmap_mmio; 2ac454ffe Jeeja KP 2015-10-22 292 7bd86a305 G Kranthi 2017-03-13 293 skl->skl_sst->dsp_ops = ops; f0a550a88 Guneshwor Singh 2017-08-02 294 cores = &skl->skl_sst->cores; f0a550a88 Guneshwor Singh 2017-08-02 295 cores->count = ops->num_cores; f0a550a88 Guneshwor Singh 2017-08-02 296 f0a550a88 Guneshwor Singh 2017-08-02 297 cores->state = kcalloc(cores->count, sizeof(*cores->state), GFP_KERNEL); f77d443c4 Subhransu S. Prusty 2017-08-22 298 if (!cores->state) { f77d443c4 Subhransu S. Prusty 2017-08-22 299 ret = -ENOMEM; f77d443c4 Subhransu S. Prusty 2017-08-22 300 goto unmap_mmio; f77d443c4 Subhransu S. Prusty 2017-08-22 301 } f0a550a88 Guneshwor Singh 2017-08-02 302 f0a550a88 Guneshwor Singh 2017-08-02 303 cores->usage_count = kcalloc(cores->count, sizeof(*cores->usage_count), f0a550a88 Guneshwor Singh 2017-08-02 304 GFP_KERNEL); f0a550a88 Guneshwor Singh 2017-08-02 305 if (!cores->usage_count) { f77d443c4 Subhransu S. Prusty 2017-08-22 306 ret = -ENOMEM; f77d443c4 Subhransu S. Prusty 2017-08-22 307 goto free_core_state; f0a550a88 Guneshwor Singh 2017-08-02 308 } 363d45385 Dharageswari R 2017-08-02 309 d255b0955 Jeeja KP 2015-07-21 310 dev_dbg(bus->dev, "dsp registration status=%d\n", ret); d255b0955 Jeeja KP 2015-07-21 311 f77d443c4 Subhransu S. Prusty 2017-08-22 312 return 0; f77d443c4 Subhransu S. Prusty 2017-08-22 313 f77d443c4 Subhransu S. Prusty 2017-08-22 314 free_core_state: f77d443c4 Subhransu S. Prusty 2017-08-22 315 kfree(cores->state); f77d443c4 Subhransu S. Prusty 2017-08-22 316 f77d443c4 Subhransu S. Prusty 2017-08-22 317 unmap_mmio: f77d443c4 Subhransu S. Prusty 2017-08-22 318 iounmap(mmio_base); f77d443c4 Subhransu S. Prusty 2017-08-22 319 d255b0955 Jeeja KP 2015-07-21 @320 return ret; d255b0955 Jeeja KP 2015-07-21 321 } d255b0955 Jeeja KP 2015-07-21 322 :::::: The code at line 320 was first introduced by commit :::::: d255b09555e8ea0e517a4d3368ec60097ac9cf96 ASoC: Intel: Skylake: Add dsp and ipc init helpers :::::: TO: Jeeja KP <jeeja.kp(a)intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: tegra: Fix unused variable warning
by Takashi Iwai 23 Aug '17

23 Aug '17

Fixes: 2a8bd83af30e ("ASoC: tegra: Remove superfluous snd_soc_jack_free_gpios() call") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/tegra/tegra_wm8903.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index ce498cc0914b..05f95944669a 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -203,7 +203,6 @@ static int tegra_wm8903_remove(struct snd_soc_card *card) snd_soc_get_pcm_runtime(card, card->dai_link[0].name); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); wm8903_mic_detect(codec, NULL, 0, 0); -- 2.14.0

2 1

[alsa-devel] [PATCH] ASoC: wm8524: remove unnecessary snd_soc_unregister_platform()
by Kuninori Morimoto 23 Aug '17

23 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> wm8524 doesn't register platform, unregister platform is not needed Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/wm8524.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index b66147e..856a695 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -233,10 +233,8 @@ static int wm8524_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8524, &wm8524_dai, 1); - if (ret < 0) { + if (ret < 0) dev_err(&pdev->dev, "Failed to register codec: %d\n", ret); - snd_soc_unregister_platform(&pdev->dev); - } return ret; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: simple-scu-card: Add pm callbacks to platform driver
by Kuninori Morimoto 23 Aug '17

23 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM operations are going to happen. This is needed to get suspend/resume working correctly when the audio is using simple-scu-card. Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-scu-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 0bb3da7..48606c6 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -300,6 +300,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev) static struct platform_driver asoc_simple_card = { .driver = { .name = "simple-scu-audio-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: audio-graph-card: Add pm callbacks to platform driver
by Kuninori Morimoto 23 Aug '17

23 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM operations are going to happen. This is needed to get suspend/resume working correctly when the audio is using audio-graph-card. Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 2f7c8cf..488c52f 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -325,6 +325,7 @@ static int asoc_graph_card_remove(struct platform_device *pdev) static struct platform_driver asoc_graph_card = { .driver = { .name = "asoc-audio-graph-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_graph_of_match, }, .probe = asoc_graph_card_probe, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: audio-graph-scu-card: Add pm callbacks to platform driver
by Kuninori Morimoto 23 Aug '17

23 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM operations are going to happen. This is needed to get suspend/resume working correctly when the audio is using audio-graph-scu-card. Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-scu-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 06b6fcf..a967aa1 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -401,6 +401,7 @@ static int asoc_graph_card_remove(struct platform_device *pdev) static struct platform_driver asoc_graph_card = { .driver = { .name = "asoc-audio-graph-scu-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_graph_of_match, }, .probe = asoc_graph_card_probe, -- 1.9.1

2 1

[alsa-devel] [PATCH 0/4] ASoC: Intel: Skylake: Fixes to free resources on init failure
by Subhransu S. Prusty 23 Aug '17

23 Aug '17

Some resources are not freed for ipc and dsp init failures. This results in random failures in a stress test environment. So fixed to free the required resources and also fixed the DSP core reference counting for firmware load failure. Subhransu S. Prusty (4): ASoC: Intel: Skylake: Fix to free dsp resource on ipc_init failure ASoC: Intel: Skylake: Fix to free resources for dsp_init failure ASoC: Intel: Skylake: Fix to free correct dev id in free_irq ASoC: Intel: Skylake: Fix DSP core ref count for init failure sound/soc/intel/skylake/bxt-sst.c | 4 +++- sound/soc/intel/skylake/cnl-sst.c | 4 +++- sound/soc/intel/skylake/skl-messages.c | 26 +++++++++++++++++++------- sound/soc/intel/skylake/skl-sst-dsp.c | 4 ++-- sound/soc/intel/skylake/skl-sst.c | 4 +++- sound/soc/intel/skylake/skl.c | 2 +- 6 files changed, 31 insertions(+), 13 deletions(-) -- 1.9.1

3 9

[alsa-devel] [PATCH][next] ASoC: tegra: remove unused variable machine, fixes build warning
by Colin King 23 Aug '17

23 Aug '17

From: Colin Ian King <colin.king(a)canonical.com> Remove unused variable 'machine', fixes build warning: sound/soc/tegra/tegra_wm8903.c: In function 'tegra_wm8903_remove': sound/soc/tegra/tegra_wm8903.c:206:23: warning: unused variable 'machine' [-Wunused-variable] struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); ^~~~~~~ Fixes: 2a8bd83af30e ("ASoC: tegra: Remove superfluous snd_soc_jack_free_gpios() call") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/tegra/tegra_wm8903.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 2384cb9cb95c..18bdae59a4df 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -203,7 +203,6 @@ static int tegra_wm8903_remove(struct snd_soc_card *card) snd_soc_get_pcm_runtime(card, card->dai_link[0].name); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); wm8903_mic_detect(codec, NULL, 0, 0); -- 2.14.1

2 1

[alsa-devel] [PATCH] ALSA: USB-MIDI: Use common error handling code in __snd_usbmidi_create()
by SF Markus Elfring 23 Aug '17

23 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 23 Aug 2017 12:20:07 +0200 Add jump targets so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/usb/midi.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index a35f41467237..bd9d02268724 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -2435,10 +2435,8 @@ int __snd_usbmidi_create(struct snd_card *card, err = -ENXIO; break; } - if (err < 0) { - kfree(umidi); - return err; - } + if (err < 0) + goto free_midi; /* create rawmidi device */ out_ports = 0; @@ -2448,23 +2446,25 @@ int __snd_usbmidi_create(struct snd_card *card, in_ports += hweight16(endpoints[i].in_cables); } err = snd_usbmidi_create_rawmidi(umidi, out_ports, in_ports); - if (err < 0) { - kfree(umidi); - return err; - } + if (err < 0) + goto free_midi; /* create endpoint/port structures */ if (quirk && quirk->type == QUIRK_MIDI_MIDIMAN) err = snd_usbmidi_create_endpoints_midiman(umidi, &endpoints[0]); else err = snd_usbmidi_create_endpoints(umidi, endpoints); - if (err < 0) { - return err; - } + if (err < 0) + goto exit; usb_autopm_get_interface_no_resume(umidi->iface); list_add_tail(&umidi->list, midi_list); return 0; + +free_midi: + kfree(umidi); +exit: + return err; } EXPORT_SYMBOL(__snd_usbmidi_create); -- 2.14.0

3 2

[alsa-devel] [PATCH] ASoC: Add delay quirk for H650e/Jabra 550a USB headsets
by Joakim Tjernlund 23 Aug '17

23 Aug '17

These headsets reports a lot of: cannot set freq 44100 to ep 0x81 and need a small delay between sample rate settings, just like Zoom R16/24. Add both headsets to the Zoom R16/24 quirk for a 1 ms delay between control msgs. Signed-off-by: Joakim Tjernlund <joakim.tjernlund(a)infinera.com> --- sound/usb/quirks.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index eb4b9f7a571e..6f395150243f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1308,10 +1308,13 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); - /* Zoom R16/24 needs a tiny delay here, otherwise requests like - * get/set frequency return as failed despite actually succeeding. + /* Zoom R16/24, Logitech H650e, Jabra 550a needs a tiny delay here, + * otherwise requests like get/set frequency return as failed despite + * actually succeeding. */ - if (chip->usb_id == USB_ID(0x1686, 0x00dd) && + if ((chip->usb_id == USB_ID(0x1686, 0x00dd) || + chip->usb_id == USB_ID(0x046d, 0x0a46) || + chip->usb_id == USB_ID(0x0b0e, 0x0349)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(1); } -- 2.13.5

2 2

[alsa-devel] [PATCH] ALSA: core: Use common error handling code in two functions
by SF Markus Elfring 23 Aug '17

23 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 22 Aug 2017 21:01:01 +0200 Add jump targets so that a bit of exception handling can be better reused at the end of these functions. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/core/pcm.c | 30 ++++++++++++++++++------------ sound/core/timer.c | 10 ++++++---- 2 files changed, 24 insertions(+), 16 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 89c7485519cb..048df9658f50 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -783,21 +783,27 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device, INIT_LIST_HEAD(&pcm->list); if (id) strlcpy(pcm->id, id, sizeof(pcm->id)); - if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) { - snd_pcm_free(pcm); - return err; - } - if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_CAPTURE, capture_count)) < 0) { - snd_pcm_free(pcm); - return err; - } - if ((err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops)) < 0) { - snd_pcm_free(pcm); - return err; - } + + err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, + playback_count); + if (err < 0) + goto free_pcm; + + err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_CAPTURE, capture_count); + if (err < 0) + goto free_pcm; + + err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops); + if (err < 0) + goto free_pcm; + if (rpcm) *rpcm = pcm; return 0; + +free_pcm: + snd_pcm_free(pcm); + return err; } /** diff --git a/sound/core/timer.c b/sound/core/timer.c index a9b9a277e00c..6d73a63f6e2b 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -2096,8 +2096,7 @@ static int __init alsa_timer_init(void) err = snd_timer_register_system(); if (err < 0) { pr_err("ALSA: unable to register system timer (%i)\n", err); - put_device(&timer_dev); - return err; + goto put_timer; } err = snd_register_device(SNDRV_DEVICE_TYPE_TIMER, NULL, 0, @@ -2105,12 +2104,15 @@ static int __init alsa_timer_init(void) if (err < 0) { pr_err("ALSA: unable to register timer device (%i)\n", err); snd_timer_free_all(); - put_device(&timer_dev); - return err; + goto put_timer; } snd_timer_proc_init(); return 0; + +put_timer: + put_device(&timer_dev); + return err; } static void __exit alsa_timer_exit(void) -- 2.14.0

2 14

[alsa-devel] [sound:topic/asus-e100h-4.13 13/17] arch/x86/platform/atom/punit_atom_debug.c:230:41: error: 'punit_freeze_nb' undeclared
by kbuild test robot 23 Aug '17

23 Aug '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/asus-e100h-4.13 head: b50dc59e3e31709349588be7a1ac8991e9c5f8a5 commit: 7cfa6efe32e12013183d68f991dd586814b740da [13/17] Intel Atom suspend: add debug check for S0ix blockers config: x86_64-randconfig-x008-201734 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 7cfa6efe32e12013183d68f991dd586814b740da # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): arch/x86/platform/atom/punit_atom_debug.c: In function 'punit_atom_debug_exit': >> arch/x86/platform/atom/punit_atom_debug.c:230:41: error: 'punit_freeze_nb' undeclared (first use in this function) intel_idle_freeze_notifier_unregister(&punit_freeze_nb.nb); ^~~~~~~~~~~~~~~ arch/x86/platform/atom/punit_atom_debug.c:230:41: note: each undeclared identifier is reported only once for each function it appears in vim +/punit_freeze_nb +230 arch/x86/platform/atom/punit_atom_debug.c 226 227 static void __exit punit_atom_debug_exit(void) 228 { 229 #ifdef CONFIG_PM_DEBUG > 230 intel_idle_freeze_notifier_unregister(&punit_freeze_nb.nb); 231 #endif 232 punit_dbgfs_unregister(); 233 } 234 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ALSA: hda - Add stereo mic quirk for Lenovo G50-70 (17aa:3978)
by Takashi Iwai 23 Aug '17

23 Aug '17

Lenovo G50-70 (17aa:3978) with Conexant codec chip requires the similar workaround for the inverted stereo dmic like other Lenovo models. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1020657 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 8c1289963c80..a81aacf684b2 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -947,6 +947,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo G50-70", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), -- 2.14.0

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[alsa-devel] [PATCH v2 0/2] Patches to add support pdm controller for rk3328
by Sugar Zhang 23 Aug '17

23 Aug '17

These patches to add support pdm for rk3328 socs. Changes in v2: - separate pinctrl pins remove unused fsync clk, it's debug pins Sugar Zhang (2): arm64: dts: rockchip: add rk3328 pdm node ASoC: rockchip: separate pinctrl pins from each other .../devicetree/bindings/sound/rockchip,pdm.txt | 7 +- arch/arm64/boot/dts/rockchip/rk3328.dtsi | 77 ++++++++++++++++++++++ 2 files changed, 82 insertions(+), 2 deletions(-) -- 2.7.4

3 3

[alsa-devel] [sound:topic/asus-e100h-4.13 5/17] punit_atom_debug.c:undefined reference to `intel_idle_freeze_notifier_unregister'
by kbuild test robot 23 Aug '17

23 Aug '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/asus-e100h-4.13 head: 412b835760686e51fecad5b77f09372b500de370 commit: 9563001ade435a8ca23bbe41bd3727746c9c0ab3 [5/17] Intel Atom suspend: add debug check for S0ix blockers config: x86_64-randconfig-ne0-08230254 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 9563001ade435a8ca23bbe41bd3727746c9c0ab3 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): arch/x86/platform/atom/punit_atom_debug.o: In function `punit_atom_debug_exit': >> punit_atom_debug.c:(.exit.text+0xc): undefined reference to `intel_idle_freeze_notifier_unregister' arch/x86/platform/atom/punit_atom_debug.o: In function `punit_atom_debug_init': >> punit_atom_debug.c:(.init.text+0xf0): undefined reference to `intel_idle_freeze_notifier_register' drivers/platform/x86/pmc_atom.o: In function `pmc_atom_init': pmc_atom.c:(.init.text+0x3a7): undefined reference to `intel_idle_freeze_notifier_register' --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ALSA: cmipci: Use common error handling code in snd_cmipci_probe()
by SF Markus Elfring 22 Aug '17

22 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 22 Aug 2017 19:58:30 +0200 Add a jump target so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/cmipci.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index a460cb63e971..27817c84f6ef 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -3295,20 +3295,23 @@ static int snd_cmipci_probe(struct pci_dev *pci, break; } - if ((err = snd_cmipci_create(card, pci, dev, &cm)) < 0) { - snd_card_free(card); - return err; - } + err = snd_cmipci_create(card, pci, dev, &cm); + if (err < 0) + goto free_card; + card->private_data = cm; - if ((err = snd_card_register(card)) < 0) { - snd_card_free(card); - return err; - } + err = snd_card_register(card); + if (err < 0) + goto free_card; + pci_set_drvdata(pci, card); dev++; return 0; +free_card: + snd_card_free(card); + return err; } static void snd_cmipci_remove(struct pci_dev *pci) -- 2.14.0

2 1

[alsa-devel] [sound:topic/asus-e100h-4.13 5/17] pmc_atom.c:undefined reference to `intel_idle_freeze_notifier_register'
by kbuild test robot 22 Aug '17

22 Aug '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/asus-e100h-4.13 head: 412b835760686e51fecad5b77f09372b500de370 commit: 9563001ade435a8ca23bbe41bd3727746c9c0ab3 [5/17] Intel Atom suspend: add debug check for S0ix blockers config: x86_64-nfsroot (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 9563001ade435a8ca23bbe41bd3727746c9c0ab3 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): drivers/platform/x86/pmc_atom.o: In function `pmc_atom_init': >> pmc_atom.c:(.init.text+0x243): undefined reference to `intel_idle_freeze_notifier_register' --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ALSA: ctxfi: Use common error handling code in two functions
by SF Markus Elfring 22 Aug '17

22 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 22 Aug 2017 18:43:42 +0200 Add jump targets so that a bit of exception handling can be better reused at the end of these functions. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/ctxfi/ctpcm.c | 42 ++++++++++++++++++++++-------------------- 1 file changed, 22 insertions(+), 20 deletions(-) diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 974978041558..dece6a81060d 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -140,27 +140,28 @@ static int ct_pcm_playback_open(struct snd_pcm_substream *substream) err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (err < 0) { - kfree(apcm); - return err; - } + if (err < 0) + goto free_pcm; + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 1024, UINT_MAX); - if (err < 0) { - kfree(apcm); - return err; - } + if (err < 0) + goto free_pcm; apcm->timer = ct_timer_instance_new(atc->timer, apcm); if (!apcm->timer) { - kfree(apcm); - return -ENOMEM; + err = -ENOMEM; + goto free_pcm; } runtime->private_data = apcm; runtime->private_free = ct_atc_pcm_free_substream; return 0; + +free_pcm: + kfree(apcm); + return err; } static int ct_pcm_playback_close(struct snd_pcm_substream *substream) @@ -286,27 +287,28 @@ static int ct_pcm_capture_open(struct snd_pcm_substream *substream) err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (err < 0) { - kfree(apcm); - return err; - } + if (err < 0) + goto free_pcm; + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 1024, UINT_MAX); - if (err < 0) { - kfree(apcm); - return err; - } + if (err < 0) + goto free_pcm; apcm->timer = ct_timer_instance_new(atc->timer, apcm); if (!apcm->timer) { - kfree(apcm); - return -ENOMEM; + err = -ENOMEM; + goto free_pcm; } runtime->private_data = apcm; runtime->private_free = ct_atc_pcm_free_substream; return 0; + +free_pcm: + kfree(apcm); + return err; } static int ct_pcm_capture_close(struct snd_pcm_substream *substream) -- 2.14.0

2 1

[alsa-devel] [PATCH] ASoC: rsnd: remove unused rsnd_xxx_to_dma()
by Kuninori Morimoto 22 Aug '17

22 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 1 - sound/soc/sh/rcar/ssi.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 8e9a1de..f67365d 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -27,7 +27,6 @@ struct rsnd_src { #define RSND_SRC_NAME_SIZE 16 #define rsnd_src_get(priv, id) ((struct rsnd_src *)(priv->src) + id) -#define rsnd_src_to_dma(src) ((src)->dma) #define rsnd_src_nr(priv) ((priv)->src_nr) #define rsnd_src_sync_is_enabled(mod) (rsnd_mod_to_src(mod)->sen.val) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 8bb4714..fffc07e 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -99,7 +99,6 @@ struct rsnd_ssi { i++) #define rsnd_ssi_get(priv, id) ((struct rsnd_ssi *)(priv->ssi) + id) -#define rsnd_ssi_to_dma(mod) ((ssi)->dma) #define rsnd_ssi_nr(priv) ((priv)->ssi_nr) #define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod) #define rsnd_ssi_flags_has(p, f) ((p)->flags & f) -- 1.9.1

2 6

[alsa-devel] [PATCH v6 0/8] ASoC: rockchip: Parse dai links from dts
by Jeffy Chen 22 Aug '17

22 Aug '17

Currently we are using a fixed list of dai links in the driver. This serial of patches would let the driver parse dai links from dts, so that we can make some of them optional for future boards. Tested on my chromebook bob(with cros 4.4 kernel), it still works after disabled rt5514 codecs in the dts. Changes in v6: Add dmic wakeup delay(not used for now). Changes in v5: Keep the dmic-delay property, but correct it with the name in dt-binding. -- Suggested-by Mark Brown <broonie(a)kernel.org> Changes in v3: Use compatible to match audio codecs -- Suggested-by Matthias Kaehlcke <mka(a)chromium.org> Changes in v2: Let rockchip,codec-names be a required property, because we plan to add more supported codecs to the fixed dai link list in the driver. Jeffy Chen (8): ASoC: rockchip: Correct 'dmic-delay' property name ASoC: rockchip: Use codec of_node and dai_name for rt5514 dsp arm64: dts: rockchip: Add rt5514 dsp for Gru arm64: dts: rockchip: Update rt5514 devices' compatible for Gru ASoC: rockchip: Parse dai links from dts ASoC: rockchip: Add support for DP codec ASoC: rockchip: Add support for DMIC codec dt-bindings: ASoC: rockchip: Update description of rockchip,codec .../bindings/sound/rockchip,rk3399-gru-sound.txt | 2 +- arch/arm64/boot/dts/rockchip/rk3399-gru.dtsi | 7 +- sound/soc/rockchip/Kconfig | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 270 ++++++++++++++------- 4 files changed, 193 insertions(+), 88 deletions(-) -- 2.11.0

2 3

[alsa-devel] [PATCH] ALSA: pcsp: Use common error handling code in snd_card_pcsp_probe()
by SF Markus Elfring 22 Aug '17

22 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 22 Aug 2017 14:00:21 +0200 Add a jump target so that a bit of exception handling can be better reused in this function. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/drivers/pcsp/pcsp.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 72e2d0012084..c8ea4fed3905 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -108,22 +108,17 @@ static int snd_card_pcsp_probe(int devnum, struct device *dev) return err; err = snd_pcsp_create(card); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err) + goto free_card; + if (!nopcm) { err = snd_pcsp_new_pcm(&pcsp_chip); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err) + goto free_card; } err = snd_pcsp_new_mixer(&pcsp_chip, nopcm); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err) + goto free_card; strcpy(card->driver, "PC-Speaker"); strcpy(card->shortname, "pcsp"); @@ -131,7 +126,8 @@ static int snd_card_pcsp_probe(int devnum, struct device *dev) pcsp_chip.port); err = snd_card_register(card); - if (err < 0) { + if (err) { +free_card: snd_card_free(card); return err; } -- 2.14.0

3 13

[alsa-devel] [PATCH] ALSA: ice1712: Add support for STAudio ADCIII
by Takashi Iwai 22 Aug '17

22 Aug '17

STAudio ADCIII has the same SSID as Hoontech STDSP24, but requires a slightly different configuration. This patch allows user to choose this model via model=staudio option to set the proper configuration for the board. Bugzilla: http://bugzilla.suse.com/show_bug.cgi?id=1048934 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/ice1712/hoontech.c | 39 ++++++++++++++++++++++++++++++++++----- sound/pci/ice1712/hoontech.h | 1 + 2 files changed, 35 insertions(+), 5 deletions(-) diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index f37ed0bc203d..fa301d31745b 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -166,7 +166,7 @@ static void snd_ice1712_stdsp24_midi2(struct snd_ice1712 *ice, int activate) mutex_unlock(&ice->gpio_mutex); } -static int snd_ice1712_hoontech_init(struct snd_ice1712 *ice) +static int hoontech_init(struct snd_ice1712 *ice, bool staudio) { struct hoontech_spec *spec; int box, chn; @@ -203,7 +203,10 @@ static int snd_ice1712_hoontech_init(struct snd_ice1712 *ice) ICE1712_STDSP24_3_INSEL(spec->boxbits, 0); /* let's go - activate only functions in first box */ - spec->config = 0; + if (staudio) + spec->config = ICE1712_STDSP24_MUTE; + else + spec->config = 0; /* ICE1712_STDSP24_MUTE | ICE1712_STDSP24_INSEL | ICE1712_STDSP24_DAREAR; */ @@ -226,9 +229,16 @@ static int snd_ice1712_hoontech_init(struct snd_ice1712 *ice) ICE1712_STDSP24_BOX_CHN4 | ICE1712_STDSP24_BOX_MIDI1 | ICE1712_STDSP24_BOX_MIDI2; - spec->boxconfig[1] = - spec->boxconfig[2] = - spec->boxconfig[3] = 0; + if (staudio) { + spec->boxconfig[1] = + spec->boxconfig[2] = + spec->boxconfig[3] = spec->boxconfig[0]; + } else { + spec->boxconfig[1] = + spec->boxconfig[2] = + spec->boxconfig[3] = 0; + } + snd_ice1712_stdsp24_darear(ice, (spec->config & ICE1712_STDSP24_DAREAR) ? 1 : 0); snd_ice1712_stdsp24_mute(ice, @@ -248,6 +258,16 @@ static int snd_ice1712_hoontech_init(struct snd_ice1712 *ice) return 0; } +static int snd_ice1712_hoontech_init(struct snd_ice1712 *ice) +{ + return hoontech_init(ice, false); +} + +static int snd_ice1712_staudio_init(struct snd_ice1712 *ice) +{ + return hoontech_init(ice, true); +} + /* * AK4524 access */ @@ -351,5 +371,14 @@ struct snd_ice1712_card_info snd_ice1712_hoontech_cards[] = { .model = "ez8", .chip_init = snd_ice1712_ez8_init, }, + { + /* STAudio ADCIII has the same SSID as Hoontech StA DSP24, + * thus identified only via the explicit model option + */ + .subvendor = ICE1712_SUBDEVICE_STAUDIO_ADCIII, /* a dummy id */ + .name = "STAudio ADCIII", + .model = "staudio", + .chip_init = snd_ice1712_staudio_init, + }, { } /* terminator */ }; diff --git a/sound/pci/ice1712/hoontech.h b/sound/pci/ice1712/hoontech.h index cc1da1e69ad1..7f94943392ce 100644 --- a/sound/pci/ice1712/hoontech.h +++ b/sound/pci/ice1712/hoontech.h @@ -34,6 +34,7 @@ #define ICE1712_SUBDEVICE_STDSP24_VALUE 0x00010010 /* A dummy id for Hoontech SoundTrack Audio DSP 24 Value */ #define ICE1712_SUBDEVICE_STDSP24_MEDIA7_1 0x16141217 /* Hoontech ST Audio DSP24 Media 7.1 */ #define ICE1712_SUBDEVICE_EVENT_EZ8 0x00010001 /* A dummy id for EZ8 */ +#define ICE1712_SUBDEVICE_STAUDIO_ADCIII 0x00010002 /* A dummy id for STAudio ADCIII */ extern struct snd_ice1712_card_info snd_ice1712_hoontech_cards[]; -- 2.14.0

1 0

[alsa-devel] [PATCH] ALSA: hda - Implement mic-mute LED mode enum
by Takashi Iwai 22 Aug '17

22 Aug '17

Dell laptops have another LED for mic-mute in addition to the master mute. The former is tied with the capture switch (in a reverse way) while the latter is tied with the master playback switch. We already have an enum control to change the behavior for the master mute LED in different ways, e.g. keeping always off or turning off at mute. But, the mic-mute LED has no such management but its behavior is hard-coded. This patch implements an enum control to change the mic-mute LED behavior like what we have for the master mute LED. The ctl provides four modes: keep-on, keep-off, follow-capture and follow-mute. The default mode is the last one, follow-mute, which follows the capture mute, i.e. LED turning on when the capture is off, and turning off when the capture is active. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/dell_wmi_helper.c | 87 ++++++++++++++++++++++++++++++++++++++--- 1 file changed, 81 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/dell_wmi_helper.c b/sound/pci/hda/dell_wmi_helper.c index 7efa7bd7acb2..44b1e15682b9 100644 --- a/sound/pci/hda/dell_wmi_helper.c +++ b/sound/pci/hda/dell_wmi_helper.c @@ -5,12 +5,47 @@ #if IS_ENABLED(CONFIG_DELL_LAPTOP) #include <linux/dell-led.h> +enum { + MICMUTE_LED_ON, + MICMUTE_LED_OFF, + MICMUTE_LED_FOLLOW_CAPTURE, + MICMUTE_LED_FOLLOW_MUTE, +}; + +static int dell_led_mode = MICMUTE_LED_FOLLOW_MUTE; +static int dell_capture; static int dell_led_value; static int (*dell_micmute_led_set_func)(int); static void (*dell_old_cap_hook)(struct hda_codec *, struct snd_kcontrol *, struct snd_ctl_elem_value *); +static void call_micmute_led_update(void) +{ + int val; + + switch (dell_led_mode) { + case MICMUTE_LED_ON: + val = 1; + break; + case MICMUTE_LED_OFF: + val = 0; + break; + case MICMUTE_LED_FOLLOW_CAPTURE: + val = dell_capture; + break; + case MICMUTE_LED_FOLLOW_MUTE: + default: + val = !dell_capture; + break; + } + + if (val == dell_led_value) + return; + dell_led_value = val; + dell_micmute_led_set_func(dell_led_value); +} + static void update_dell_wmi_micmute_led(struct hda_codec *codec, struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -22,15 +57,54 @@ static void update_dell_wmi_micmute_led(struct hda_codec *codec, return; if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { /* TODO: How do I verify if it's a mono or stereo here? */ - int val = (ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]) ? 0 : 1; - if (val == dell_led_value) - return; - dell_led_value = val; - if (dell_micmute_led_set_func) - dell_micmute_led_set_func(dell_led_value); + dell_capture = (ucontrol->value.integer.value[0] || + ucontrol->value.integer.value[1]); + call_micmute_led_update(); } } +static int dell_mic_mute_led_mode_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[] = { + "On", "Off", "Follow Capture", "Follow Mute", + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static int dell_mic_mute_led_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = dell_led_mode; + return 0; +} + +static int dell_mic_mute_led_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned int mode; + + mode = ucontrol->value.enumerated.item[0]; + if (mode > MICMUTE_LED_FOLLOW_MUTE) + mode = MICMUTE_LED_FOLLOW_MUTE; + if (mode == dell_led_mode) + return 0; + dell_led_mode = mode; + call_micmute_led_update(); + return 1; +} + +static const struct snd_kcontrol_new dell_mic_mute_mode_ctls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Mute-LED Mode", + .info = dell_mic_mute_led_mode_info, + .get = dell_mic_mute_led_mode_get, + .put = dell_mic_mute_led_mode_put, + }, + {} +}; static void alc_fixup_dell_wmi(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -55,6 +129,7 @@ static void alc_fixup_dell_wmi(struct hda_codec *codec, dell_old_cap_hook = spec->gen.cap_sync_hook; spec->gen.cap_sync_hook = update_dell_wmi_micmute_led; removefunc = false; + add_mixer(spec, dell_mic_mute_mode_ctls); } } -- 2.14.0

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[alsa-devel] [PATCH v5 0/9] ASoC: rockchip: Parse dai links from dts
by Jeffy Chen 22 Aug '17

22 Aug '17

Currently we are using a fixed list of dai links in the driver. This serial of patches would let the driver parse dai links from dts, so that we can make some of them optional for future boards. Tested on my chromebook bob(with cros 4.4 kernel), it still works after disabled rt5514 codecs in the dts. Changes in v5: Allow searching for dai driver name in snd_soc_find_dai, instead of hacking non-zero device id to avoid legay dai naming. Changes in v4: Use non-zero drv id to avoid legacy dai naming instead of switching to snd_soc_register_codec. Changes in v3: Use compatible to match audio codecs -- Suggested-by Matthias Kaehlcke <mka(a)chromium.org> Changes in v2: Let rockchip,codec-names be a required property, because we plan to add more supported codecs to the fixed dai link list in the driver. Jeffy Chen (9): ASoC: rockchip: Remove obsolete dmic-delay ASoC: soc-core: Allow searching dai driver name in snd_soc_find_dai ASoC: rockchip: Use codec of_node and dai_name for rt5514 dsp arm64: dts: rockchip: Add rt5514 dsp for Gru arm64: dts: rockchip: Update rt5514 devices' compatible for Gru ASoC: rockchip: Parse dai links from dts ASoC: rockchip: Add support for DP codec ASoC: rockchip: Add support for DMIC codec dt-bindings: ASoC: rockchip: Update description of rockchip,codec .../bindings/sound/rockchip,rk3399-gru-sound.txt | 9 +- arch/arm64/boot/dts/rockchip/rk3399-gru.dtsi | 7 +- sound/soc/rockchip/Kconfig | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 263 ++++++++++++++------- sound/soc/soc-core.c | 5 +- 5 files changed, 185 insertions(+), 101 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH] ALSA: firewire: add const qualifier to identifiers for read-only symbols
by Takashi Sakamoto 22 Aug '17

22 Aug '17

Drivers in ALSA firewire stack still includes some symbols which can be moved to a section for read-only symbols. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob.h | 2 +- sound/firewire/bebob/bebob_focusrite.c | 6 +++--- sound/firewire/bebob/bebob_maudio.c | 2 +- sound/firewire/bebob/bebob_terratec.c | 2 +- sound/firewire/bebob/bebob_yamaha_terratec.c | 2 +- sound/firewire/dice/dice.c | 2 +- sound/firewire/fireface/ff-protocol-ff400.c | 2 +- sound/firewire/fireface/ff.c | 2 +- sound/firewire/fireface/ff.h | 4 ++-- sound/firewire/fireworks/fireworks_proc.c | 2 +- sound/firewire/motu/motu.c | 6 +++--- sound/firewire/oxfw/oxfw-scs1x.c | 11 +++++------ sound/firewire/tascam/tascam.c | 2 +- 13 files changed, 22 insertions(+), 23 deletions(-) diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index 17678d6ab5a2..df1b1e94c43c 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -58,7 +58,7 @@ enum snd_bebob_clock_type { struct snd_bebob_clock_spec { unsigned int num; const char *const *labels; - enum snd_bebob_clock_type *types; + const enum snd_bebob_clock_type *types; int (*get)(struct snd_bebob *bebob, unsigned int *id); }; struct snd_bebob_rate_spec { diff --git a/sound/firewire/bebob/bebob_focusrite.c b/sound/firewire/bebob/bebob_focusrite.c index f11090057949..52b8b61ecddd 100644 --- a/sound/firewire/bebob/bebob_focusrite.c +++ b/sound/firewire/bebob/bebob_focusrite.c @@ -103,12 +103,12 @@ saffire_write_quad(struct snd_bebob *bebob, u64 offset, u32 value) &data, sizeof(__be32), 0); } -static enum snd_bebob_clock_type saffirepro_10_clk_src_types[] = { +static const enum snd_bebob_clock_type saffirepro_10_clk_src_types[] = { SND_BEBOB_CLOCK_TYPE_INTERNAL, SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* S/PDIF */ SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* Word Clock */ }; -static enum snd_bebob_clock_type saffirepro_26_clk_src_types[] = { +static const enum snd_bebob_clock_type saffirepro_26_clk_src_types[] = { SND_BEBOB_CLOCK_TYPE_INTERNAL, SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* S/PDIF */ SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* ADAT1 */ @@ -201,7 +201,7 @@ saffirepro_both_clk_src_get(struct snd_bebob *bebob, unsigned int *id) } const struct snd_bebob_spec saffire_le_spec; -static enum snd_bebob_clock_type saffire_both_clk_src_types[] = { +static const enum snd_bebob_clock_type saffire_both_clk_src_types[] = { SND_BEBOB_CLOCK_TYPE_INTERNAL, SND_BEBOB_CLOCK_TYPE_EXTERNAL, }; diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index d10208f92edf..bd55620c6a47 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -340,7 +340,7 @@ static int special_set_rate(struct snd_bebob *bebob, unsigned int rate) } /* Clock source control for special firmware */ -static enum snd_bebob_clock_type special_clk_types[] = { +static const enum snd_bebob_clock_type special_clk_types[] = { SND_BEBOB_CLOCK_TYPE_INTERNAL, /* With digital mute */ SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* SPDIF/ADAT */ SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* Word Clock */ diff --git a/sound/firewire/bebob/bebob_terratec.c b/sound/firewire/bebob/bebob_terratec.c index 2fdaf93e7a8d..9770c2127a7a 100644 --- a/sound/firewire/bebob/bebob_terratec.c +++ b/sound/firewire/bebob/bebob_terratec.c @@ -8,7 +8,7 @@ #include "./bebob.h" -static enum snd_bebob_clock_type phase88_rack_clk_src_types[] = { +static const enum snd_bebob_clock_type phase88_rack_clk_src_types[] = { SND_BEBOB_CLOCK_TYPE_INTERNAL, SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* S/PDIF */ SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* Word Clock */ diff --git a/sound/firewire/bebob/bebob_yamaha_terratec.c b/sound/firewire/bebob/bebob_yamaha_terratec.c index a6be3e7138e0..8bd78fef3516 100644 --- a/sound/firewire/bebob/bebob_yamaha_terratec.c +++ b/sound/firewire/bebob/bebob_yamaha_terratec.c @@ -31,7 +31,7 @@ * Yamaha GO 44 and GO 46. Yamaha and Terratec had cooperated for these models. */ -static enum snd_bebob_clock_type clk_src_types[] = { +static const enum snd_bebob_clock_type clk_src_types[] = { SND_BEBOB_CLOCK_TYPE_INTERNAL, SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* S/PDIF */ }; diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 25e9f77275c4..4ddb4cdd054b 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -26,7 +26,7 @@ MODULE_LICENSE("GPL v2"); */ static bool force_two_pcm_support(struct fw_unit *unit) { - const char *const models[] = { + static const char *const models[] = { /* TC Electronic models. */ "StudioKonnekt48", /* Focusrite models. */ diff --git a/sound/firewire/fireface/ff-protocol-ff400.c b/sound/firewire/fireface/ff-protocol-ff400.c index fcec6de80eeb..12aa15df435d 100644 --- a/sound/firewire/fireface/ff-protocol-ff400.c +++ b/sound/firewire/fireface/ff-protocol-ff400.c @@ -356,7 +356,7 @@ static void ff400_dump_clock_config(struct snd_ff *ff, snd_iprintf(buffer, "Sync to clock source: %s\n", src); } -struct snd_ff_protocol snd_ff_protocol_ff400 = { +const struct snd_ff_protocol snd_ff_protocol_ff400 = { .get_clock = ff400_get_clock, .begin_session = ff400_begin_session, .finish_session = ff400_finish_session, diff --git a/sound/firewire/fireface/ff.c b/sound/firewire/fireface/ff.c index eee7c8eac7a6..4974bc7980e9 100644 --- a/sound/firewire/fireface/ff.c +++ b/sound/firewire/fireface/ff.c @@ -157,7 +157,7 @@ static void snd_ff_remove(struct fw_unit *unit) } } -static struct snd_ff_spec spec_ff400 = { +static const struct snd_ff_spec spec_ff400 = { .name = "Fireface400", .pcm_capture_channels = {18, 14, 10}, .pcm_playback_channels = {18, 14, 10}, diff --git a/sound/firewire/fireface/ff.h b/sound/firewire/fireface/ff.h index 3cb812a50030..64df44beb950 100644 --- a/sound/firewire/fireface/ff.h +++ b/sound/firewire/fireface/ff.h @@ -47,7 +47,7 @@ struct snd_ff_spec { unsigned int midi_in_ports; unsigned int midi_out_ports; - struct snd_ff_protocol *protocol; + const struct snd_ff_protocol *protocol; }; struct snd_ff { @@ -112,7 +112,7 @@ struct snd_ff_protocol { u64 midi_rx_port_1_reg; }; -extern struct snd_ff_protocol snd_ff_protocol_ff400; +extern const struct snd_ff_protocol snd_ff_protocol_ff400; int snd_ff_transaction_register(struct snd_ff *ff); int snd_ff_transaction_reregister(struct snd_ff *ff); diff --git a/sound/firewire/fireworks/fireworks_proc.c b/sound/firewire/fireworks/fireworks_proc.c index beb0a0ffee57..9c21f31b8b21 100644 --- a/sound/firewire/fireworks/fireworks_proc.c +++ b/sound/firewire/fireworks/fireworks_proc.c @@ -12,7 +12,7 @@ static inline const char* get_phys_name(struct snd_efw_phys_grp *grp, bool input) { - const char *const ch_type[] = { + static const char *const ch_type[] = { "Analog", "S/PDIF", "ADAT", "S/PDIF or ADAT", "Mirroring", "Headphones", "I2S", "Guitar", "Pirzo Guitar", "Guitar String", }; diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 2bbb764822fc..0d6b526105ab 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -194,7 +194,7 @@ static void motu_bus_update(struct fw_unit *unit) snd_motu_transaction_reregister(motu); } -static struct snd_motu_spec motu_828mk2 = { +static const struct snd_motu_spec motu_828mk2 = { .name = "828mk2", .protocol = &snd_motu_protocol_v2, .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | @@ -208,7 +208,7 @@ static struct snd_motu_spec motu_828mk2 = { .analog_out_ports = 8, }; -static struct snd_motu_spec motu_828mk3 = { +static const struct snd_motu_spec motu_828mk3 = { .name = "828mk3", .protocol = &snd_motu_protocol_v3, .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | @@ -225,7 +225,7 @@ static struct snd_motu_spec motu_828mk3 = { .analog_out_ports = 8, }; -static struct snd_motu_spec motu_audio_express = { +static const struct snd_motu_spec motu_audio_express = { .name = "AudioExpress", .protocol = &snd_motu_protocol_v3, .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | diff --git a/sound/firewire/oxfw/oxfw-scs1x.c b/sound/firewire/oxfw/oxfw-scs1x.c index 93209ebd9121..02d595665898 100644 --- a/sound/firewire/oxfw/oxfw-scs1x.c +++ b/sound/firewire/oxfw/oxfw-scs1x.c @@ -297,12 +297,6 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *stream, int up) } } -static const struct snd_rawmidi_ops midi_capture_ops = { - .open = midi_capture_open, - .close = midi_capture_close, - .trigger = midi_capture_trigger, -}; - static int midi_playback_open(struct snd_rawmidi_substream *stream) { return 0; @@ -363,6 +357,11 @@ void snd_oxfw_scs1x_update(struct snd_oxfw *oxfw) int snd_oxfw_scs1x_add(struct snd_oxfw *oxfw) { + static const struct snd_rawmidi_ops midi_capture_ops = { + .open = midi_capture_open, + .close = midi_capture_close, + .trigger = midi_capture_trigger, + }; static const struct snd_rawmidi_ops midi_playback_ops = { .open = midi_playback_open, .close = midi_playback_close, diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index 9dc93a7eb9da..44ad41fb7374 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -12,7 +12,7 @@ MODULE_DESCRIPTION("TASCAM FireWire series Driver"); MODULE_AUTHOR("Takashi Sakamoto <o-takashi(a)sakamocchi.jp>"); MODULE_LICENSE("GPL v2"); -static struct snd_tscm_spec model_specs[] = { +static const struct snd_tscm_spec model_specs[] = { { .name = "FW-1884", .has_adat = true, -- 2.11.0

2 1

[alsa-devel] GPIO issue with RT5670
by Pierre-Louis Bossart 22 Aug '17

22 Aug '17

the commit f10e4bf6632b5be11cea875b66ba959833a69258 gpio: acpi: Even more tighten up ACPI GPIO lookups now generates the following issue [ 9.694204] rt5670 i2c-10EC5672:00: ASoC: Cannot get gpio at index 0: -2 [ 9.694293] rt5670 i2c-10EC5672:00: Adding jack GPIO failed This is the 3rd occurrence of an audio issue with this commit [1][2], what is the recommended fix here since this codec is not only used in ACPI-based devices and do we need to check for more issues in audio drivers? Thanks -Pierre [1] https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/commit/?h… [2] https://bugzilla.kernel.org/show_bug.cgi?id=115531

3 3

[alsa-devel] [PATCH 0/8] ASoC: jack: Manage gpios via devres
by Takashi Iwai 22 Aug '17

22 Aug '17

Hi, while debugging the recent GPIO regression with (still not-merged) cx2072x codec driver, I noticed the driver misses the snd_soc_jack_free_gpios() call. This patch set is rather making it redundant, change the ASoC jack code to manage gpios via devres. It follows the cleanups of superfluous snd_soc_jack_free_gpios() in each driver. This resulted in lots of reductions of remove callbacks. Takashi === Takashi Iwai (8): ASoC: jack: Manage gpios via devres ASoC: fsl: Remove superfluous snd_soc_jack_free_gpios() call ASoC: simple-card: Remove superfluous snd_soc_jack_free_gpios() call ASoC: intel: Remove superfluous snd_soc_jack_free_gpios() call ASoC: omap: Remove superfluous snd_soc_jack_free_gpios() call ASoC: pxa: Remove superfluous snd_soc_jack_free_gpios() call ASoC: samsung: Remove superfluous snd_soc_jack_free_gpios() call ASoC: tegra: Remove superfluous snd_soc_jack_free_gpios() call sound/soc/fsl/imx-es8328.c | 3 -- sound/soc/generic/simple-card.c | 10 ------- sound/soc/intel/boards/byt-max98090.c | 12 -------- sound/soc/omap/ams-delta.c | 10 ------- sound/soc/omap/omap-twl4030.c | 13 --------- sound/soc/omap/rx51.c | 9 ------ sound/soc/pxa/hx4700.c | 8 ------ sound/soc/samsung/h1940_uda1380.c | 9 ------ sound/soc/samsung/rx1950_uda1380.c | 10 ------- sound/soc/samsung/smartq_wm8987.c | 9 ------ sound/soc/soc-jack.c | 52 +++++++++++++++++++++++++++-------- sound/soc/tegra/tegra_alc5632.c | 13 --------- sound/soc/tegra/tegra_max98090.c | 18 ------------ sound/soc/tegra/tegra_rt5640.c | 13 --------- sound/soc/tegra/tegra_rt5677.c | 18 ------------ sound/soc/tegra/tegra_wm8903.c | 5 ---- 16 files changed, 41 insertions(+), 171 deletions(-) -- 2.14.0

2 16

[alsa-devel] [bug] alsa-tools-1.1.3 compilation fails with clang due to wrong inline declaration.
by moxian 22 Aug '17

22 Aug '17

I've tried compiling alsa-tools-1.3.0 with clang, and the compilation fails with the following: $ CC=clang CXX=clang++ make ./as10k1 <...snip...> clang -fno-strict-aliasing -O2 -Wall -pipe -g -o as10k1 as10k1.o parse.o assemble.o macro.o as10k1.o: In function `header': /home/moxian/work/alsa-tools/alsa-tools-1.1.3/as10k1/as10k1.c:524: undefined reference to `output_tram_line' /home/moxian/work/alsa-tools/alsa-tools-1.1.3/as10k1/as10k1.c:527: undefined reference to `output_tram_line' /home/moxian/work/alsa-tools/alsa-tools-1.1.3/as10k1/as10k1.c:539: undefined reference to `output_tram_line' /home/moxian/work/alsa-tools/alsa-tools-1.1.3/as10k1/as10k1.c:543: undefined reference to `output_tram_line' clang-3.9: error: linker command failed with exit code 1 (use -v to see invocation) make[2]: *** [Makefile:320: as10k1] Error 1 make[2]: Leaving directory '/home/moxian/work/alsa-tools/alsa-tools-1.1.3/as10k1' make[1]: *** [Makefile:354: all-recursive] Error 1 make[1]: Leaving directory '/home/moxian/work/alsa-tools/alsa-tools-1.1.3/as10k1' make: *** [Makefile:9: all] Error 1 Clang devs claim that it is not a bug in clang (as I thought originally), but rather invalid code, which happens to compile with gcc by accident. This is due to `output_tram_line` function being declared as `inline void output_tram_line`. The intent was probably to declare it as `static inline void output_tram_line`. Please see https://bugs.llvm.org/show_bug.cgi?id=33768 for more details. Adding `static` to the function definition makes everything compile (at least everything that compiles with gcc already).

2 1

[alsa-devel] [PATCH] ASoC: Intel: Headset button support in kabylake machine driver
by Harsha Priya 22 Aug '17

22 Aug '17

This patch adds headset button support for kabylake machine driver (kbl_rt5663_rt5514_max98927). Signed-off-by: Hsin-yu Chao <hychao(a)google.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 441a397..b350aa7 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -18,6 +18,7 @@ * GNU General Public License for more details. */ +#include <linux/input.h> #include <linux/module.h> #include <linux/platform_device.h> #include <sound/core.h> @@ -174,6 +175,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) int ret; struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_jack *jack; /* * Headset buttons map to the google Reference headset. @@ -188,6 +190,12 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } + jack = &ctx->kabylake_headset; + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + rt5663_set_jack_detect(codec, &ctx->kabylake_headset); ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "DMIC"); -- 1.9.1

3 2

[alsa-devel] [PATCH] alsa-lib: topology: Fix private data for BEs
by Liam Girdwood 22 Aug '17

22 Aug '17

Private data was not being added to BEs. Fix this. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/data.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/src/topology/data.c b/src/topology/data.c index 5b83b9f..f96ff9b 100644 --- a/src/topology/data.c +++ b/src/topology/data.c @@ -50,7 +50,9 @@ struct snd_soc_tplg_private *get_priv_data(struct tplg_elem *elem) case SND_TPLG_TYPE_DAI: priv = &elem->dai->priv; break; - + case SND_TPLG_TYPE_BE: + priv = &elem->link->priv; + break; default: SNDERR("error: '%s': no support for private data for type %d\n", elem->id, elem->type); -- 1.9.1

2 1

Re: [alsa-devel] [PATCH] ASoC: dapm: Avoid creating kcontrol for params
by anish kumar 22 Aug '17

22 Aug '17

> > When num_params is defined as one in case of > codec to codec dai link, there is no point in > allocating memory for kcontrol. > > Signed-off-by: anish kumar <yesanishhere(a)gmail.com> > --- > include/sound/soc-dapm.h | 1 - > sound/soc/soc-dapm.c | 147 ++++++++++++++++++++++++----------------------- >>This is a very big change and that commit log doesn't describe it in any >>detail or really motivate it. What exactly does this do and why? Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol (this is currently happening) and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly and also it remove unnecessary checks.

2 1

[alsa-devel] [PATCH] ASoC: rt5670: Fix GPIO headset detection regression
by Takashi Iwai 22 Aug '17

22 Aug '17

RT5670 codec driver and its machine driver for Intel CHT assume the implicit GPIO mapping on the index 0 while BIOS on most devices don't provide it. The recent commit f10e4bf6632b ("gpio: acpi: Even more tighten up ACPI GPIO lookups") restricts such cases and it resulted in a regression where the headset jack setup fails like: rt5670 i2c-10EC5672:00: ASoC: Cannot get gpio at index 0: -2 rt5670 i2c-10EC5672:00: Adding jack GPIO failed For fixing this, we need to provide the GPIO mapping explicitly in the machine driver. Also this patch corrects the string to be passed to gpiolib to match with the pre-given mapping, too. Fixes: f10e4bf6632b ("gpio: acpi: Even more tighten up ACPI GPIO lookups") Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=115531 Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/rt5670.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 10 ++++++++++ 2 files changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0ec7985ed306..054b613cb0d0 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -567,7 +567,7 @@ int rt5670_set_jack_detect(struct snd_soc_codec *codec, rt5670->jack = jack; rt5670->hp_gpio.gpiod_dev = codec->dev; - rt5670->hp_gpio.name = "headphone detect"; + rt5670->hp_gpio.name = "headset"; rt5670->hp_gpio.report = SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2; rt5670->hp_gpio.debounce_time = 150; diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index bc2a52de06a3..f597d5582223 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -184,6 +184,13 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, return 0; } +static const struct acpi_gpio_params headset_gpios = { 0, 0, false }; + +static const struct acpi_gpio_mapping cht_rt5672_gpios[] = { + { "headset-gpios", &headset_gpios, 1 }, + {}, +}; + static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { int ret; @@ -191,6 +198,9 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_codec *codec = codec_dai->codec; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); + if (devm_acpi_dev_add_driver_gpios(codec->dev, cht_rt5672_gpios)) + dev_warn(runtime->dev, "Unable to add GPIO mapping table\n"); + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); if (ret < 0) { -- 2.14.0

2 3

[alsa-devel] [PATCH] ALSA: core: Fix unexpected error at replacing user TLV
by Takashi Iwai 22 Aug '17

22 Aug '17

When user tries to replace the user-defined control TLV, the kernel checks the change of its content via memcmp(). The problem is that the kernel passes the return value from memcmp() as is. memcmp() gives a non-zero negative value depending on the comparison result, and this shall be recognized as an error code. The patch covers that corner-case, return 1 properly for the changed TLV. Fixes: 8aa9b586e420 ("[ALSA] Control API - more robust TLV implementation") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index 3c6be1452e35..4525e127afd9 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1137,7 +1137,7 @@ static int snd_ctl_elem_user_tlv(struct snd_kcontrol *kcontrol, mutex_lock(&ue->card->user_ctl_lock); change = ue->tlv_data_size != size; if (!change) - change = memcmp(ue->tlv_data, new_data, size); + change = memcmp(ue->tlv_data, new_data, size) != 0; kfree(ue->tlv_data); ue->tlv_data = new_data; ue->tlv_data_size = size; -- 2.14.0

1 0

[alsa-devel] usb 1-2: 1:1: cannot set freq 44100 to ep 0x81 and potential fix
by Joakim Tjernlund 22 Aug '17

22 Aug '17

I have two USB headsets (Logitech Inc Logitech Stereo H650e and Jabra Jabra UC VOICE 550a) who both causes lots of: usb 1-2: 1:1: cannot set freq 44100 to ep 0x81 errors when using them for pidgin audio chats. The actual freq varies though. I found that adding a small delay here: diff -u sound/usb/clock.c.org sound/usb/clock.c --- sound/usb/clock.c.org 2017-08-18 14:38:05.597977669 +0200 +++ sound/usb/clock.c 2017-08-18 14:20:14.653661513 +0200 @@ -304,6 +304,7 @@ iface, fmt->altsetting, rate, ep); return err; } + msleep(1); /* Don't check the sample rate for devices which we know don't * support reading */ it seems one cannot do these settings too fast Ant better idea for a fix? kernel 4.9.44 Jocke PS. When starting a chat with pidgin-sipe, I see the same sample rate freq setting twice, not sure where that comes from.

2 2

[alsa-devel] [PATCH v4 0/9] ASoC: rockchip: Parse dai links from dts
by Jeffy Chen 22 Aug '17

22 Aug '17

Currently we are using a fixed list of dai links in the driver. This serial of patches would let the driver parse dai links from dts, so that we can make some of them optional for future boards. Tested on my chromebook bob(with cros 4.4 kernel), it still works after disabled rt5514 codecs in the dts. Changes in v4: Use non-zero drv id to avoid legacy dai naming instead of switching to snd_soc_register_codec. Changes in v3: Use compatible to match audio codecs -- Suggested-by Matthias Kaehlcke <mka(a)chromium.org> Changes in v2: Let rockchip,codec-names be a required property, because we plan to add more supported codecs to the fixed dai link list in the driver. Jeffy Chen (9): ASoC: rt5514: Avoid legacy dai naming ASoC: rockchip: Remove obsolete dmic-delay ASoC: rockchip: Use codec of_node and dai_name for rt5514 dsp arm64: dts: rockchip: Add rt5514 dsp for Gru arm64: dts: rockchip: Update rt5514 devices' compatible for Gru ASoC: rockchip: Parse dai links from dts ASoC: rockchip: Add support for DP codec ASoC: rockchip: Add support for DMIC codec dt-bindings: ASoC: rockchip: Update description of rockchip,codec .../bindings/sound/rockchip,rk3399-gru-sound.txt | 9 +- arch/arm64/boot/dts/rockchip/rk3399-gru.dtsi | 7 +- sound/soc/codecs/rt5514-spi.c | 2 +- sound/soc/rockchip/Kconfig | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 263 ++++++++++++++------- 5 files changed, 183 insertions(+), 100 deletions(-) -- 2.11.0

4 12

[alsa-devel] [PATCH v2] ASoC: rsnd: remove unused rsnd_xxx_to_dma()
by Kuninori Morimoto 22 Aug '17

22 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - based on latest mark/topic/rcar b65cb7a590274bf5dbf5bbccbbd0c495c27ab14d ("ASoC: rsnd: tidyup comments position/space/tab") sound/soc/sh/rcar/src.c | 1 - sound/soc/sh/rcar/ssi.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 4640039..510b68a 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -27,7 +27,6 @@ struct rsnd_src { #define RSND_SRC_NAME_SIZE 16 #define rsnd_src_get(priv, id) ((struct rsnd_src *)(priv->src) + id) -#define rsnd_src_to_dma(src) ((src)->dma) #define rsnd_src_nr(priv) ((priv)->src_nr) #define rsnd_src_sync_is_enabled(mod) (rsnd_mod_to_src(mod)->sen.val) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 8bb4714..fffc07e 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -99,7 +99,6 @@ struct rsnd_ssi { i++) #define rsnd_ssi_get(priv, id) ((struct rsnd_ssi *)(priv->ssi) + id) -#define rsnd_ssi_to_dma(mod) ((ssi)->dma) #define rsnd_ssi_nr(priv) ((priv)->ssi_nr) #define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod) #define rsnd_ssi_flags_has(p, f) ((p)->flags & f) -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: simple-scu-card: Parse off codec widgets
by Daniel Baluta 22 Aug '17

22 Aug '17

Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- Documentation/devicetree/bindings/sound/simple-scu-card.txt | 1 + sound/soc/generic/simple-scu-card.c | 4 ++++ 2 files changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-scu-card.txt b/Documentation/devicetree/bindings/sound/simple-scu-card.txt index 327d229..32f8dbc 100644 --- a/Documentation/devicetree/bindings/sound/simple-scu-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-scu-card.txt @@ -24,6 +24,7 @@ Optional subnode properties: - simple-audio-card,convert-rate : platform specified sampling rate convert - simple-audio-card,convert-channels : platform specified converted channel size (2 - 8 ch) - simple-audio-card,prefix : see routing +- simple-audio-card,widgets : Please refer to widgets.txt. - simple-audio-card,routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the connection's sink, the second being the connection's source. Valid names for sources. diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index a75b385..0bb3da7 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -191,6 +191,10 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) if (!node) return -EINVAL; + ret = asoc_simple_card_of_parse_widgets(card, PREFIX); + if (ret < 0) + return ret; + ret = asoc_simple_card_of_parse_routing(card, PREFIX, 0); if (ret < 0) return ret; -- 2.7.4

3 4

[alsa-devel] [PATCH v4 0/3] ASoC: codecs: msm8916-wcd-analog: Add support to MBHC
by srinivas.kandagatla＠linaro.org 21 Aug '17

21 Aug '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset adds support to MBHC(Multibutton headset control) block in PM8916 analog block. MBHC support comes from 2 blocks first mechanical headset detection and second headset type, 5 button detection. This patchset adds support to: 1> Support to NC and NO type of headset Jacks. 2> Mechanical insertion and detection of headset jack. 3> Detect a 3 pole Headphone and a 4 pole Headset. 4> Detect 5 buttons. Tested it on DB410c with Audio Mezz board with 4 pole and 3 pole and during playback and recording use cases. Changes since v3(https://lkml.org/lkml/2017/8/2/609): - dropped ASoC: codecs: msm8916-wcd-analog: move codec reset to probe - fixed rampup time for micbias spotted by Mark. - Added tested-by from Dimien - Dropped dt patch from this series. Srinivas Kandagatla (3): ASoC: codecs: msm8916-wcd-analog: get micbias voltage from dt ASoC: codecs: msm8916-wcd-analog: add MBHC support ASoC: qcom: apq8016-sbc: Add support to Headset JACK .../bindings/sound/qcom,msm8916-wcd-analog.txt | 18 +- sound/soc/codecs/msm8916-wcd-analog.c | 400 ++++++++++++++++++++- sound/soc/qcom/apq8016_sbc.c | 34 ++ 3 files changed, 444 insertions(+), 8 deletions(-) -- 2.11.0

2 6

[alsa-devel] [PATCH v2 0/6] ASoC: codecs: msm8916-wcd-analog: Add support to MBHC
by srinivas.kandagatla＠linaro.org 21 Aug '17

21 Aug '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset adds support to MBHC(Multibutton headset control) block in PM8916 analog block. MBHC support comes from2 blocks first mechanical headset detection and second headset type, 5 button detection. This patchset adds support to: 1> Support to NC and NO type of headset Jacks. 2> Mechanical insertion and detection of headset jack. 3> Detect a 3 pole Headphone and a 4 pole Headset. 4> Detect 5 buttons. Damien sent a similar patchset to add support to mechanical detection, but that patch has issues and will not work on most usecases (for example after a playback/capture session, multicodec case). So I only picked up the BIT mask patch from that series. Tested it on DB410c with Audio Mezz board with 4 pole and 3 pole and during playback and recording use cases. Changes since v1(https://www.spinics.net/lists/kernel/msg2565463.html): - Micbias left to hardware default value suggested by Mark. - Updated commit log to reflect low power state for mbhc - cleaned up code spotted by Mark. - Moved the button voltage threshold to dt, suggested by Mark - Fixed key release issue reported by Damien Srinivas Kandagatla (6): ASoC: jack: fix snd_soc_codec_set_jack return error ASoC: codecs: msm8916-wcd-analog: move codec reset to probe ASoC: codecs: msm8916-wcd-analog: get micbias voltage from dt ASoC: codecs: msm8916-wcd-analog: add MBHC support ASoC: qcom: apq8016-sbc: Add support to Headset JACK arm64: dts: apq8016-sbc: add mbhc buttons support .../bindings/sound/qcom,msm8916-wcd-analog.txt | 18 +- arch/arm64/boot/dts/qcom/apq8016-sbc.dtsi | 2 + sound/soc/codecs/msm8916-wcd-analog.c | 412 +++++++++++++++++++-- sound/soc/qcom/apq8016_sbc.c | 34 ++ sound/soc/soc-jack.c | 2 +- 5 files changed, 444 insertions(+), 24 deletions(-) -- 2.9.3

4 11

[alsa-devel] [PATCH 03/12] ASoC: Intel: Remove unnecessary function call
by Peng Donglin 21 Aug '17

21 Aug '17

First of all,the address of pdev->dev is assigned to card->dev, when calling snd_soc_register_card,the function dev_set_drvdata(card->dev, card) copies the value the variable card to card->dev->driver_data.After that,the function platform_set_drvdata does the same copy operation, so i think that the latter copy operation can be removed. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/intel/boards/bytcht_da7213.c | 1 - sound/soc/intel/boards/bytcht_es8316.c | 1 - sound/soc/intel/boards/bytcr_rt5651.c | 1 - sound/soc/intel/boards/cht_bsw_max98090_ti.c | 1 - sound/soc/intel/boards/cht_bsw_rt5645.c | 1 - sound/soc/intel/boards/cht_bsw_rt5672.c | 1 - 6 files changed, 6 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index 18873e2..fbc870b 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -265,7 +265,6 @@ static int bytcht_da7213_probe(struct platform_device *pdev) "snd_soc_register_card failed %d\n", ret_val); return ret_val; } - platform_set_drvdata(pdev, card); return ret_val; } diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 5263546..5e6e8e9 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -282,7 +282,6 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); return ret; } - platform_set_drvdata(pdev, &byt_cht_es8316_card); return ret; } diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 4a3516b..77702aa 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -310,7 +310,6 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) ret_val); return ret_val; } - platform_set_drvdata(pdev, &byt_rt5651_card); return ret_val; } diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 20755ec..8583c0c 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -300,7 +300,6 @@ static int snd_cht_mc_probe(struct platform_device *pdev) "snd_soc_register_card failed %d\n", ret_val); return ret_val; } - platform_set_drvdata(pdev, &snd_soc_card_cht); return ret_val; } diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 5bcde01..0e6a24c 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -699,7 +699,6 @@ static int snd_cht_mc_probe(struct platform_device *pdev) "snd_soc_register_card failed %d\n", ret_val); return ret_val; } - platform_set_drvdata(pdev, card); return ret_val; } diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index bc2a52d..e8bd22b 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -448,7 +448,6 @@ static int snd_cht_mc_probe(struct platform_device *pdev) "snd_soc_register_card failed %d\n", ret_val); return ret_val; } - platform_set_drvdata(pdev, &snd_soc_card_cht); return ret_val; } -- 1.9.1

2 1

[alsa-devel] [PATCH 12/12] ASoC: ux500: Remove unnecessary function call
by Peng Donglin 21 Aug '17

21 Aug '17

First of all,the address of pdev->dev is assigned to mop500_card.dev, then the function platform_set_drvdata copies the value the variable card to pdev->dev.driver_data,but when calling snd_soc_register_card, the function dev_set_drvdata(card->dev, card) will also do the same copy operation,so i think that the former copy operation can be removed. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/ux500/mop500.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index ba9fc09..070a688 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -115,7 +115,6 @@ static int mop500_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n", __func__, mop500_card.name); - platform_set_drvdata(pdev, &mop500_card); snd_soc_card_set_drvdata(&mop500_card, NULL); -- 1.9.1

2 1

[alsa-devel] [PATCH 02/12] ASoC: mediatek: Remove unnecessary function call
by Peng Donglin 21 Aug '17

21 Aug '17

First of all,the address of pdev->dev is assigned to card->dev,then the function platform_set_drvdata copies the value the variable card to pdev->dev.driver_data, but when calling snd_soc_register_card,the function dev_set_drvdata(card->dev, card) will also do the same copy operation,so i think that the former copy operation can be removed. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650.c | 1 - 3 files changed, 3 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 5e383eb..99c1521 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -222,7 +222,6 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) mt8173_rt5650_rt5514_codecs[1].of_node; card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index fed1f15..42de84c 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -279,7 +279,6 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) } card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index a784708..e69c141 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -311,7 +311,6 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) return -EINVAL; } card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) -- 1.9.1

2 1

[alsa-devel] [PATCH 11/12] ASoC: tegra: Remove unnecessary function call
by Peng Donglin 21 Aug '17

21 Aug '17

First of all,the address of pdev->dev is assigned to card->dev,then the function platform_set_drvdata copies the value the variable card to pdev->dev.driver_data, but when calling snd_soc_register_card,the function dev_set_drvdata(card->dev, card) will also do the same copy operation,so i think that the former copy operation can be removed. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/tegra/tegra_alc5632.c | 1 - sound/soc/tegra/tegra_max98090.c | 1 - sound/soc/tegra/tegra_rt5640.c | 1 - sound/soc/tegra/tegra_rt5677.c | 1 - sound/soc/tegra/tegra_sgtl5000.c | 1 - sound/soc/tegra/tegra_wm8753.c | 1 - sound/soc/tegra/tegra_wm8903.c | 1 - sound/soc/tegra/tegra_wm9712.c | 1 - sound/soc/tegra/trimslice.c | 1 - 9 files changed, 9 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 0509902..053fe4c 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -173,7 +173,6 @@ static int tegra_alc5632_probe(struct platform_device *pdev) return -ENOMEM; card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, alc5632); alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index c34a54d..41feee9 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -229,7 +229,6 @@ static int tegra_max98090_probe(struct platform_device *pdev) return -ENOMEM; card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 93a3568..02f27fe 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -174,7 +174,6 @@ static int tegra_rt5640_probe(struct platform_device *pdev) return -ENOMEM; card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); machine->gpio_hp_det = of_get_named_gpio_flags( diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index ebf58d0..3dbf0e1 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -222,7 +222,6 @@ static int tegra_rt5677_probe(struct platform_device *pdev) return -ENOMEM; card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index 6dda01f..45a4aa9 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -124,7 +124,6 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) return -ENOMEM; card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); ret = snd_soc_of_parse_card_name(card, "nvidia,model"); diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index d0ab0026..23a810e 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -132,7 +132,6 @@ static int tegra_wm8753_driver_probe(struct platform_device *pdev) return -ENOMEM; card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); ret = snd_soc_of_parse_card_name(card, "nvidia,model"); diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index dbfb492..d4a4a2b 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -252,7 +252,6 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev) return -ENOMEM; card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); machine->gpio_spkr_en = of_get_named_gpio(np, "nvidia,spkr-en-gpios", diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index c9cd224..864a334 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -81,7 +81,6 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev) return -ENOMEM; card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); machine->codec = platform_device_alloc("wm9712-codec", -1); diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index c9dcad9..99bcdd9 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -127,7 +127,6 @@ static int tegra_snd_trimslice_probe(struct platform_device *pdev) return -ENOMEM; card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, trimslice); trimslice_tlv320aic23_dai.codec_of_node = of_parse_phandle(np, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl-asoc-card: don't print EPROBE_DEFER as error
by Lucas Stach 21 Aug '17

21 Aug '17

Probe deferral is to be expected during normal operation, so avoid printing an error when it is encountered. Signed-off-by: Lucas Stach <l.stach(a)pengutronix.de> --- sound/soc/fsl/fsl-asoc-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 9998aea23597..2db4d0c80d33 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -683,7 +683,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(&priv->card, priv); ret = devm_snd_soc_register_card(&pdev->dev, &priv->card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); asrc_fail: -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: dapm: Avoid creating kcontrol for params
by yesanishhere＠gmail.com 21 Aug '17

21 Aug '17

From: anish kumar <yesanishhere(a)gmail.com> When num_params is defined as one in case of codec to codec dai link, there is no point in allocating memory for kcontrol. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- include/sound/soc-dapm.h | 1 - sound/soc/soc-dapm.c | 147 ++++++++++++++++++++++++----------------------- 2 files changed, 76 insertions(+), 72 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 344b96c..dc5c336 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -582,7 +582,6 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ - unsigned int num_params; /* number of params for dai links */ unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcef67a..0a26905 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1032,10 +1032,6 @@ static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) struct snd_soc_dapm_context *dapm = w->dapm; struct snd_card *card = dapm->card->snd_card; - /* create control for links with > 1 config */ - if (w->num_params <= 1) - return 0; - /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, @@ -3770,108 +3766,125 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, if (ucontrol->value.enumerated.item[0] == w->params_select) return 0; - if (ucontrol->value.enumerated.item[0] >= w->num_params) - return -EINVAL; - w->params_select = ucontrol->value.enumerated.item[0]; return 0; } -int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, - unsigned int num_params, - struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) +struct snd_kcontrol_new *snd_soc_dapm_new_kcontrol(struct snd_soc_card *card, + char *link_name, + const struct snd_soc_pcm_stream *params, + int num_params) { - struct snd_soc_dapm_widget template; - struct snd_soc_dapm_widget *w; - char *link_name; - int ret, count; - unsigned long private_value; const char **w_param_text; + int count; + unsigned long private_value; struct soc_enum w_param_enum[] = { SOC_ENUM_SINGLE(0, 0, 0, NULL), }; struct snd_kcontrol_new kcontrol_dai_link[] = { SOC_ENUM_EXT(NULL, w_param_enum[0], - snd_soc_dapm_dai_link_get, - snd_soc_dapm_dai_link_put), + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), }; + struct snd_kcontrol_new *kcontrol_news = NULL; const struct snd_soc_pcm_stream *config = params; w_param_text = devm_kcalloc(card->dev, num_params, sizeof(char *), GFP_KERNEL); if (!w_param_text) - return -ENOMEM; - - link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", - source->name, sink->name); - if (!link_name) { - ret = -ENOMEM; - goto outfree_w_param; - } + return NULL; for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { dev_warn(card->dapm.dev, - "ASoC: anonymous config %d for dai link %s\n", - count, link_name); + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); w_param_text[count] = devm_kasprintf(card->dev, GFP_KERNEL, - "Anonymous Configuration %d", - count); - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + "Anonymous Configuration %d", + count); } else { w_param_text[count] = devm_kmemdup(card->dev, - config->stream_name, - strlen(config->stream_name) + 1, - GFP_KERNEL); - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + config->stream_name, + strlen(config->stream_name) + 1, + GFP_KERNEL); } + if (!w_param_text[count]) + goto outfree_w_param; config++; } + w_param_enum[0].items = num_params; w_param_enum[0].texts = w_param_text; - memset(&template, 0, sizeof(template)); - template.reg = SND_SOC_NOPM; - template.id = snd_soc_dapm_dai_link; - template.name = link_name; - template.event = snd_soc_dai_link_event; - template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD; - template.num_kcontrols = 1; - /* duplicate w_param_enum on heap so that memory persists */ private_value = (unsigned long) devm_kmemdup(card->dev, - (void *)(kcontrol_dai_link[0].private_value), - sizeof(struct soc_enum), GFP_KERNEL); + (void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); if (!private_value) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", - link_name); - ret = -ENOMEM; - goto outfree_link_name; + link_name); + goto outfree_w_param; } kcontrol_dai_link[0].private_value = private_value; /* duplicate kcontrol_dai_link on heap so that memory persists */ - template.kcontrol_news = - devm_kmemdup(card->dev, &kcontrol_dai_link[0], - sizeof(struct snd_kcontrol_new), - GFP_KERNEL); - if (!template.kcontrol_news) { + kcontrol_news = + devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + if (!kcontrol_news) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", - link_name); - ret = -ENOMEM; + link_name); goto outfree_private_value; } + return kcontrol_news; + +outfree_private_value: + devm_kfree(card->dev, (void *)private_value); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + if (w_param_text[count] != NULL) + devm_kfree(card->dev, (void *)w_param_text[count]); + devm_kfree(card->dev, w_param_text); + return NULL; +} + +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + unsigned int num_params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + char *link_name; + int ret; + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); + if (!link_name) + return -ENOMEM; + + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + template.id = snd_soc_dapm_dai_link; + template.name = link_name; + template.event = snd_soc_dai_link_event; + template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD; + template.kcontrol_news = NULL; + + /* allocate memory for control, only in case of multiple configs */ + if (num_params > 1) { + template.num_kcontrols = 1; + template.kcontrol_news = + snd_soc_dapm_new_kcontrol(card, link_name, params, num_params); + if (!template.kcontrol_news) { + ret = -ENOMEM; + goto outfree_link_name; + } + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); @@ -3892,7 +3905,6 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, } w->params = params; - w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) @@ -3903,15 +3915,8 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, w); outfree_kcontrol_news: devm_kfree(card->dev, (void *)template.kcontrol_news); -outfree_private_value: - devm_kfree(card->dev, (void *)private_value); outfree_link_name: devm_kfree(card->dev, link_name); -outfree_w_param: - for (count = 0 ; count < num_params; count++) - devm_kfree(card->dev, (void *)w_param_text[count]); - devm_kfree(card->dev, w_param_text); - return ret; } -- 2.5.4 (Apple Git-61)

2 1

[alsa-devel] [PATCH 0/2] ALSA: firewire-motu: add support for AudioExpress
by Takashi Sakamoto 21 Aug '17

21 Aug '17

Hi, This patchset is to add support for MOTU AudioExpress. Any PCM frames and MIDI messages are transferred in rx/tx isochronous packets. Correspondingly, this adds an arrangement for specification flag for position of existence flag of MIDI messages. I note that this unit has a quirk of acknowledge code against request subaction on asynchronous communication. In detail, please refer to 2nd patch of this set. To easy my development, I applied a patch to 'firewire-ohci' module on my develop ment tree. You can see the content in the end of this message. It's really a band-aid for the aim and can not be merged as what it is. Unfortunately, current implementations of IEC 61883-1/6 packet streaming engine is not good enough in a point to presentation timestamp, as I reported[1]. Any plaback stream for the unit still causes periodical noise. [1] [alsa-devel] Status of ALSA firewire stack at Linux v4.12 development period http://mailman.alsa-project.org/pipermail/alsa-devel/2017-April/120061.html -------- 8< -------- diff --git a/drivers/firewire/ohci.c b/drivers/firewire/ohci.c index 8bf89267dc25..06a0906a9b56 100644 --- a/drivers/firewire/ohci.c +++ b/drivers/firewire/ohci.c @@ -1499,6 +1499,10 @@ static int handle_at_packet(struct context *context, packet->ack = evt - 0x10; break; + case 0x10: + packet->ack = ACK_PENDING; + break; + case OHCI1394_evt_no_status: if (context->flushing) { packet->ack = RCODE_GENERATION; -------- 8< -------- Takashi Sakamoto (2): ALSA: firewire-motu: add specification flag for position of flag for MIDI messages ALSA: firewire-motu: add support for MOTU Audio Express sound/firewire/motu/motu-pcm.c | 2 +- sound/firewire/motu/motu-protocol-v2.c | 5 ----- sound/firewire/motu/motu-protocol-v3.c | 5 ----- sound/firewire/motu/motu-stream.c | 38 ++++++++++++++++++++++++++++++++-- sound/firewire/motu/motu.c | 24 ++++++++++++++++++--- sound/firewire/motu/motu.h | 6 +++++- 6 files changed, 63 insertions(+), 17 deletions(-) -- 2.11.0

2 3

[alsa-devel] [PATCH 09/12] ASoC: qcom: Remove unnecessary function call
by Peng Donglin 21 Aug '17

21 Aug '17

First of all,the address of pdev->dev is assigned to card->dev,then the function platform_set_drvdata copies the value the variable card to pdev->dev.driver_data, but when calling snd_soc_register_card,the function dev_set_drvdata(card->dev, card) will also do the same copy operation,so i think that the former copy operation can be removed. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/qcom/storm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c index c5207af..a9fa972 100644 --- a/sound/soc/qcom/storm.c +++ b/sound/soc/qcom/storm.c @@ -99,7 +99,6 @@ static int storm_platform_probe(struct platform_device *pdev) return -ENOMEM; card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); ret = snd_soc_of_parse_card_name(card, "qcom,model"); if (ret) { -- 1.9.1

2 1

[alsa-devel] [PATCH 08/12] ASoC: qcom: Remove useless function call
by Peng Donglin 21 Aug '17

21 Aug '17

The function platform_set_drvdata(pdev, data) copies the value of the variable data to pdev->dev.driver_data,but when calling snd_soc_register_card,the function dev_set_drvdata(card->dev, card) will override it, so i think that the former copy operation is useless and can be removed. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/qcom/apq8016_sbc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index d084d74..5292591 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -191,7 +191,6 @@ static int apq8016_sbc_platform_probe(struct platform_device *pdev) if (IS_ERR(data->spkr_iomux)) return PTR_ERR(data->spkr_iomux); - platform_set_drvdata(pdev, data); snd_soc_card_set_drvdata(card, data); return devm_snd_soc_register_card(&pdev->dev, card); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: tidyup comments position/space/tab
by Kuninori Morimoto 21 Aug '17

21 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 2 +- sound/soc/sh/rcar/src.c | 7 +++---- 2 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 362b307..17220c9 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -604,8 +604,8 @@ static int rsnd_dmapp_attach(struct rsnd_dai_stream *io, dma_addr_t in_addr; } dma_addrs[3][2][3] = { /* SRC */ + /* Capture */ {{{ 0, 0 }, - /* Capture */ { RDMA_SRC_O_N(src, id), RDMA_SRC_I_P(src, id) }, { RDMA_CMD_O_N(src, id), RDMA_SRC_I_P(src, id) } }, /* Playback */ diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index f67365d..510b68a 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -107,7 +107,6 @@ unsigned int rsnd_src_get_rate(struct rsnd_priv *priv, int is_play = rsnd_io_is_play(io); /* - * * Playback * runtime_rate -> [SRC] -> convert_rate * @@ -202,13 +201,13 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, use_src = (fin != fout) | rsnd_src_sync_is_enabled(mod); /* - * SRC_ADINR + * SRC_ADINR */ adinr = rsnd_get_adinr_bit(mod, io) | rsnd_runtime_channel_original(io); /* - * SRC_IFSCR / SRC_IFSVR + * SRC_IFSCR / SRC_IFSVR */ ifscr = 0; fsrate = 0; @@ -222,7 +221,7 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, } /* - * SRC_SRCCR / SRC_ROUTE_MODE0 + * SRC_SRCCR / SRC_ROUTE_MODE0 */ cr = 0x00011110; route = 0x0; -- 1.9.1

1 0

[alsa-devel] [PATCH] input/keyboard: fix platform_no_drv_owner.cocci warnings
by Julia Lawall 20 Aug '17

20 Aug '17

No need to set .owner here. The core will do it. Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Fixes: 52cf590518d7 ("input/keyboard: Add support for Dollar Cove TI power button") Signed-off-by: Julia Lawall <julia.lawall(a)lip6.fr> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- dc_ti_pwrbtn.c | 1 - 1 file changed, 1 deletion(-) --- a/drivers/input/keyboard/dc_ti_pwrbtn.c +++ b/drivers/input/keyboard/dc_ti_pwrbtn.c @@ -75,7 +75,6 @@ static int dc_ti_pwrbtn_probe(struct pla static struct platform_driver dc_ti_pwrbtn_driver = { .driver = { .name = DRIVER_NAME, - .owner = THIS_MODULE, }, .probe = dc_ti_pwrbtn_probe, };

3 3

[alsa-devel] [PATCH] ALSA: firewire-motu: destroy stream data surely at failure of card initialization
by Takashi Sakamoto 20 Aug '17

20 Aug '17

When failing sound card registration after initializing stream data, this module leaves allocated data in stream data. This commit fixes the bug. Fixes: 9b2bb4f2f4a2 ('ALSA: firewire-motu: add stream management functionality') Cc: <stable(a)vger.kernel.org> # v4.12+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index bf779cfeef0d..59a270406353 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -128,6 +128,7 @@ static void do_registration(struct work_struct *work) return; error: snd_motu_transaction_unregister(motu); + snd_motu_stream_destroy_duplex(motu); snd_card_free(motu->card); dev_info(&motu->unit->device, "Sound card registration failed: %d\n", err); -- 2.11.0

2 1

[alsa-devel] [PATCH] ALSA: firewire: fix NULL pointer dereference when releasing uninitialized data of iso-resource
by Takashi Sakamoto 20 Aug '17

20 Aug '17

When calling 'iso_resource_free()' for uninitialized data, this function causes NULL pointer dereference due to its 'unit' member. This occurs when unplugging audio and music units on IEEE 1394 bus at failure of card registration. This commit fixes the bug. The bug exists since kernel v4.5. Fixes: 324540c4e05c ('ALSA: fireface: postpone sound card registration') at v4.12 Fixes: 8865a31e0fd8 ('ALSA: firewire-motu: postpone sound card registration') at v4.12 Fixes: b610386c8afb ('ALSA: firewire-tascam: deleyed registration of sound card') at v4.7 Fixes: 86c8dd7f4da3 ('ALSA: firewire-digi00x: delayed registration of sound card') at v4.7 Fixes: 6c29230e2a5f ('ALSA: oxfw: delayed registration of sound card') at v4.7 Fixes: 7d3c1d5901aa ('ALSA: fireworks: delayed registration of sound card') at v4.7 Fixes: 04a2c73c97eb ('ALSA: bebob: delayed registration of sound card') at v4.7 Fixes: b59fb1900b4f ('ALSA: dice: postpone card registration') at v4.5 Cc: <stable(a)vger.kernel.org> # v4.5+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/iso-resources.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/firewire/iso-resources.c b/sound/firewire/iso-resources.c index f0e4d502d604..066b5df666f4 100644 --- a/sound/firewire/iso-resources.c +++ b/sound/firewire/iso-resources.c @@ -210,9 +210,14 @@ EXPORT_SYMBOL(fw_iso_resources_update); */ void fw_iso_resources_free(struct fw_iso_resources *r) { - struct fw_card *card = fw_parent_device(r->unit)->card; + struct fw_card *card; int bandwidth, channel; + /* Not initialized. */ + if (r->unit == NULL) + return; + card = fw_parent_device(r->unit)->card; + mutex_lock(&r->mutex); if (r->allocated) { -- 2.11.0

2 1

[alsa-devel] [PATCH 0/3] ALSA: control: fix issue of concurrent access of ELEM_READ/ELEM_WRITE
by Takashi Sakamoto 20 Aug '17

20 Aug '17

Hi, This patchset is to fix an issue of concurrent access to operate ELEM_READ/ELEM_WRITE, which I found in my previous patch: [PATCH 0/5] ALSA: control: fix issue of concurrent access for TLV operations http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/123561.html In current implementation of ALSA control core, concurrent accesses with ELEM_READ/ELEM_WRITE are managed with counting semaphore, however lock acquisition for both of these operations is a type of 'read'. In this case, ELEM_READ can be going to be handled for a target element when ELEM_WRITE is handled for the element. A purpose of the ELEM_WRITE operation is to change state of the target element. It's better to guarantee that concurrent accesses to the same element with these two operations are properly managed. For the above aim, this patchset uses the counting semaphore as 'write' lock for ELEM_WRITE operation. But I can imagine an discussion to add 'per-element' lock for this issue. In a comment of patch 03, I investigate it. Please refer to it. Takashi Sakamoto (3): ALSA: control: queue events within locking of controls_rwsem for ELEM_WRITE operation ALSA: control: code refactoring for ELEM_READ/ELEM_WRITE operations ALSA: control: use counting semaphore as write lock for ELEM_WRITE operation sound/core/control.c | 78 +++++++++++++++++++++++++--------------------------- 1 file changed, 38 insertions(+), 40 deletions(-) -- 2.11.0

2 4

[alsa-devel] [PATCH 10/12] ASoC: sun4i-codec: Remove unnecessary function call
by Peng Donglin 20 Aug '17

20 Aug '17

First of all,the address of pdev->dev is assigned to card->dev in create_card,then the function platform_set_drvdata copies the value the variable card to pdev->dev.driver_data, but when calling snd_soc_register_card,the function dev_set_drvdata(card->dev, card) will also do the same copy operation,so i think that the former copy operation can be removed. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/sunxi/sun4i-codec.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 1500699..82beeae 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1655,7 +1655,6 @@ static int sun4i_codec_probe(struct platform_device *pdev) goto err_unregister_codec; } - platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, scodec); ret = snd_soc_register_card(card); -- 1.9.1

1 0

[alsa-devel] [PATCH 07/12] ASoC: mxs-sgtl5000: Remove unnecessary function call
by Peng Donglin 20 Aug '17

20 Aug '17

First of all,the address of pdev->dev is assigned to card->dev,then the function platform_set_drvdata copies the value the variable card to pdev->dev.driver_data, but when calling snd_soc_register_card,the function dev_set_drvdata(card->dev, card) will also do the same copy operation,so i think that the former copy operation can be removed. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/mxs/mxs-sgtl5000.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index a96276e..2ed3240 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -140,7 +140,6 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev) } card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { -- 1.9.1

1 0

[alsa-devel] [PATCH 06/12] ASoC: rockchip: Remove unnecessary function call
by Peng Donglin 20 Aug '17

20 Aug '17

First of all,the address of pdev->dev is assigned to card->dev,then the function platform_set_drvdata copies the value the variable card to pdev->dev.driver_data, but when calling snd_soc_register_card,the function dev_set_drvdata(card->dev, card) will also do the same copy operation,so i think that the former copy operation can be removed. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/rockchip/rk3288_hdmi_analog.c | 2 -- sound/soc/rockchip/rk3399_gru_sound.c | 1 - 2 files changed, 3 deletions(-) diff --git a/sound/soc/rockchip/rk3288_hdmi_analog.c b/sound/soc/rockchip/rk3288_hdmi_analog.c index dbc53e4..37ef162 100644 --- a/sound/soc/rockchip/rk3288_hdmi_analog.c +++ b/sound/soc/rockchip/rk3288_hdmi_analog.c @@ -272,8 +272,6 @@ static int snd_rk_mc_probe(struct platform_device *pdev) return ret; } - platform_set_drvdata(pdev, card); - return ret; } diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 3475c61..70c0908 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -362,7 +362,6 @@ static int rockchip_sound_probe(struct platform_device *pdev) rockchip_dailinks[DAILINK_RT5514_DSP].platform_name = kstrdup_const(dev_name(dev), GFP_KERNEL); card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) -- 1.9.1

1 0

[alsa-devel] [PATCH 05/12] ASoC: atmel: Remove unnecessary function call
by Peng Donglin 20 Aug '17

20 Aug '17

The function platform_set_drvdata copies the value the variable card to card->dev->driver_data, then the address of &pdev->dev is assigned to card->dev in atmel_pdmic_asoc_card_init, but when calling snd_soc_register_card,the function dev_set_drvdata(card->dev, card) will do the same copy operation, so i think that the former copy operation can be removed. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/atmel/atmel-pdmic.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c index c917df7..8e3d34b 100644 --- a/sound/soc/atmel/atmel-pdmic.c +++ b/sound/soc/atmel/atmel-pdmic.c @@ -694,7 +694,6 @@ static int atmel_pdmic_probe(struct platform_device *pdev) } snd_soc_card_set_drvdata(card, dd); - platform_set_drvdata(pdev, card); ret = atmel_pdmic_asoc_card_init(dev, card); if (ret) { -- 1.9.1

1 0

[alsa-devel] [PATCH 04/12] ASoC: atmel: Remove unnecessary function call
by Peng Donglin 20 Aug '17

20 Aug '17

The function platform_set_drvdata copies the value the variable card to card->dev->driver_data, then the address of &pdev->dev is assigned to card->dev in atmel_classd_asoc_card_init, but when calling snd_soc_register_card,the function dev_set_drvdata(card->dev, card) will do the same copy operation, so i think that the former copy operation can be removed. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/atmel/atmel-classd.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index b7ef8c5..0cd7caa 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -652,7 +652,6 @@ static int atmel_classd_probe(struct platform_device *pdev) } snd_soc_card_set_drvdata(card, dd); - platform_set_drvdata(pdev, card); ret = atmel_classd_asoc_card_init(dev, card); if (ret) { -- 1.9.1

1 0

[alsa-devel] [PATCH 01/12] ASoC: s3c24xx_uda134x: Remove unnecessary function call
by Peng Donglin 20 Aug '17

20 Aug '17

First of all,the address of pdev->dev is assigned to card->dev,then the function platform_set_drvdata copies the value the variable card to pdev->dev.driver_data, but when calling snd_soc_register_card,the function dev_set_drvdata(card->dev, card) will also do the same copy operation,so i think that the former copy operation can be removed. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/samsung/s3c24xx_uda134x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 55538e3..758e46e 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -237,7 +237,6 @@ static int s3c24xx_uda134x_probe(struct platform_device *pdev) mutex_init(&priv->clk_lock); card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, priv); ret = devm_snd_soc_register_card(&pdev->dev, card); -- 1.9.1

1 0

[alsa-devel] [PATCH 00/12] Remove unnecessary function call
by Peng Donglin 20 Aug '17

20 Aug '17

First of all,the address of pdev->dev is assigned to card->dev,then the function platform_set_drvdata copies the value the variable card to pdev->dev.driver_data, but when calling snd_soc_register_card,the function dev_set_drvdata(card->dev, card) will also do the same copy operation,so i think that the former copy operation can be removed. Peng Donglin (12): ASoC: s3c24xx_uda134x: Remove unnecessary function call ASoC: mediatek: Remove unnecessary function call ASoC: Intel: Remove unnecessary function call ASoC: atmel: Remove unnecessary function call ASoC: atmel: Remove unnecessary function call ASoC: rockchip: Remove unnecessary function call ASoC: mxs-sgtl5000: Remove unnecessary function call ASoC: qcom: Remove useless function call ASoC: qcom: Remove unnecessary function call ASoC: sun4i-codec: Remove unnecessary function call ASoC: tegra: Remove unnecessary function call ASoC: ux500: Remove unnecessary function call sound/soc/atmel/atmel-classd.c | 1 - sound/soc/atmel/atmel-pdmic.c | 1 - sound/soc/intel/boards/bytcht_da7213.c | 1 - sound/soc/intel/boards/bytcht_es8316.c | 1 - sound/soc/intel/boards/bytcr_rt5651.c | 1 - sound/soc/intel/boards/cht_bsw_max98090_ti.c | 1 - sound/soc/intel/boards/cht_bsw_rt5645.c | 1 - sound/soc/intel/boards/cht_bsw_rt5672.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650.c | 1 - sound/soc/mxs/mxs-sgtl5000.c | 1 - sound/soc/qcom/apq8016_sbc.c | 1 - sound/soc/qcom/storm.c | 1 - sound/soc/rockchip/rk3288_hdmi_analog.c | 2 -- sound/soc/rockchip/rk3399_gru_sound.c | 1 - sound/soc/samsung/s3c24xx_uda134x.c | 1 - sound/soc/sunxi/sun4i-codec.c | 1 - sound/soc/tegra/tegra_alc5632.c | 1 - sound/soc/tegra/tegra_max98090.c | 1 - sound/soc/tegra/tegra_rt5640.c | 1 - sound/soc/tegra/tegra_rt5677.c | 1 - sound/soc/tegra/tegra_sgtl5000.c | 1 - sound/soc/tegra/tegra_wm8753.c | 1 - sound/soc/tegra/tegra_wm8903.c | 1 - sound/soc/tegra/tegra_wm9712.c | 1 - sound/soc/tegra/trimslice.c | 1 - sound/soc/ux500/mop500.c | 1 - 28 files changed, 29 deletions(-) -- 1.9.1

1 0

[alsa-devel] [alsa-utils] alsaloop falls into loop after SIGCONT
by Dimitrios Semitsoglou-Tsiapos 19 Aug '17

19 Aug '17

Hello everyone, I've ran into some issues while writing a wrapper around alsaloop. It's possible to make it fall into infinite loops of silence. Here is how to trigger one such loop: alsaloop -C cloop -P Audio -t 100000 -v... & # Read attached asoundrc sleep 2 mpc play pkill -STOP -x alsaloop; sleep 0.1; pkill -CONT -x alsaloop On the great majority of times, this happens: ... pool took 68171us playback hw:Audio/capture cloop: pollfds handle playback hw:Audio: delay 2160 / 38400 / 0 capture cloop: delay 15120 / 38400 / 0 playback hw:Audio/capture cloop: prevents = 0x4, crevents = 0x0 playback hw:Audio/capture cloop: queued 5040/18000 samples playback hw:Audio: end delay 5040 / 38400 / 0 capture cloop: end delay 18000 / 38400 / 0 playback hw:Audio/capture cloop: processing time 69us pool took 192670us playback hw:Audio/capture cloop: pollfds handle playback hw:Audio: delay error: Broken pipe / 38400 / 0 capture cloop: delay 8400 / 38400 / 0 playback hw:Audio/capture cloop: prevents = 0xc, crevents = 0x1 underrun for playback hw:Audio playback hw:Audio/capture cloop: xrun sync 0 1 sync: cdelay=18000(18000), pdelay=9600(9600), fill=4800 (delay=27600), src_out=0 sync: cbufcount=9600, pbufcount=9600 sync: capt stop removed 9600 samples playback hw:Audio/capture cloop: xrun sync 0 1 sync: cdelay=18000(18000), pdelay=0(0), fill=4800 (delay=18000), src_out=0 sync: cbufcount=0, pbufcount=0 sync: capt stop removed 0 samples playback hw:Audio/capture cloop: xrun sync 0 1 sync: cdelay=18000(18000), pdelay=0(0), fill=4800 (delay=18000), src_out=0 sync: cbufcount=0, pbufcount=0 sync: capt stop removed 0 samples playback hw:Audio/capture cloop: xrun sync 0 1 ... Moving outwards, some observations, * `remove_samples()` falls into `if (loop->play->buf == loop->capt->buf)`, which is what decides to always remove 0 samples after some point (as seen in output). * cdelay (from `snd_pcm_delay(capt->handle, &cdelay)`) is consistently 18000, meaning that `delay1 > fill` in `xrun_sync()` will always be true as `fill` is fixed during the run. If one forces this check to fail (by setting cdelay to 0), alsaloop recovers, with no audible delay compared to what's being played back. * It is this code that leads to `xrun()` in the first place: if (avail == -EPIPE) { if ((err = xrun(lhandle)) < 0) * The issue is hinted even earlier, as after err = snd_pcm_poll_descriptors_revents(play->handle, fds, play->pollfd_count, &prevents); `(prevents & POLLERR)` is true. * Depending on when STOP is sent, the same sometimes holds true after `thread_job1`'s err = poll(pfds, j, wake); on `pfds[i].revents`. I am not sure what the correct way of fixing this is, especially because I couldn't figure out why `snd_pcm_delay` returns this specific value. I can only guess it is related to aloop. Other ways of throwing alsaloop into infinite loops is unplugging a soundcard that it's using (I believe it should exit in this case) and suspending/resuming the system. I haven't gone through the codepaths in those cases carefully yet, but I'm interested in fixing them eventually. Please let me know if I can provide more useful output, Dimitri [alsa-info.sh]http://www.alsa-project.org/db/?f=89eb0056b0876a3b4fb377c39d5b…

1 0

[alsa-devel] [PATCH 00/13] constify alsa snd_pcm_ops structures
by Arvind Yadav 19 Aug '17

19 Aug '17

snd_pcm_ops are not supposed to change at runtime. All functions working with snd_pcm_ops provided by <sound/pcm.h> work with const snd_pcm_ops. So mark the non-const structs as const. Arvind Yadav (13): [PATCH 01/13] ALSA: aoa: constify snd_pcm_ops structures [PATCH 02/13] ALSA: arm: constify snd_pcm_ops structures [PATCH 03/13] ALSA: atmel: constify snd_pcm_ops structures [PATCH 04/13] ALSA: drivers: constify snd_pcm_ops structures [PATCH 05/13] ALSA: firewire: constify snd_pcm_ops structures [PATCH 06/13] ALSA: mips: constify snd_pcm_ops structures [PATCH 07/13] ALSA: parisc: constify snd_pcm_ops structures [PATCH 08/13] ALSA: pcmcia: constify snd_pcm_ops structures [PATCH 09/13] ALSA: ppc: constify snd_pcm_ops structures [PATCH 10/13] ALSA: sh: constify snd_pcm_ops structures [PATCH 11/13] ALSA: sparc: constify snd_pcm_ops structures [PATCH 12/13] ALSA: spi: constify snd_pcm_ops structures [PATCH 13/13] ALSA: usb: constify snd_pcm_ops structures sound/aoa/soundbus/i2sbus/pcm.c | 4 +- sound/arm/aaci.c | 4 +- sound/arm/pxa2xx-pcm.c | 2 +- sound/atmel/ac97c.c | 4 +- sound/drivers/aloop.c | 4 +- sound/drivers/ml403-ac97cr.c | 4 +- sound/firewire/isight.c | 2 +- sound/firewire/motu/motu-pcm.c | 4 +- sound/mips/hal2.c | 4 +- sound/mips/sgio2audio.c | 6 +- sound/parisc/harmony.c | 4 +- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 2 +- sound/ppc/pmac.c | 4 +- sound/ppc/snd_ps3.c | 2 +- sound/sh/aica.c | 2 +- sound/sh/sh_dac_audio.c | 2 +- sound/sparc/amd7930.c | 4 +- sound/sparc/cs4231.c | 4 +- sound/sparc/dbri.c | 2 +- sound/spi/at73c213.c | 2 +- sound/usb/6fire/pcm.c | 2 +- sound/usb/caiaq/audio.c | 2 +- sound/usb/hiface/pcm.c | 2 +- sound/usb/misc/ua101.c | 4 +- sound/usb/pcm.c | 4 +- sound/usb/usx2y/usbusx2yaudio.c | 2 +- sound/usb/usx2y/usx2yhwdeppcm.c | 2 +- 27 files changed, 42 insertions(+), 42 deletions(-) -- 1.9.1

2 14

[alsa-devel] [PATCH 1/2] ASoC: cs43130: Add support for CS43130 codec
by Li Xu 18 Aug '17

18 Aug '17

Add support for Cirrus Logic CS43130 codec. Support: I2S/DSP PCM playback. DoP/DSD playback. HP detection and DC/AC impedance measurement. Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs43130.c | 2675 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs43130.h | 545 +++++++++ 4 files changed, 3228 insertions(+) create mode 100644 sound/soc/codecs/cs43130.c create mode 100644 sound/soc/codecs/cs43130.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 010811e..1f65336 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -60,6 +60,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_I2C if I2C select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C + select SND_SOC_CS43130 if I2C select SND_SOC_CS4349 if I2C select SND_SOC_CS47L24 if MFD_CS47L24 select SND_SOC_CS53L30 if I2C @@ -488,6 +489,11 @@ config SND_SOC_CS42XX8_I2C select SND_SOC_CS42XX8 select REGMAP_I2C +# Cirrus Logic CS43130 HiFi DAC +config SND_SOC_CS43130 + tristate "Cirrus Logic CS43130 CODEC" + depends on I2C + # Cirrus Logic CS4349 HiFi DAC config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 12dc2f7..896ef5a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -53,6 +53,7 @@ snd-soc-cs4271-i2c-objs := cs4271-i2c.o snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o +snd-soc-cs43130-objs := cs43130.o snd-soc-cs4349-objs := cs4349.o snd-soc-cs47l24-objs := cs47l24.o snd-soc-cs53l30-objs := cs53l30.o @@ -292,6 +293,7 @@ obj-$(CONFIG_SND_SOC_CS4271_I2C) += snd-soc-cs4271-i2c.o obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o +obj-$(CONFIG_SND_SOC_CS43130) += snd-soc-cs43130.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CS47L24) += snd-soc-cs47l24.o obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c new file mode 100644 index 0000000..de22919 --- /dev/null +++ b/sound/soc/codecs/cs43130.c @@ -0,0 +1,2675 @@ +/* + * cs43130.c -- CS43130 ALSA Soc Audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Authors: Li Xu <li.xu(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/pm_runtime.h> +#include <linux/of_irq.h> +#include <linux/completion.h> +#include <linux/mutex.h> +#include <linux/workqueue.h> +#include <sound/jack.h> + +#include "cs43130.h" + +static const struct reg_default cs43130_reg_defaults[] = { + {CS43130_SYS_CLK_CTL_1, 0x06}, + {CS43130_SP_SRATE, 0x01}, + {CS43130_SP_BITSIZE, 0x05}, + {CS43130_PAD_INT_CFG, 0x03}, + {CS43130_PWDN_CTL, 0xFE}, + {CS43130_CRYSTAL_SET, 0x04}, + {CS43130_PLL_SET_1, 0x00}, + {CS43130_PLL_SET_2, 0x00}, + {CS43130_PLL_SET_3, 0x00}, + {CS43130_PLL_SET_4, 0x00}, + {CS43130_PLL_SET_5, 0x40}, + {CS43130_PLL_SET_6, 0x10}, + {CS43130_PLL_SET_7, 0x80}, + {CS43130_PLL_SET_8, 0x03}, + {CS43130_PLL_SET_9, 0x02}, + {CS43130_PLL_SET_10, 0x02}, + {CS43130_CLKOUT_CTL, 0x00}, + {CS43130_ASP_NUM_1, 0x01}, + {CS43130_ASP_NUM_2, 0x00}, + {CS43130_ASP_DEN_1, 0x08}, + {CS43130_ASP_DEN_2, 0x00}, + {CS43130_ASP_LRCK_HI_TIME_1, 0x1F}, + {CS43130_ASP_LRCK_HI_TIME_2, 0x00}, + {CS43130_ASP_LRCK_PERIOD_1, 0x3F}, + {CS43130_ASP_LRCK_PERIOD_2, 0x00}, + {CS43130_ASP_CLOCK_CONF, 0x0C}, + {CS43130_ASP_FRAME_CONF, 0x0A}, + {CS43130_XSP_NUM_1, 0x01}, + {CS43130_XSP_NUM_2, 0x00}, + {CS43130_XSP_DEN_1, 0x02}, + {CS43130_XSP_DEN_2, 0x00}, + {CS43130_XSP_LRCK_HI_TIME_1, 0x1F}, + {CS43130_XSP_LRCK_HI_TIME_2, 0x00}, + {CS43130_XSP_LRCK_PERIOD_1, 0x3F}, + {CS43130_XSP_LRCK_PERIOD_2, 0x00}, + {CS43130_XSP_CLOCK_CONF, 0x0C}, + {CS43130_XSP_FRAME_CONF, 0x0A}, + {CS43130_ASP_CH_1_LOC, 0x00}, + {CS43130_ASP_CH_2_LOC, 0x00}, + {CS43130_ASP_CH_1_SZ_EN, 0x06}, + {CS43130_ASP_CH_2_SZ_EN, 0x0E}, + {CS43130_XSP_CH_1_LOC, 0x00}, + {CS43130_XSP_CH_2_LOC, 0x00}, + {CS43130_XSP_CH_1_SZ_EN, 0x06}, + {CS43130_XSP_CH_2_SZ_EN, 0x0E}, + {CS43130_DSD_VOL_B, 0x78}, + {CS43130_DSD_VOL_A, 0x78}, + {CS43130_DSD_PATH_CTL_1, 0xA8}, + {CS43130_DSD_INT_CFG, 0x00}, + {CS43130_DSD_PATH_CTL_2, 0x02}, + {CS43130_DSD_PCM_MIX_CTL, 0x00}, + {CS43130_DSD_PATH_CTL_3, 0x40}, + {CS43130_HP_OUT_CTL_1, 0x30}, + {CS43130_PCM_FILT_OPT, 0x02}, + {CS43130_PCM_VOL_B, 0x78}, + {CS43130_PCM_VOL_A, 0x78}, + {CS43130_PCM_PATH_CTL_1, 0xA8}, + {CS43130_PCM_PATH_CTL_2, 0x00}, + {CS43130_CLASS_H_CTL, 0x1E}, + {CS43130_HP_DETECT, 0x04}, + {CS43130_HP_LOAD_1, 0x00}, + {CS43130_HP_MEAS_LOAD_1, 0x00}, + {CS43130_HP_MEAS_LOAD_2, 0x00}, + {CS43130_INT_MASK_1, 0xFF}, + {CS43130_INT_MASK_2, 0xFF}, + {CS43130_INT_MASK_3, 0xFF}, + {CS43130_INT_MASK_4, 0xFF}, + {CS43130_INT_MASK_5, 0xFF}, +}; + +static bool cs43130_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5: + case CS43130_HP_DC_STAT_1 ... CS43130_HP_DC_STAT_2: + case CS43130_HP_AC_STAT_1 ... CS43130_HP_AC_STAT_2: + return true; + default: + return false; + } +} + +static bool cs43130_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS43130_DEVID_AB ... CS43130_SYS_CLK_CTL_1: + case CS43130_SP_SRATE ... CS43130_PAD_INT_CFG: + case CS43130_PWDN_CTL: + case CS43130_CRYSTAL_SET: + case CS43130_PLL_SET_1 ... CS43130_PLL_SET_5: + case CS43130_PLL_SET_6: + case CS43130_PLL_SET_7: + case CS43130_PLL_SET_8: + case CS43130_PLL_SET_9: + case CS43130_PLL_SET_10: + case CS43130_CLKOUT_CTL: + case CS43130_ASP_NUM_1 ... CS43130_ASP_FRAME_CONF: + case CS43130_XSP_NUM_1 ... CS43130_XSP_FRAME_CONF: + case CS43130_ASP_CH_1_LOC: + case CS43130_ASP_CH_2_LOC: + case CS43130_ASP_CH_1_SZ_EN: + case CS43130_ASP_CH_2_SZ_EN: + case CS43130_XSP_CH_1_LOC: + case CS43130_XSP_CH_2_LOC: + case CS43130_XSP_CH_1_SZ_EN: + case CS43130_XSP_CH_2_SZ_EN: + case CS43130_DSD_VOL_B ... CS43130_DSD_PATH_CTL_3: + case CS43130_HP_OUT_CTL_1: + case CS43130_PCM_FILT_OPT ... CS43130_PCM_PATH_CTL_2: + case CS43130_CLASS_H_CTL: + case CS43130_HP_DETECT: + case CS43130_HP_STATUS: + case CS43130_HP_LOAD_1: + case CS43130_HP_MEAS_LOAD_1: + case CS43130_HP_MEAS_LOAD_2: + case CS43130_HP_DC_STAT_1: + case CS43130_HP_DC_STAT_2: + case CS43130_HP_AC_STAT_1: + case CS43130_HP_AC_STAT_2: + case CS43130_HP_LOAD_STAT: + case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5: + case CS43130_INT_MASK_1 ... CS43130_INT_MASK_5: + return true; + default: + return false; + } +} + +static bool cs43130_precious_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5: + return true; + default: + return false; + } +} + +struct cs43130_pll_params { + unsigned int pll_in; + u8 sclk_prediv; + u8 pll_div_int; + u32 pll_div_frac; + u8 pll_mode; + u8 pll_divout; + unsigned int pll_out; + u8 pll_cal_ratio; +}; + +static const struct cs43130_pll_params pll_ratio_table[] = { + {9600000, 0x02, 0x49, 0x800000, 0x00, 0x08, 22579200, 151}, + {9600000, 0x02, 0x50, 0x000000, 0x00, 0x08, 24576000, 164}, + + {11289600, 0x02, 0X40, 0, 0x01, 0x08, 22579200, 128}, + {11289600, 0x02, 0x44, 0x06F700, 0x0, 0x08, 24576000, 139}, + + {12000000, 0x02, 0x49, 0x800000, 0x00, 0x0A, 22579200, 120}, + {12000000, 0x02, 0x40, 0x000000, 0x00, 0x08, 24576000, 131}, + + {12288000, 0x02, 0x49, 0x800000, 0x01, 0x0A, 22579200, 118}, + {12288000, 0x02, 0x40, 0x000000, 0x01, 0x08, 24576000, 128}, + + {13000000, 0x02, 0x45, 0x797680, 0x01, 0x0A, 22579200, 111}, + {13000000, 0x02, 0x3C, 0x7EA940, 0x01, 0x08, 24576000, 121}, + + {19200000, 0x03, 0x49, 0x800000, 0x00, 0x08, 22579200, 151}, + {19200000, 0x03, 0x50, 0x000000, 0x00, 0x08, 24576000, 164}, + + {22579200, 0, 0, 0, 0, 0, 22579200, 0}, + {22579200, 0x03, 0x44, 0x06F700, 0x00, 0x08, 24576000, 139}, + + {24000000, 0x03, 0x49, 0x800000, 0x00, 0x0A, 22579200, 120}, + {24000000, 0x03, 0x40, 0x000000, 0x00, 0x08, 24576000, 131}, + + {24576000, 0x03, 0x49, 0x800000, 0x01, 0x0A, 22579200, 118}, + {24576000, 0, 0, 0, 0, 0, 24576000, 0}, + + {26000000, 0x03, 0x45, 0x797680, 0x01, 0x0A, 22579200, 111}, + {26000000, 0x03, 0x3C, 0x7EA940, 0x01, 0x08, 24576000, 121}, +}; + +static const struct cs43130_pll_params *cs43130_get_pll_table( + unsigned int freq_in, unsigned int freq_out) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) { + if (pll_ratio_table[i].pll_in == freq_in && + pll_ratio_table[i].pll_out == freq_out) + return &pll_ratio_table[i]; + } + + return NULL; +} + +static int cs43130_pll_config(struct snd_soc_codec *codec) +{ + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + const struct cs43130_pll_params *pll_entry; + + dev_dbg(codec->dev, "cs43130->mclk = %u, cs43130->mclk_int = %u\n", + cs43130->mclk, cs43130->mclk_int); + + pll_entry = cs43130_get_pll_table(cs43130->mclk, cs43130->mclk_int); + if (!pll_entry) + return -EINVAL; + + if (pll_entry->pll_cal_ratio == 0) { + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_1, + CS43130_PLL_START_MASK, 0); + + cs43130->pll_bypass = true; + return 0; + } + + cs43130->pll_bypass = false; + + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_2, + CS43130_PLL_DIV_DATA_MASK, + pll_entry->pll_div_frac >> + CS43130_PLL_DIV_FRAC_0_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_3, + CS43130_PLL_DIV_DATA_MASK, + pll_entry->pll_div_frac >> + CS43130_PLL_DIV_FRAC_1_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_4, + CS43130_PLL_DIV_DATA_MASK, + pll_entry->pll_div_frac >> + CS43130_PLL_DIV_FRAC_2_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_PLL_SET_5, + pll_entry->pll_div_int); + regmap_write(cs43130->regmap, CS43130_PLL_SET_6, pll_entry->pll_divout); + regmap_write(cs43130->regmap, CS43130_PLL_SET_7, + pll_entry->pll_cal_ratio); + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_8, + CS43130_PLL_MODE_MASK, + pll_entry->pll_mode << CS43130_PLL_MODE_SHIFT); + regmap_write(cs43130->regmap, CS43130_PLL_SET_9, + pll_entry->sclk_prediv); + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_1, + CS43130_PLL_START_MASK, 1); + + return 0; +} + +static int cs43130_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + int ret = 0; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (freq_in) { + case 9600000: + case 11289600: + case 12000000: + case 12288000: + case 13000000: + case 19200000: + case 22579200: + case 24000000: + case 24576000: + case 26000000: + cs43130->mclk = freq_in; + break; + default: + dev_err(codec->dev, + "unsupported pll input reference clock:%d\n", freq_in); + return -EINVAL; + } + + switch (freq_out) { + case 22579200: + cs43130->mclk_int = freq_out; + break; + case 24576000: + cs43130->mclk_int = freq_out; + break; + default: + dev_err(codec->dev, + "unsupported pll output ref clock: %u\n", freq_out); + return -EINVAL; + } + + ret = cs43130_pll_config(codec); + dev_dbg(codec->dev, "cs43130->pll_bypass = %d", cs43130->pll_bypass); + return ret; +} + +static int cs43130_change_clksrc(struct snd_soc_codec *codec, + enum cs43130_mclk_src_sel src) +{ + int ret; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + int mclk_int_decoded; + + if (src == cs43130->mclk_int_src) { + /* clk source has not changed */ + return 0; + } + + switch (cs43130->mclk_int) { + case CS43130_MCLK_22M: + mclk_int_decoded = CS43130_MCLK_22P5; + break; + case CS43130_MCLK_24M: + mclk_int_decoded = CS43130_MCLK_24P5; + break; + default: + dev_err(codec->dev, "Invalid MCLK INT freq: %u\n", cs43130->mclk_int); + return -EINVAL; + } + + switch (src) { + case CS43130_MCLK_SRC_EXT: + cs43130->pll_bypass = true; + cs43130->mclk_int_src = CS43130_MCLK_SRC_EXT; + if (cs43130->xtal_ibias == CS43130_XTAL_UNUSED) { + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, + 1 << CS43130_PDN_XTAL_SHIFT); + } else { + reinit_completion(&cs43130->xtal_rdy); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_RDY_INT_MASK, 0); + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, 0); + ret = wait_for_completion_timeout(&cs43130->xtal_rdy, + msecs_to_jiffies(100)); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_RDY_INT_MASK, + 1 << CS43130_XTAL_RDY_INT_SHIFT); + if (ret == 0) { + dev_err(codec->dev, "Timeout waiting for XTAL_READY interrupt\n"); + return -ETIMEDOUT; + } + } + + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_SRC_SEL_MASK, + src << CS43130_MCLK_SRC_SEL_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_INT_MASK, + mclk_int_decoded << CS43130_MCLK_INT_SHIFT); + usleep_range(150, 200); + + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_PLL_MASK, + 1 << CS43130_PDN_PLL_SHIFT); + break; + case CS43130_MCLK_SRC_PLL: + cs43130->pll_bypass = false; + cs43130->mclk_int_src = CS43130_MCLK_SRC_PLL; + if (cs43130->xtal_ibias == CS43130_XTAL_UNUSED) { + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, + 1 << CS43130_PDN_XTAL_SHIFT); + } else { + reinit_completion(&cs43130->xtal_rdy); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_RDY_INT_MASK, 0); + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, 0); + ret = wait_for_completion_timeout(&cs43130->xtal_rdy, + msecs_to_jiffies(100)); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_RDY_INT_MASK, + 1 << CS43130_XTAL_RDY_INT_SHIFT); + if (ret == 0) { + dev_err(codec->dev, "Timeout waiting for XTAL_READY interrupt\n"); + return -ETIMEDOUT; + } + } + + reinit_completion(&cs43130->pll_rdy); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_PLL_RDY_INT_MASK, 0); + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_PLL_MASK, 0); + ret = wait_for_completion_timeout(&cs43130->pll_rdy, + msecs_to_jiffies(100)); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_PLL_RDY_INT_MASK, + 1 << CS43130_PLL_RDY_INT_SHIFT); + if (ret == 0) { + dev_err(codec->dev, "Timeout waiting for PLL_READY interrupt\n"); + return -ETIMEDOUT; + } + + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_SRC_SEL_MASK, + src << CS43130_MCLK_SRC_SEL_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_INT_MASK, + mclk_int_decoded << CS43130_MCLK_INT_SHIFT); + usleep_range(150, 200); + break; + case CS43130_MCLK_SRC_RCO: + cs43130->mclk_int_src = CS43130_MCLK_SRC_RCO; + + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_SRC_SEL_MASK, + src << CS43130_MCLK_SRC_SEL_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_INT_MASK, + CS43130_MCLK_22P5 << CS43130_MCLK_INT_SHIFT); + usleep_range(150, 200); + + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, + 1 << CS43130_PDN_XTAL_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_PLL_MASK, + 1 << CS43130_PDN_PLL_SHIFT); + break; + default: + dev_err(codec->dev, "Invalid MCLK source value\n"); + return -EINVAL; + } + + return 0; +} + +static const struct cs43130_bitwidth_map cs43130_bitwidth_table[] = { + {8, CS43130_SP_BIT_SIZE_8, CS43130_CH_BIT_SIZE_8}, + {16, CS43130_SP_BIT_SIZE_16, CS43130_CH_BIT_SIZE_16}, + {24, CS43130_SP_BIT_SIZE_24, CS43130_CH_BIT_SIZE_24}, + {32, CS43130_SP_BIT_SIZE_32, CS43130_CH_BIT_SIZE_32}, +}; + +static const struct cs43130_bitwidth_map *cs43130_get_bitwidth_table( + unsigned int bitwidth) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs43130_bitwidth_table); i++) { + if (cs43130_bitwidth_table[i].bitwidth == bitwidth) + return &cs43130_bitwidth_table[i]; + } + + return NULL; +} + +static int cs43130_set_bitwidth(int dai_id, unsigned int bitwidth_dai, + struct regmap *regmap) +{ + const struct cs43130_bitwidth_map *bw_map; + + bw_map = cs43130_get_bitwidth_table(bitwidth_dai); + if (!bw_map) + return -EINVAL; + + switch (dai_id) { + case CS43130_ASP_PCM_DAI: + case CS43130_ASP_DOP_DAI: + regmap_update_bits(regmap, CS43130_ASP_CH_1_SZ_EN, + CS43130_CH_BITSIZE_MASK, bw_map->ch_bit); + regmap_update_bits(regmap, CS43130_ASP_CH_2_SZ_EN, + CS43130_CH_BITSIZE_MASK, bw_map->ch_bit); + regmap_update_bits(regmap, CS43130_SP_BITSIZE, + CS43130_ASP_BITSIZE_MASK, bw_map->sp_bit); + break; + case CS43130_XSP_DOP_DAI: + regmap_update_bits(regmap, CS43130_XSP_CH_1_SZ_EN, + CS43130_CH_BITSIZE_MASK, bw_map->ch_bit); + regmap_update_bits(regmap, CS43130_XSP_CH_2_SZ_EN, + CS43130_CH_BITSIZE_MASK, bw_map->ch_bit); + regmap_update_bits(regmap, CS43130_SP_BITSIZE, + CS43130_XSP_BITSIZE_MASK, bw_map->sp_bit << + CS43130_XSP_BITSIZE_SHIFT); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct cs43130_rate_map cs43130_rate_table[] = { + {32000, CS43130_ASP_SPRATE_32K}, + {44100, CS43130_ASP_SPRATE_44_1K}, + {48000, CS43130_ASP_SPRATE_48K}, + {88200, CS43130_ASP_SPRATE_88_2K}, + {96000, CS43130_ASP_SPRATE_96K}, + {176400, CS43130_ASP_SPRATE_176_4K}, + {192000, CS43130_ASP_SPRATE_192K}, + {352800, CS43130_ASP_SPRATE_352_8K}, + {384000, CS43130_ASP_SPRATE_384K}, +}; + +static const struct cs43130_rate_map *cs43130_get_rate_table(int fs) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs43130_rate_table); i++) { + if (cs43130_rate_table[i].fs == fs) + return &cs43130_rate_table[i]; + } + + return NULL; +} + +static const struct cs43130_clk_gen *cs43130_get_clk_gen(int mclk_int, int fs, + const struct cs43130_clk_gen *clk_gen_table, int len_clk_gen_table) +{ + int i; + + for (i = 0; i < len_clk_gen_table; i++) { + if (clk_gen_table[i].mclk_int == mclk_int && + clk_gen_table[i].fs == fs) + return &clk_gen_table[i]; + } + + return NULL; +} + +static int cs43130_set_sp_fmt(int dai_id, unsigned int bitwidth_sclk, + struct snd_pcm_hw_params *params, + struct cs43130_private *cs43130) +{ + u16 frm_size; + u16 hi_size; + u8 frm_delay; + u8 frm_phase; + u8 frm_data; + u8 sclk_edge; + u8 lrck_edge; + u8 clk_data; + u8 loc_ch1; + u8 loc_ch2; + u8 dai_mode_val; + const struct cs43130_clk_gen *clk_gen; + + switch (cs43130->dais[dai_id].dai_format) { + case SND_SOC_DAIFMT_I2S: + hi_size = bitwidth_sclk; + frm_delay = 2; + frm_phase = 0; + break; + case SND_SOC_DAIFMT_LEFT_J: + hi_size = bitwidth_sclk; + frm_delay = 2; + frm_phase = 1; + break; + case SND_SOC_DAIFMT_DSP_A: + hi_size = 1; + frm_delay = 2; + frm_phase = 1; + break; + case SND_SOC_DAIFMT_DSP_B: + hi_size = 1; + frm_delay = 0; + frm_phase = 1; + break; + default: + return -EINVAL; + } + + switch (cs43130->dais[dai_id].dai_mode) { + case SND_SOC_DAIFMT_CBS_CFS: + dai_mode_val = 0; + break; + case SND_SOC_DAIFMT_CBM_CFM: + dai_mode_val = 1; + break; + default: + return -EINVAL; + } + + frm_size = bitwidth_sclk * params_channels(params); + sclk_edge = 1; + lrck_edge = 0; + loc_ch1 = 0; + loc_ch2 = bitwidth_sclk * (params_channels(params) - 1); + + frm_data = frm_delay & CS43130_SP_FSD_MASK; + frm_data |= (frm_phase << CS43130_SP_STP_SHIFT) & CS43130_SP_STP_MASK; + + clk_data = lrck_edge & CS43130_SP_LCPOL_IN_MASK; + clk_data |= (lrck_edge << CS43130_SP_LCPOL_OUT_SHIFT) & + CS43130_SP_LCPOL_OUT_MASK; + clk_data |= (sclk_edge << CS43130_SP_SCPOL_IN_SHIFT) & + CS43130_SP_SCPOL_IN_MASK; + clk_data |= (sclk_edge << CS43130_SP_SCPOL_OUT_SHIFT) & + CS43130_SP_SCPOL_OUT_MASK; + clk_data |= (dai_mode_val << CS43130_SP_MODE_SHIFT) & + CS43130_SP_MODE_MASK; + + switch (dai_id) { + case CS43130_ASP_PCM_DAI: + case CS43130_ASP_DOP_DAI: + regmap_update_bits(cs43130->regmap, CS43130_ASP_LRCK_PERIOD_1, + CS43130_SP_LCPR_DATA_MASK, (frm_size - 1) >> + CS43130_SP_LCPR_LSB_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_ASP_LRCK_PERIOD_2, + CS43130_SP_LCPR_DATA_MASK, (frm_size - 1) >> + CS43130_SP_LCPR_MSB_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_ASP_LRCK_HI_TIME_1, + CS43130_SP_LCHI_DATA_MASK, (hi_size - 1) >> + CS43130_SP_LCHI_LSB_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_ASP_LRCK_HI_TIME_2, + CS43130_SP_LCHI_DATA_MASK, (hi_size - 1) >> + CS43130_SP_LCHI_MSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_ASP_FRAME_CONF, frm_data); + regmap_write(cs43130->regmap, CS43130_ASP_CH_1_LOC, loc_ch1); + regmap_write(cs43130->regmap, CS43130_ASP_CH_2_LOC, loc_ch2); + regmap_update_bits(cs43130->regmap, CS43130_ASP_CH_1_SZ_EN, + CS43130_CH_EN_MASK, 1 << CS43130_CH_EN_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_ASP_CH_2_SZ_EN, + CS43130_CH_EN_MASK, 1 << CS43130_CH_EN_SHIFT); + regmap_write(cs43130->regmap, CS43130_ASP_CLOCK_CONF, clk_data); + break; + case CS43130_XSP_DOP_DAI: + regmap_update_bits(cs43130->regmap, CS43130_XSP_LRCK_PERIOD_1, + CS43130_SP_LCPR_DATA_MASK, (frm_size - 1) >> + CS43130_SP_LCPR_LSB_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_XSP_LRCK_PERIOD_2, + CS43130_SP_LCPR_DATA_MASK, (frm_size - 1) >> + CS43130_SP_LCPR_MSB_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_XSP_LRCK_HI_TIME_1, + CS43130_SP_LCHI_DATA_MASK, (hi_size - 1) >> + CS43130_SP_LCHI_LSB_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_XSP_LRCK_HI_TIME_2, + CS43130_SP_LCHI_DATA_MASK, (hi_size - 1) >> + CS43130_SP_LCHI_MSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_XSP_FRAME_CONF, frm_data); + regmap_write(cs43130->regmap, CS43130_XSP_CH_1_LOC, loc_ch1); + regmap_write(cs43130->regmap, CS43130_XSP_CH_2_LOC, loc_ch2); + regmap_update_bits(cs43130->regmap, CS43130_XSP_CH_1_SZ_EN, + CS43130_CH_EN_MASK, 1 << CS43130_CH_EN_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_XSP_CH_2_SZ_EN, + CS43130_CH_EN_MASK, 1 << CS43130_CH_EN_SHIFT); + regmap_write(cs43130->regmap, CS43130_XSP_CLOCK_CONF, clk_data); + break; + default: + return -EINVAL; + } + + switch (frm_size) { + case 16: + clk_gen = cs43130_get_clk_gen(cs43130->mclk_int, + params_rate(params), + cs43130_16_clk_gen, + ARRAY_SIZE(cs43130_16_clk_gen)); + break; + case 32: + clk_gen = cs43130_get_clk_gen(cs43130->mclk_int, + params_rate(params), + cs43130_32_clk_gen, + ARRAY_SIZE(cs43130_32_clk_gen)); + break; + case 48: + clk_gen = cs43130_get_clk_gen(cs43130->mclk_int, + params_rate(params), + cs43130_48_clk_gen, + ARRAY_SIZE(cs43130_48_clk_gen)); + break; + case 64: + clk_gen = cs43130_get_clk_gen(cs43130->mclk_int, + params_rate(params), + cs43130_64_clk_gen, + ARRAY_SIZE(cs43130_64_clk_gen)); + break; + default: + return -EINVAL; + } + + if (!clk_gen) + return -EINVAL; + + switch (dai_id) { + case CS43130_ASP_PCM_DAI: + case CS43130_ASP_DOP_DAI: + regmap_write(cs43130->regmap, CS43130_ASP_DEN_1, + (clk_gen->den & CS43130_SP_M_LSB_DATA_MASK) >> + CS43130_SP_M_LSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_ASP_DEN_2, + (clk_gen->den & CS43130_SP_M_MSB_DATA_MASK) >> + CS43130_SP_M_MSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_ASP_NUM_1, + (clk_gen->num & CS43130_SP_N_LSB_DATA_MASK) >> + CS43130_SP_N_LSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_ASP_NUM_2, + (clk_gen->num & CS43130_SP_N_MSB_DATA_MASK) >> + CS43130_SP_N_MSB_DATA_SHIFT); + break; + case CS43130_XSP_DOP_DAI: + regmap_write(cs43130->regmap, CS43130_XSP_DEN_1, + (clk_gen->den & CS43130_SP_M_LSB_DATA_MASK) >> + CS43130_SP_M_LSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_XSP_DEN_2, + (clk_gen->den & CS43130_SP_M_MSB_DATA_MASK) >> + CS43130_SP_M_MSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_XSP_NUM_1, + (clk_gen->num & CS43130_SP_N_LSB_DATA_MASK) >> + CS43130_SP_N_LSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_XSP_NUM_2, + (clk_gen->num & CS43130_SP_N_MSB_DATA_MASK) >> + CS43130_SP_N_MSB_DATA_SHIFT); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs43130_pcm_dsd_mix(bool en, struct regmap *regmap) +{ + if (en) { + regmap_update_bits(regmap, CS43130_DSD_PCM_MIX_CTL, + CS43130_MIX_PCM_PREP_MASK, + 1 << CS43130_MIX_PCM_PREP_SHIFT); + usleep_range(6000, 6050); + regmap_update_bits(regmap, CS43130_DSD_PCM_MIX_CTL, + CS43130_MIX_PCM_DSD_MASK, + 1 << CS43130_MIX_PCM_DSD_SHIFT); + } else { + regmap_update_bits(regmap, CS43130_DSD_PCM_MIX_CTL, + CS43130_MIX_PCM_DSD_MASK, + 0 << CS43130_MIX_PCM_DSD_SHIFT); + usleep_range(1600, 1650); + regmap_update_bits(regmap, CS43130_DSD_PCM_MIX_CTL, + CS43130_MIX_PCM_PREP_MASK, + 0 << CS43130_MIX_PCM_PREP_SHIFT); + } + + return 0; +} + +static int cs43130_dsd_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + unsigned int required_clk; + u8 dsd_speed; + + mutex_lock(&cs43130->clk_mutex); + if (!cs43130->clk_req) { + /* no DAI is currently using clk */ + if (!(CS43130_MCLK_22M % params_rate(params))) + required_clk = CS43130_MCLK_22M; + else + required_clk = CS43130_MCLK_24M; + + cs43130_set_pll(codec, 0, 0, cs43130->mclk, required_clk); + if (cs43130->pll_bypass) + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_EXT); + else + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_PLL); + } + + cs43130->clk_req++; + if (cs43130->clk_req == 2) + cs43130_pcm_dsd_mix(true, cs43130->regmap); + mutex_unlock(&cs43130->clk_mutex); + + switch (params_rate(params)) { + case 176400: + dsd_speed = 0; + break; + case 352800: + dsd_speed = 1; + break; + default: + dev_err(codec->dev, "Rate(%u) not supported\n", + params_rate(params)); + return -EINVAL; + } + + if (cs43130->dais[dai->id].dai_mode == SND_SOC_DAIFMT_CBM_CFM) + regmap_update_bits(cs43130->regmap, CS43130_DSD_INT_CFG, + CS43130_DSD_MASTER, CS43130_DSD_MASTER); + else + regmap_update_bits(cs43130->regmap, CS43130_DSD_INT_CFG, + CS43130_DSD_MASTER, 0); + + regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2, + CS43130_DSD_SPEED_MASK, + dsd_speed << CS43130_DSD_SPEED_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2, + CS43130_DSD_SRC_MASK, CS43130_DSD_SRC_DSD << + CS43130_DSD_SRC_SHIFT); + + return 0; +} + +static int cs43130_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + const struct cs43130_rate_map *rate_map; + unsigned int sclk = cs43130->dais[dai->id].sclk; + unsigned int bitwidth_sclk; + unsigned int bitwidth_dai = (unsigned int)(params_width(params)); + unsigned int required_clk; + u8 dsd_speed; + + mutex_lock(&cs43130->clk_mutex); + if (!cs43130->clk_req) { + /* no DAI is currently using clk */ + if (!(CS43130_MCLK_22M % params_rate(params))) + required_clk = CS43130_MCLK_22M; + else + required_clk = CS43130_MCLK_24M; + + cs43130_set_pll(codec, 0, 0, cs43130->mclk, required_clk); + if (cs43130->pll_bypass) + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_EXT); + else + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_PLL); + } + + cs43130->clk_req++; + if (cs43130->clk_req == 2) + cs43130_pcm_dsd_mix(true, cs43130->regmap); + mutex_unlock(&cs43130->clk_mutex); + + if (dai->id == CS43130_ASP_DOP_DAI || + dai->id == CS43130_XSP_DOP_DAI) { + /* DoP bitwidth is always 24-bit */ + bitwidth_dai = 24; + sclk = params_rate(params) * bitwidth_dai * + params_channels(params); + + switch (params_rate(params)) { + case 176400: + dsd_speed = 0; + break; + case 352800: + dsd_speed = 1; + break; + default: + dev_err(codec->dev, "Rate(%u) not supported\n", + params_rate(params)); + return -EINVAL; + } + + regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2, + CS43130_DSD_SPEED_MASK, + dsd_speed << CS43130_DSD_SPEED_SHIFT); + } else { + /* ASP PCM format */ + rate_map = cs43130_get_rate_table(params_rate(params)); + if (!rate_map) + return -EINVAL; + + regmap_write(cs43130->regmap, CS43130_SP_SRATE, rate_map->val); + } + + if (dai->id == CS43130_ASP_DOP_DAI) { + regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2, + CS43130_DSD_SRC_MASK, CS43130_DSD_SRC_ASP << + CS43130_DSD_SRC_SHIFT); + } else if (dai->id == CS43130_XSP_DOP_DAI) { + regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2, + CS43130_DSD_SRC_MASK, CS43130_DSD_SRC_XSP << + CS43130_DSD_SRC_SHIFT); + } + + + if (!sclk && cs43130->dais[dai->id].dai_mode == SND_SOC_DAIFMT_CBM_CFM) + /* Calculate SCLK in master mode if unassigned */ + sclk = params_rate(params) * bitwidth_dai * + params_channels(params); + + if (!sclk) { + /* at this point, SCLK must be set */ + dev_err(codec->dev, "SCLK freq is not set\n"); + return -EINVAL; + } + + bitwidth_sclk = (sclk / params_rate(params)) / params_channels(params); + if (bitwidth_sclk < bitwidth_dai) { + dev_err(codec->dev, "Format not supported: SCLK freq is too low\n"); + return -EINVAL; + } + + dev_dbg(codec->dev, + "sclk = %u, fs = %d, bitwidth_dai = %u\n", + sclk, params_rate(params), bitwidth_dai); + + dev_dbg(codec->dev, + "bitwidth_sclk = %u, num_ch = %u\n", + bitwidth_sclk, params_channels(params)); + + cs43130_set_bitwidth(dai->id, bitwidth_dai, cs43130->regmap); + cs43130_set_sp_fmt(dai->id, bitwidth_sclk, params, cs43130); + + return 0; +} + +static int cs43130_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + mutex_lock(&cs43130->clk_mutex); + cs43130->clk_req--; + if (!cs43130->clk_req) { + /* no DAI is currently using clk */ + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_RCO); + cs43130_pcm_dsd_mix(false, cs43130->regmap); + } + mutex_unlock(&cs43130->clk_mutex); + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(pcm_vol_tlv, -12750, 50, 1); + +static const char * const pcm_ch_text[] = { + "Left-Right Ch", + "Left-Left Ch", + "Right-Left Ch", + "Right-Right Ch", +}; + +static const struct reg_sequence pcm_ch_en_seq[] = { + {CS43130_DXD1, 0x99}, + {0x180005, 0x8C}, + {0x180007, 0xAB}, + {0x180015, 0x31}, + {0x180017, 0xB2}, + {0x180025, 0x30}, + {0x180027, 0x84}, + {0x180035, 0x9C}, + {0x180037, 0xAE}, + {0x18000D, 0x24}, + {0x18000F, 0xA3}, + {0x18001D, 0x05}, + {0x18001F, 0xD4}, + {0x18002D, 0x0B}, + {0x18002F, 0xC7}, + {0x18003D, 0x71}, + {0x18003F, 0xE7}, + {CS43130_DXD1, 0}, +}; + +static const struct reg_sequence pcm_ch_dis_seq[] = { + {CS43130_DXD1, 0x99}, + {0x180005, 0x24}, + {0x180007, 0xA3}, + {0x180015, 0x05}, + {0x180017, 0xD4}, + {0x180025, 0x0B}, + {0x180027, 0xC7}, + {0x180035, 0x71}, + {0x180037, 0xE7}, + {0x18000D, 0x8C}, + {0x18000F, 0xAB}, + {0x18001D, 0x31}, + {0x18001F, 0xB2}, + {0x18002D, 0x30}, + {0x18002F, 0x84}, + {0x18003D, 0x9C}, + {0x18003F, 0xAE}, + {CS43130_DXD1, 0}, +}; + +static int cs43130_pcm_ch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_soc_get_enum_double(kcontrol, ucontrol); +} + +static int cs43130_pcm_ch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (item[0] >= e->items) + return -EINVAL; + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + + if (cs43130->dev_id == CS43131_CHIP_ID || + cs43130->dev_id == CS43198_CHIP_ID) { + if (val >= 2) + regmap_multi_reg_write(cs43130->regmap, pcm_ch_en_seq, + ARRAY_SIZE(pcm_ch_en_seq)); + else + regmap_multi_reg_write(cs43130->regmap, pcm_ch_dis_seq, + ARRAY_SIZE(pcm_ch_dis_seq)); + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + +static SOC_ENUM_SINGLE_DECL(pcm_ch_enum, CS43130_PCM_PATH_CTL_2, 0, + pcm_ch_text); + +static const char * const pcm_spd_texts[] = { + "Fast", + "Slow", +}; + +static SOC_ENUM_SINGLE_DECL(pcm_spd_enum, CS43130_PCM_FILT_OPT, 7, + pcm_spd_texts); + +static const char * const enum_texts[] = { + "Off", + "On", +}; + +static SOC_ENUM_SINGLE_DECL(pcm_phs_enum, CS43130_PCM_FILT_OPT, 6, enum_texts); + +static SOC_ENUM_SINGLE_DECL(pcm_nos_enum, CS43130_PCM_FILT_OPT, 5, enum_texts); + +static SOC_ENUM_SINGLE_DECL(pcm_high_enum, CS43130_PCM_FILT_OPT, 1, enum_texts); + +static SOC_ENUM_SINGLE_DECL(pcm_dmp_enum, CS43130_PCM_FILT_OPT, 0, enum_texts); + +static const char * const dsd_texts[] = { + "Off", + "BCKA Mode", + "BCKD Mode", +}; + +static const unsigned int dsd_values[] = { + CS43130_DSD_SRC_DSD, + CS43130_DSD_SRC_ASP, + CS43130_DSD_SRC_XSP, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(dsd_enum, CS43130_DSD_INT_CFG, 0, 0x03, + dsd_texts, dsd_values); + +static const struct snd_kcontrol_new cs43130_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS43130_PCM_VOL_A, CS43130_PCM_VOL_B, 0, 0xFF, 1, + pcm_vol_tlv), + SOC_DOUBLE_R_TLV("Master DSD Playback Volume", + CS43130_DSD_VOL_A, CS43130_DSD_VOL_B, 0, 0xFF, 1, + pcm_vol_tlv), + SOC_ENUM_EXT("PCM Ch Select", pcm_ch_enum, cs43130_pcm_ch_get, + cs43130_pcm_ch_put), + SOC_ENUM("PCM Filter Speed", pcm_spd_enum), + SOC_ENUM("PCM Phase Compensation", pcm_phs_enum), + SOC_ENUM("PCM Nonoversampling Emulation", pcm_nos_enum), + SOC_ENUM("PCM High-pass Filter", pcm_high_enum), + SOC_ENUM("PCM De-emphasis Filter", pcm_dmp_enum), + SOC_ENUM("DSD Phase Modulation", dsd_enum), +}; + +static const struct reg_sequence pcm_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD7, 0x01}, + {CS43130_DXD8, 0}, + {CS43130_DXD9, 0x01}, + {CS43130_DXD3, 0x12}, + {CS43130_DXD4, 0}, + {CS43130_DXD10, 0x28}, + {CS43130_DXD11, 0x28}, + {CS43130_DXD1, 0}, +}; + +static const struct reg_sequence dsd_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD7, 0x01}, + {CS43130_DXD8, 0}, + {CS43130_DXD9, 0x01}, + {CS43130_DXD3, 0x12}, + {CS43130_DXD4, 0}, + {CS43130_DXD10, 0x1E}, + {CS43130_DXD11, 0x20}, + {CS43130_DXD1, 0}, +}; + +static const struct reg_sequence pop_free_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD12, 0x0A}, + {CS43130_DXD1, 0}, +}; + +static const struct reg_sequence pop_free_seq2[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD13, 0x20}, + {CS43130_DXD1, 0}, +}; + +static int cs43130_dsd_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (cs43130->dev_id == CS43130_CHIP_ID || + cs43130->dev_id == CS4399_CHIP_ID) + regmap_multi_reg_write(cs43130->regmap, dsd_seq, + ARRAY_SIZE(dsd_seq)); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static int cs43130_pcm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (cs43130->dev_id == CS43130_CHIP_ID || + cs43130->dev_id == CS4399_CHIP_ID) + regmap_multi_reg_write(cs43130->regmap, pcm_seq, + ARRAY_SIZE(pcm_seq)); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static const struct reg_sequence dac_postpmu_seq[] = { + {CS43130_DXD9, 0x0C}, + {CS43130_DXD3, 0x10}, + {CS43130_DXD4, 0x20}, +}; + +static const struct reg_sequence dac_postpmd_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD6, 0x01}, + {CS43130_DXD1, 0}, +}; + +static int cs43130_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (cs43130->dev_id == CS43130_CHIP_ID || + cs43130->dev_id == CS4399_CHIP_ID) + regmap_multi_reg_write(cs43130->regmap, pop_free_seq, + ARRAY_SIZE(pop_free_seq)); + else + /* CS43131 or CS43198 */ + regmap_multi_reg_write(cs43130->regmap, pop_free_seq2, + ARRAY_SIZE(pop_free_seq2)); + break; + case SND_SOC_DAPM_POST_PMU: + usleep_range(10000, 10050); + + regmap_write(cs43130->regmap, CS43130_DXD1, 0x99); + + if (cs43130->dev_id == CS43130_CHIP_ID || + cs43130->dev_id == CS4399_CHIP_ID) { + regmap_multi_reg_write(cs43130->regmap, dac_postpmu_seq, + ARRAY_SIZE(dac_postpmu_seq)); + /* + * Per datasheet, Sec. PCM Power-Up Sequence. + * According to Design, CS43130_DXD12 must be 0 to meet + * THDN and Dynamic Range spec. + */ + msleep(1000); + regmap_write(cs43130->regmap, CS43130_DXD12, 0); + } else { + /* CS43131 or CS43198 */ + usleep_range(12000, 12010); + regmap_write(cs43130->regmap, CS43130_DXD13, 0); + } + + regmap_write(cs43130->regmap, CS43130_DXD1, 0); + break; + case SND_SOC_DAPM_POST_PMD: + if (cs43130->dev_id == CS43130_CHIP_ID || + cs43130->dev_id == CS4399_CHIP_ID) + regmap_multi_reg_write(cs43130->regmap, dac_postpmd_seq, + ARRAY_SIZE(dac_postpmd_seq)); + break; + default: + dev_err(codec->dev, "Invalid DAC event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static const struct reg_sequence hpin_prepmd_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD15, 0x64}, + {CS43130_DXD14, 0}, + {CS43130_DXD2, 0}, + {CS43130_DXD1, 0}, +}; + +static const struct reg_sequence hpin_postpmu_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD2, 1}, + {CS43130_DXD14, 0xDC}, + {CS43130_DXD15, 0xE4}, + {CS43130_DXD1, 0}, +}; + +static int cs43130_hpin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + regmap_multi_reg_write(cs43130->regmap, hpin_prepmd_seq, + ARRAY_SIZE(hpin_prepmd_seq)); + break; + case SND_SOC_DAPM_PRE_PMU: + regmap_multi_reg_write(cs43130->regmap, hpin_postpmu_seq, + ARRAY_SIZE(hpin_postpmu_seq)); + break; + default: + dev_err(codec->dev, "Invalid HPIN event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static const struct snd_soc_dapm_widget digital_hp_widgets[] = { + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + + SND_SOC_DAPM_AIF_IN_E("ASPIN PCM", NULL, 0, CS43130_PWDN_CTL, + CS43130_PDN_ASP_SHIFT, 1, cs43130_pcm_event, + SND_SOC_DAPM_PRE_PMU), + + SND_SOC_DAPM_AIF_IN_E("ASPIN DoP", NULL, 0, CS43130_PWDN_CTL, + CS43130_PDN_ASP_SHIFT, 1, cs43130_dsd_event, + SND_SOC_DAPM_PRE_PMU), + + SND_SOC_DAPM_AIF_IN_E("XSPIN DoP", NULL, 0, CS43130_PWDN_CTL, + CS43130_PDN_XSP_SHIFT, 1, cs43130_dsd_event, + SND_SOC_DAPM_PRE_PMU), + + SND_SOC_DAPM_AIF_IN_E("XSPIN DSD", NULL, 0, CS43130_PWDN_CTL, + CS43130_PDN_DSDIF_SHIFT, 1, cs43130_dsd_event, + SND_SOC_DAPM_PRE_PMU), + + SND_SOC_DAPM_DAC("DSD", NULL, CS43130_DSD_PATH_CTL_2, + CS43130_DSD_EN_SHIFT, 0), + + SND_SOC_DAPM_DAC_E("HiFi DAC", NULL, CS43130_PWDN_CTL, + CS43130_PDN_HP_SHIFT, 1, cs43130_dac_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD)), +}; + +static const struct snd_soc_dapm_widget analog_hp_widgets[] = { + SND_SOC_DAPM_DAC_E("Analog Playback", NULL, CS43130_HP_OUT_CTL_1, + CS43130_HP_IN_EN_SHIFT, 0, cs43130_hpin_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), +}; + +static struct snd_soc_dapm_widget all_hp_widgets[ + ARRAY_SIZE(digital_hp_widgets) + + ARRAY_SIZE(analog_hp_widgets)]; + +static const struct snd_soc_dapm_route digital_hp_routes[] = { + {"ASPIN PCM", NULL, "ASP PCM Playback"}, + {"ASPIN DoP", NULL, "ASP DoP Playback"}, + {"XSPIN DoP", NULL, "XSP DoP Playback"}, + {"XSPIN DSD", NULL, "XSP DSD Playback"}, + {"DSD", NULL, "ASPIN DoP"}, + {"DSD", NULL, "XSPIN DoP"}, + {"DSD", NULL, "XSPIN DSD"}, + {"HiFi DAC", NULL, "ASPIN PCM"}, + {"HiFi DAC", NULL, "DSD"}, + {"HPOUTA", NULL, "HiFi DAC"}, + {"HPOUTB", NULL, "HiFi DAC"}, +}; + +static const struct snd_soc_dapm_route analog_hp_routes[] = { + {"HPOUTA", NULL, "Analog Playback"}, + {"HPOUTB", NULL, "Analog Playback"}, +}; + +static struct snd_soc_dapm_route all_hp_routes[ + ARRAY_SIZE(digital_hp_routes) + + ARRAY_SIZE(analog_hp_routes)]; + +static const unsigned int cs43130_asp_src_rates[] = { + 32000, 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 +}; + +static const struct snd_pcm_hw_constraint_list cs43130_asp_constraints = { + .count = ARRAY_SIZE(cs43130_asp_src_rates), + .list = cs43130_asp_src_rates, +}; + +static int cs43130_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs43130_asp_constraints); +} + +static const unsigned int cs43130_dop_src_rates[] = { + 176400, 352800, +}; + +static const struct snd_pcm_hw_constraint_list cs43130_dop_constraints = { + .count = ARRAY_SIZE(cs43130_dop_src_rates), + .list = cs43130_dop_src_rates, +}; + +static int cs43130_dop_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs43130_dop_constraints); +} + +static int cs43130_pcm_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs43130->dais[codec_dai->id].dai_mode = SND_SOC_DAIFMT_CBS_CFS; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs43130->dais[codec_dai->id].dai_mode = SND_SOC_DAIFMT_CBM_CFM; + break; + default: + dev_err(codec->dev, "unsupported mode\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_DSP_B; + break; + default: + dev_err(codec->dev, + "unsupported audio format\n"); + return -EINVAL; + } + + dev_dbg(codec->dev, "dai_id = %d, dai_mode = %u, dai_format = %u\n", + codec_dai->id, + cs43130->dais[codec_dai->id].dai_mode, + cs43130->dais[codec_dai->id].dai_format); + + return 0; +} + +static int cs43130_dsd_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs43130->dais[codec_dai->id].dai_mode = SND_SOC_DAIFMT_CBS_CFS; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs43130->dais[codec_dai->id].dai_mode = SND_SOC_DAIFMT_CBM_CFM; + break; + default: + dev_err(codec->dev, "Unsupported DAI format.\n"); + return -EINVAL; + } + + dev_dbg(codec->dev, "dai_mode = 0x%x\n", + cs43130->dais[codec_dai->id].dai_mode); + + return 0; +} + +static const struct reg_sequence mute_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD3, 0x12}, + {CS43130_DXD5, 0x02}, + {CS43130_DXD4, 0x12}, + {CS43130_DXD1, 0}, +}; + +static const struct reg_sequence unmute_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD3, 0x10}, + {CS43130_DXD5, 0}, + {CS43130_DXD4, 0x16}, + {CS43130_DXD1, 0}, +}; + +static int cs43130_pcm_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + if (mute) { + if (cs43130->dev_id == CS43130_CHIP_ID || + cs43130->dev_id == CS4399_CHIP_ID) { + regmap_multi_reg_write(cs43130->regmap, mute_seq, + ARRAY_SIZE(mute_seq)); + regmap_update_bits(cs43130->regmap, + CS43130_PCM_PATH_CTL_1, + CS43130_MUTE_MASK, CS43130_MUTE_EN); + /* + * PCM Power Down Sequence + * According to Design, 130ms is preferred. + */ + msleep(130); + } else { + regmap_update_bits(cs43130->regmap, + CS43130_PCM_PATH_CTL_1, + CS43130_MUTE_MASK, CS43130_MUTE_EN); + } + } else { + /* Unmute */ + regmap_update_bits(cs43130->regmap, CS43130_PCM_PATH_CTL_1, + CS43130_MUTE_MASK, 0); + if (cs43130->dev_id == CS43130_CHIP_ID || + cs43130->dev_id == CS4399_CHIP_ID) + regmap_multi_reg_write(cs43130->regmap, unmute_seq, + ARRAY_SIZE(unmute_seq)); + } + + return 0; +} + +static int cs43130_dsd_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + if (mute) { + if (cs43130->dev_id == CS43130_CHIP_ID || + cs43130->dev_id == CS4399_CHIP_ID) { + regmap_multi_reg_write(cs43130->regmap, mute_seq, + ARRAY_SIZE(mute_seq)); + regmap_update_bits(cs43130->regmap, + CS43130_DSD_PATH_CTL_1, + CS43130_MUTE_MASK, CS43130_MUTE_EN); + /* + * DSD Power Down Sequence + * According to Design, 130ms is preferred. + */ + msleep(130); + } else { + regmap_update_bits(cs43130->regmap, + CS43130_DSD_PATH_CTL_1, + CS43130_MUTE_MASK, CS43130_MUTE_EN); + } + } else { + /* Unmute */ + regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_1, + CS43130_MUTE_MASK, 0); + if (cs43130->dev_id == CS43130_CHIP_ID || + cs43130->dev_id == CS4399_CHIP_ID) + regmap_multi_reg_write(cs43130->regmap, unmute_seq, + ARRAY_SIZE(unmute_seq)); + } + + return 0; +} + +static int cs43130_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + cs43130->dais[codec_dai->id].sclk = freq; + dev_dbg(codec->dev, "dai_id = %d, sclk = %u\n", codec_dai->id, + cs43130->dais[codec_dai->id].sclk); + + return 0; +} + +static const struct snd_soc_dai_ops cs43130_pcm_ops = { + .startup = cs43130_pcm_startup, + .hw_params = cs43130_hw_params, + .hw_free = cs43130_hw_free, + .set_sysclk = cs43130_set_sysclk, + .set_fmt = cs43130_pcm_set_fmt, + .digital_mute = cs43130_pcm_mute, +}; + +static const struct snd_soc_dai_ops cs43130_dop_ops = { + .startup = cs43130_dop_startup, + .hw_params = cs43130_hw_params, + .hw_free = cs43130_hw_free, + .set_sysclk = cs43130_set_sysclk, + .set_fmt = cs43130_pcm_set_fmt, + .digital_mute = cs43130_dsd_mute, +}; + +static const struct snd_soc_dai_ops cs43130_dsd_ops = { + .startup = cs43130_dop_startup, + .hw_params = cs43130_dsd_hw_params, + .hw_free = cs43130_hw_free, + .set_fmt = cs43130_dsd_set_fmt, + .digital_mute = cs43130_dsd_mute, +}; + +static struct snd_soc_dai_driver cs43130_dai[] = { + { + .name = "cs43130-asp-pcm", + .id = CS43130_ASP_PCM_DAI, + .playback = { + .stream_name = "ASP PCM Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS43130_PCM_FORMATS, + }, + .ops = &cs43130_pcm_ops, + .symmetric_rates = 1, + }, + { + .name = "cs43130-asp-dop", + .id = CS43130_ASP_DOP_DAI, + .playback = { + .stream_name = "ASP DoP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS43130_DOP_FORMATS, + }, + .ops = &cs43130_dop_ops, + .symmetric_rates = 1, + }, + { + .name = "cs43130-xsp-dop", + .id = CS43130_XSP_DOP_DAI, + .playback = { + .stream_name = "XSP DoP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS43130_DOP_FORMATS, + }, + .ops = &cs43130_dop_ops, + .symmetric_rates = 1, + }, + { + .name = "cs43130-xsp-dsd", + .id = CS43130_XSP_DSD_DAI, + .playback = { + .stream_name = "XSP DSD Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS43130_DOP_FORMATS, + }, + .ops = &cs43130_dsd_ops, + }, + +}; + +static int cs43130_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, + int dir) +{ + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "clk_id = %d, source = %d, freq = %d, dir = %d\n", + clk_id, source, freq, dir); + + switch (freq) { + case CS43130_MCLK_22M: + case CS43130_MCLK_24M: + cs43130->mclk = freq; + break; + default: + dev_err(codec->dev, "Invalid MCLK INT freq: %u\n", freq); + return -EINVAL; + } + + if (source == CS43130_MCLK_SRC_EXT) { + cs43130->pll_bypass = true; + } else { + dev_err(codec->dev, "Invalid MCLK source\n"); + return -EINVAL; + } + + return 0; +} + +static inline u16 cs43130_get_ac_reg_val(u16 ac_freq) +{ + /* + * Depending on mclk_int, + * "HPLOAD_MEAS_FREQ_LSB/MSB" counts AC freq in + * 5.86Hz (mclk_int = 24.576MHz) + * or 5.94Hz (mclk_int = 22.5792MHz) steps. + * Because integer step conversion is much easier, + * I choose to use mclk_int = 22.5792MHz and approximate + * 5.94Hz as 6Hz step. + * The approximation worst case scenario occur at 20KHz, + * with approximated freq = 19.798KHz. + * Error freq = 202Hz + */ + + return ac_freq / 6; +} + +static char *hpload_msgs[] = { + "HPLOAD_DC_L:%u\n", + "HPLOAD_DC_R:%u\n", +}; + +static char *hpload_ac_msgs[] = { + "HPLOAD_AC_L:%u:%u\n", + "HPLOAD_AC_R:%u:%u\n", +}; + +static ssize_t cs43130_show_hpload(struct device *dev, + struct device_attribute *attr, char *buf) +{ + int i = 0, j, k, ret; + struct i2c_client *client = to_i2c_client(dev); + struct cs43130_private *cs43130 = i2c_get_clientdata(client); + + if (!cs43130->hpload_done) + return scnprintf(buf, PAGE_SIZE, "NO_HPLOAD\n"); + + for (j = 0; j < ARRAY_SIZE(hpload_msgs); j++) { + ret = scnprintf(buf + i, PAGE_SIZE - i, hpload_msgs[j], + cs43130->hpload_dc[j]); + if (!ret) + return i; + + i = i + ret; + } + + if (cs43130->ac_meas) { + for (k = 0; k < CS43130_AC_FREQ; k++) { + for (j = 0; j < ARRAY_SIZE(hpload_ac_msgs); j++) { + ret = scnprintf(buf + i, PAGE_SIZE - i, + hpload_ac_msgs[j], + cs43130->ac_freq[k], + cs43130->hpload_ac[k][j]); + if (!ret) + return i; + + i = i + ret; + } + } + } + + return i; +} + +static DEVICE_ATTR(hpload, S_IRUGO, cs43130_show_hpload, NULL); + +static struct reg_sequence hp_en_cal_seq[] = { + {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, + {CS43130_HP_MEAS_LOAD_1, 0}, + {CS43130_HP_MEAS_LOAD_2, 0}, + {CS43130_INT_MASK_4, 0}, + {CS43130_DXD1, 0x99}, + {CS43130_DXD16, 0xBB}, + {CS43130_DXD12, 0x01}, + {CS43130_DXD19, 0xCB}, + {CS43130_DXD17, 0x95}, + {CS43130_DXD18, 0x0B}, + {CS43130_DXD1, 0}, + {CS43130_HP_LOAD_1, 0x80}, +}; + +static struct reg_sequence hp_en_cal_seq2[] = { + {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, + {CS43130_HP_MEAS_LOAD_1, 0}, + {CS43130_HP_MEAS_LOAD_2, 0}, + {CS43130_INT_MASK_4, 0}, + {CS43130_HP_LOAD_1, 0x80}, +}; + +static struct reg_sequence hp_dis_cal_seq[] = { + {CS43130_HP_LOAD_1, 0x80}, + {CS43130_DXD1, 0x99}, + {CS43130_DXD12, 0}, + {CS43130_DXD1, 0}, + {CS43130_HP_LOAD_1, 0}, +}; + +static struct reg_sequence hp_dis_cal_seq2[] = { + {CS43130_HP_LOAD_1, 0x80}, + {CS43130_HP_LOAD_1, 0}, +}; + +static struct reg_sequence hp_dc_ch_l_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD19, 0x0A}, + {CS43130_DXD17, 0x93}, + {CS43130_DXD18, 0x0A}, + {CS43130_DXD1, 0}, + {CS43130_HP_LOAD_1, 0x80}, + {CS43130_HP_LOAD_1, 0x81}, +}; + +static struct reg_sequence hp_dc_ch_l_seq2[] = { + {CS43130_HP_LOAD_1, 0x80}, + {CS43130_HP_LOAD_1, 0x81}, +}; + +static struct reg_sequence hp_dc_ch_r_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD19, 0x8A}, + {CS43130_DXD17, 0x15}, + {CS43130_DXD18, 0x06}, + {CS43130_DXD1, 0}, + {CS43130_HP_LOAD_1, 0x90}, + {CS43130_HP_LOAD_1, 0x91}, +}; + +static struct reg_sequence hp_dc_ch_r_seq2[] = { + {CS43130_HP_LOAD_1, 0x90}, + {CS43130_HP_LOAD_1, 0x91}, +}; + +static struct reg_sequence hp_ac_ch_l_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD19, 0x0A}, + {CS43130_DXD17, 0x93}, + {CS43130_DXD18, 0x0A}, + {CS43130_DXD1, 0}, + {CS43130_HP_LOAD_1, 0x80}, + {CS43130_HP_LOAD_1, 0x82}, +}; + +static struct reg_sequence hp_ac_ch_l_seq2[] = { + {CS43130_HP_LOAD_1, 0x80}, + {CS43130_HP_LOAD_1, 0x82}, +}; + +static struct reg_sequence hp_ac_ch_r_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD19, 0x8A}, + {CS43130_DXD17, 0x15}, + {CS43130_DXD18, 0x06}, + {CS43130_DXD1, 0}, + {CS43130_HP_LOAD_1, 0x90}, + {CS43130_HP_LOAD_1, 0x92}, +}; + +static struct reg_sequence hp_ac_ch_r_seq2[] = { + {CS43130_HP_LOAD_1, 0x90}, + {CS43130_HP_LOAD_1, 0x92}, +}; + +static struct reg_sequence hp_cln_seq[] = { + {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, + {CS43130_HP_MEAS_LOAD_1, 0}, + {CS43130_HP_MEAS_LOAD_2, 0}, +}; + +struct reg_sequences { + struct reg_sequence *seq; + int size; + unsigned int msk; +}; + +static struct reg_sequences hpload_seq1[] = { + { + .seq = hp_en_cal_seq, + .size = ARRAY_SIZE(hp_en_cal_seq), + .msk = CS43130_HPLOAD_ON_INT, + }, + { + .seq = hp_dc_ch_l_seq, + .size = ARRAY_SIZE(hp_dc_ch_l_seq), + .msk = CS43130_HPLOAD_DC_INT, + }, + { + .seq = hp_ac_ch_l_seq, + .size = ARRAY_SIZE(hp_ac_ch_l_seq), + .msk = CS43130_HPLOAD_AC_INT, + }, + { + .seq = hp_dis_cal_seq, + .size = ARRAY_SIZE(hp_dis_cal_seq), + .msk = CS43130_HPLOAD_OFF_INT, + }, + { + .seq = hp_en_cal_seq, + .size = ARRAY_SIZE(hp_en_cal_seq), + .msk = CS43130_HPLOAD_ON_INT, + }, + { + .seq = hp_dc_ch_r_seq, + .size = ARRAY_SIZE(hp_dc_ch_r_seq), + .msk = CS43130_HPLOAD_DC_INT, + }, + { + .seq = hp_ac_ch_r_seq, + .size = ARRAY_SIZE(hp_ac_ch_r_seq), + .msk = CS43130_HPLOAD_AC_INT, + }, +}; + +static struct reg_sequences hpload_seq2[] = { + { + .seq = hp_en_cal_seq2, + .size = ARRAY_SIZE(hp_en_cal_seq2), + .msk = CS43130_HPLOAD_ON_INT, + }, + { + .seq = hp_dc_ch_l_seq2, + .size = ARRAY_SIZE(hp_dc_ch_l_seq2), + .msk = CS43130_HPLOAD_DC_INT, + }, + { + .seq = hp_ac_ch_l_seq2, + .size = ARRAY_SIZE(hp_ac_ch_l_seq2), + .msk = CS43130_HPLOAD_AC_INT, + }, + { + .seq = hp_dis_cal_seq2, + .size = ARRAY_SIZE(hp_dis_cal_seq2), + .msk = CS43130_HPLOAD_OFF_INT, + }, + { + .seq = hp_en_cal_seq2, + .size = ARRAY_SIZE(hp_en_cal_seq2), + .msk = CS43130_HPLOAD_ON_INT, + }, + { + .seq = hp_dc_ch_r_seq2, + .size = ARRAY_SIZE(hp_dc_ch_r_seq2), + .msk = CS43130_HPLOAD_DC_INT, + }, + { + .seq = hp_ac_ch_r_seq2, + .size = ARRAY_SIZE(hp_ac_ch_r_seq2), + .msk = CS43130_HPLOAD_AC_INT, + }, +}; + +static int cs43130_update_hpload(unsigned int msk, int ac_idx, + struct cs43130_private *cs43130) +{ + bool left_ch = true; + unsigned int reg; + u32 addr; + u16 impedance; + struct snd_soc_codec *codec = cs43130->codec; + + if (msk != CS43130_HPLOAD_DC_INT && + msk != CS43130_HPLOAD_AC_INT) + return 0; + + regmap_read(cs43130->regmap, CS43130_HP_LOAD_1, &reg); + if (reg & CS43130_HPLOAD_CHN_SEL) + left_ch = false; + + if (msk == CS43130_HPLOAD_DC_INT) + addr = CS43130_HP_DC_STAT_1; + else + addr = CS43130_HP_AC_STAT_1; + + regmap_read(cs43130->regmap, addr, &reg); + impedance = reg >> 3; + regmap_read(cs43130->regmap, addr + 1, &reg); + impedance |= reg << 5; + + if (msk == CS43130_HPLOAD_DC_INT) { + if (left_ch) + cs43130->hpload_dc[0] = impedance; + else + cs43130->hpload_dc[1] = impedance; + + dev_dbg(codec->dev, "HP DC impedance (Ch %u): %u\n", !left_ch, + impedance); + } else { + if (left_ch) + cs43130->hpload_ac[ac_idx][0] = impedance; + else + cs43130->hpload_ac[ac_idx][1] = impedance; + + dev_dbg(codec->dev, "HP AC (%u Hz) impedance (Ch %u): %u\n", + cs43130->ac_freq[ac_idx], !left_ch, impedance); + } + + return 0; +} + +static int cs43130_hpload_proc(struct cs43130_private *cs43130, + struct reg_sequence *seq, int seq_size, + unsigned int rslt_msk, int ac_idx) +{ + int ret; + unsigned int msk; + u16 ac_reg_val; + struct snd_soc_codec *codec = cs43130->codec; + + reinit_completion(&cs43130->hpload_evt); + + if (rslt_msk == CS43130_HPLOAD_AC_INT) { + ac_reg_val = cs43130_get_ac_reg_val(cs43130->ac_freq[ac_idx]); + regmap_update_bits(cs43130->regmap, CS43130_HP_LOAD_1, + CS43130_HPLOAD_AC_START, 0); + regmap_update_bits(cs43130->regmap, CS43130_HP_MEAS_LOAD_1, + CS43130_HP_MEAS_LOAD_MASK, + ac_reg_val >> CS43130_HP_MEAS_LOAD_1_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_HP_MEAS_LOAD_2, + CS43130_HP_MEAS_LOAD_MASK, + ac_reg_val >> CS43130_HP_MEAS_LOAD_2_SHIFT); + } + + regmap_multi_reg_write(cs43130->regmap, seq, + seq_size); + + ret = wait_for_completion_timeout(&cs43130->hpload_evt, + msecs_to_jiffies(1000)); + regmap_read(cs43130->regmap, CS43130_INT_MASK_4, &msk); + if (!ret) { + dev_err(codec->dev, "Timeout waiting for HPLOAD interrupt\n"); + return -1; + } + + dev_dbg(codec->dev, "HP load stat: %x, INT_MASK_4: %x\n", + cs43130->hpload_stat, msk); + if ((cs43130->hpload_stat & (CS43130_HPLOAD_NO_DC_INT | + CS43130_HPLOAD_UNPLUG_INT | + CS43130_HPLOAD_OOR_INT)) || + !(cs43130->hpload_stat & rslt_msk)) { + dev_dbg(codec->dev, "HP load measure failed\n"); + return -1; + } + + return 0; +} + +static const struct reg_sequence hv_seq[][2] = { + { + {CS43130_CLASS_H_CTL, 0x1C}, + {CS43130_HP_OUT_CTL_1, 0x10}, + }, + { + {CS43130_CLASS_H_CTL, 0x1E}, + {CS43130_HP_OUT_CTL_1, 0x20}, + }, + { + {CS43130_CLASS_H_CTL, 0x1E}, + {CS43130_HP_OUT_CTL_1, 0x30}, + }, +}; + +static int cs43130_set_hv(struct regmap *regmap, u16 hpload_dc, + const u16 *dc_threshold) +{ + int i; + + for (i = 0; i < CS43130_DC_THRESHOLD; i++) { + if (hpload_dc <= dc_threshold[i]) + break; + } + + regmap_multi_reg_write(regmap, hv_seq[i], ARRAY_SIZE(hv_seq[i])); + + return 0; +} + +static void cs43130_imp_meas(struct work_struct *wk) +{ + unsigned int reg, seq_size; + int i, ret, ac_idx; + struct cs43130_private *cs43130; + struct snd_soc_codec *codec; + struct reg_sequences *hpload_seq; + + cs43130 = container_of(wk, struct cs43130_private, work); + codec = cs43130->codec; + + if (!cs43130->mclk) + return; + + cs43130->hpload_done = false; + + mutex_lock(&cs43130->clk_mutex); + if (!cs43130->clk_req) { + /* clk not in use */ + cs43130_set_pll(codec, 0, 0, cs43130->mclk, CS43130_MCLK_22M); + if (cs43130->pll_bypass) + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_EXT); + else + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_PLL); + } + + cs43130->clk_req++; + mutex_unlock(&cs43130->clk_mutex); + + regmap_read(cs43130->regmap, CS43130_INT_STATUS_4, &reg); + + if (cs43130->dev_id == CS43130_CHIP_ID) { + hpload_seq = hpload_seq1; + seq_size = ARRAY_SIZE(hpload_seq1); + } else { + /* CS434131 */ + hpload_seq = hpload_seq2; + seq_size = ARRAY_SIZE(hpload_seq2); + } + + i = 0; + ac_idx = 0; + while (i < seq_size) { + ret = cs43130_hpload_proc(cs43130, hpload_seq[i].seq, + hpload_seq[i].size, + hpload_seq[i].msk, ac_idx); + if (ret < 0) + goto exit; + + cs43130_update_hpload(hpload_seq[i].msk, ac_idx, cs43130); + + if (cs43130->ac_meas && + hpload_seq[i].msk == CS43130_HPLOAD_AC_INT && + ac_idx < CS43130_AC_FREQ - 1) { + ac_idx++; + } else { + ac_idx = 0; + i++; + } + } + cs43130->hpload_done = true; + + if (cs43130->hpload_dc[0] >= CS43130_LINEOUT_LOAD) + snd_soc_jack_report(&cs43130->jack, CS43130_JACK_LINEOUT, + CS43130_JACK_MASK); + else + snd_soc_jack_report(&cs43130->jack, CS43130_JACK_HEADPHONE, + CS43130_JACK_MASK); + + dev_dbg(codec->dev, "Set HP output control. DC threshold\n"); + for (i = 0; i < CS43130_DC_THRESHOLD; i++) + dev_dbg(codec->dev, "DC threshold[%d]: %u.\n", i, + cs43130->dc_threshold[i]); + + cs43130_set_hv(cs43130->regmap, cs43130->hpload_dc[0], + cs43130->dc_threshold); + +exit: + if (cs43130->dev_id == CS43130_CHIP_ID) + cs43130_hpload_proc(cs43130, hp_dis_cal_seq, + ARRAY_SIZE(hp_dis_cal_seq), + CS43130_HPLOAD_OFF_INT, ac_idx); + else + /* CS434131 */ + cs43130_hpload_proc(cs43130, hp_dis_cal_seq2, + ARRAY_SIZE(hp_dis_cal_seq2), + CS43130_HPLOAD_OFF_INT, ac_idx); + + regmap_multi_reg_write(cs43130->regmap, hp_cln_seq, + ARRAY_SIZE(hp_cln_seq)); + + mutex_lock(&cs43130->clk_mutex); + cs43130->clk_req--; + /* clk not in use */ + if (!cs43130->clk_req) + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_RCO); + mutex_unlock(&cs43130->clk_mutex); +} + +static irqreturn_t cs43130_irq_thread(int irq, void *data) +{ + struct cs43130_private *cs43130 = (struct cs43130_private *)data; + struct snd_soc_codec *codec = cs43130->codec; + unsigned int stickies[CS43130_NUM_INT]; + unsigned int irq_occurrance = 0; + unsigned int masks[CS43130_NUM_INT]; + int i, j; + + for (i = 0; i < ARRAY_SIZE(stickies); i++) { + regmap_read(cs43130->regmap, CS43130_INT_STATUS_1 + i, + &stickies[i]); + regmap_read(cs43130->regmap, CS43130_INT_MASK_1 + i, + &masks[i]); + } + + for (i = 0; i < ARRAY_SIZE(stickies); i++) { + stickies[i] = stickies[i] & (~masks[i]); + for (j = 0; j < 8; j++) + irq_occurrance += (stickies[i] >> j) & 1; + } + dev_dbg(codec->dev, "number of interrupts occurred (%u)\n", + irq_occurrance); + + if (!irq_occurrance) + return IRQ_NONE; + + if (stickies[0] & CS43130_XTAL_RDY_INT) { + complete(&cs43130->xtal_rdy); + return IRQ_HANDLED; + } + + if (stickies[0] & CS43130_PLL_RDY_INT) { + complete(&cs43130->pll_rdy); + return IRQ_HANDLED; + } + + if (stickies[3] & CS43130_HPLOAD_NO_DC_INT) { + cs43130->hpload_stat = stickies[3]; + dev_err(codec->dev, + "DC load has not completed before AC load (%x)\n", + cs43130->hpload_stat); + complete(&cs43130->hpload_evt); + return IRQ_HANDLED; + } + + if (stickies[3] & CS43130_HPLOAD_UNPLUG_INT) { + cs43130->hpload_stat = stickies[3]; + dev_err(codec->dev, "HP unplugged during measurement (%x)\n", + cs43130->hpload_stat); + complete(&cs43130->hpload_evt); + return IRQ_HANDLED; + } + + if (stickies[3] & CS43130_HPLOAD_OOR_INT) { + cs43130->hpload_stat = stickies[3]; + dev_err(codec->dev, "HP load out of range (%x)\n", + cs43130->hpload_stat); + complete(&cs43130->hpload_evt); + return IRQ_HANDLED; + } + + if (stickies[3] & CS43130_HPLOAD_AC_INT) { + cs43130->hpload_stat = stickies[3]; + dev_dbg(codec->dev, "HP AC load measurement done (%x)\n", + cs43130->hpload_stat); + complete(&cs43130->hpload_evt); + return IRQ_HANDLED; + } + + if (stickies[3] & CS43130_HPLOAD_DC_INT) { + cs43130->hpload_stat = stickies[3]; + dev_dbg(codec->dev, "HP DC load measurement done (%x)\n", + cs43130->hpload_stat); + complete(&cs43130->hpload_evt); + return IRQ_HANDLED; + } + + if (stickies[3] & CS43130_HPLOAD_ON_INT) { + cs43130->hpload_stat = stickies[3]; + dev_dbg(codec->dev, "HP load state machine on done (%x)\n", + cs43130->hpload_stat); + complete(&cs43130->hpload_evt); + return IRQ_HANDLED; + } + + if (stickies[3] & CS43130_HPLOAD_OFF_INT) { + cs43130->hpload_stat = stickies[3]; + dev_dbg(codec->dev, "HP load state machine off done (%x)\n", + cs43130->hpload_stat); + complete(&cs43130->hpload_evt); + return IRQ_HANDLED; + } + + if (stickies[0] & CS43130_XTAL_ERR_INT) { + dev_err(codec->dev, "Crystal err: clock is not running\n"); + return IRQ_HANDLED; + } + + if (stickies[0] & CS43130_HP_UNPLUG_INT) { + dev_dbg(codec->dev, "HP unplugged\n"); + cs43130->hpload_done = false; + snd_soc_jack_report(&cs43130->jack, 0, CS43130_JACK_MASK); + return IRQ_HANDLED; + } + + if (stickies[0] & CS43130_HP_PLUG_INT) { + if (cs43130->dc_meas && !cs43130->hpload_done && + !work_busy(&cs43130->work)) { + dev_dbg(codec->dev, "HP load queue work\n"); + queue_work(cs43130->wq, &cs43130->work); + } + + snd_soc_jack_report(&cs43130->jack, SND_JACK_MECHANICAL, + CS43130_JACK_MASK); + return IRQ_HANDLED; + } + + return IRQ_NONE; +} + +static int cs43130_probe(struct snd_soc_codec *codec) +{ + int ret; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_card *card = codec->component.card; + unsigned int reg; + + cs43130->codec = codec; + + if (cs43130->xtal_ibias != CS43130_XTAL_UNUSED) { + regmap_update_bits(cs43130->regmap, CS43130_CRYSTAL_SET, + CS43130_XTAL_IBIAS_MASK, + cs43130->xtal_ibias); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_ERR_INT, 0); + } + + if (!cs43130->idle_bias_off) { + ret = snd_soc_card_jack_new(card, "Headphone", + CS43130_JACK_MASK, + &cs43130->jack, NULL, 0); + if (ret < 0) { + dev_err(codec->dev, "Cannot create jack\n"); + return ret; + } + + cs43130->hpload_done = false; + if (cs43130->dc_meas) { + ret = device_create_file(codec->dev, &dev_attr_hpload); + if (ret < 0) { + dev_err(codec->dev, + "Cannot create HP load file\n"); + return ret; + } + + cs43130->wq = + create_singlethread_workqueue("cs43130_hp"); + INIT_WORK(&cs43130->work, cs43130_imp_meas); + } + + regmap_read(cs43130->regmap, CS43130_INT_STATUS_1, &reg); + regmap_read(cs43130->regmap, CS43130_HP_STATUS, &reg); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_HP_PLUG_INT | + CS43130_HP_UNPLUG_INT, 0); + regmap_update_bits(cs43130->regmap, CS43130_HP_DETECT, + CS43130_HP_DETECT_CTRL_MASK, 0); + regmap_update_bits(cs43130->regmap, CS43130_HP_DETECT, + CS43130_HP_DETECT_CTRL_MASK, + CS43130_HP_DETECT_CTRL_MASK); + } + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs43130 = { + .probe = cs43130_probe, + .component_driver = { + .controls = cs43130_snd_controls, + .num_controls = ARRAY_SIZE(cs43130_snd_controls), + }, + .set_sysclk = cs43130_codec_set_sysclk, + .set_pll = cs43130_set_pll, +}; + +static const struct regmap_config cs43130_regmap = { + .reg_bits = 24, + .pad_bits = 8, + .val_bits = 8, + + .max_register = CS43130_LASTREG, + .reg_defaults = cs43130_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs43130_reg_defaults), + .readable_reg = cs43130_readable_register, + .precious_reg = cs43130_precious_register, + .volatile_reg = cs43130_volatile_register, + .cache_type = REGCACHE_RBTREE, + .use_single_rw = true, /* needed for regcache_sync */ +}; + +static u16 const cs43130_ac_freq[CS43130_AC_FREQ] = { + 24, + 43, + 93, + 200, + 431, + 928, + 2000, + 4309, + 9283, + 20000, +}; + +static u16 const cs43130_dc_threshold[CS43130_DC_THRESHOLD] = { + 50, + 120, +}; + +static int cs43130_handle_device_data(struct i2c_client *i2c_client, + struct cs43130_private *cs43130) +{ + struct device_node *np = i2c_client->dev.of_node; + unsigned int val; + int i; + + if (of_property_read_u32(np, "cirrus,xtal-ibias", &val) < 0) { + /* Crystal is unused. System clock is used for external MCLK */ + cs43130->xtal_ibias = CS43130_XTAL_UNUSED; + return 0; + } + + switch (val) { + case 1: + cs43130->xtal_ibias = CS43130_XTAL_IBIAS_7_5UA; + break; + case 2: + cs43130->xtal_ibias = CS43130_XTAL_IBIAS_12_5UA; + break; + case 3: + cs43130->xtal_ibias = CS43130_XTAL_IBIAS_15UA; + break; + default: + dev_err(&i2c_client->dev, + "Invalid cirrus,xtal-ibias value: %d\n", val); + return -EINVAL; + } + + cs43130->dc_meas = of_property_read_bool(np, "cirrus,dc-measure"); + cs43130->ac_meas = of_property_read_bool(np, "cirrus,ac-measure"); + + if (of_property_read_u16_array(np, "cirrus,ac-freq", cs43130->ac_freq, + CS43130_AC_FREQ) < 0) { + for (i = 0; i < CS43130_AC_FREQ; i++) + cs43130->ac_freq[i] = cs43130_ac_freq[i]; + } + + if (of_property_read_u16_array(np, "cirrus,dc-threshold", + cs43130->dc_threshold, + CS43130_DC_THRESHOLD) < 0) { + for (i = 0; i < CS43130_DC_THRESHOLD; i++) + cs43130->dc_threshold[i] = cs43130_dc_threshold[i]; + } + + cs43130->idle_bias_off = of_property_read_bool(np, + "cirrus,idle-bias-off"); + + return 0; +} + +static int cs43130_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs43130_private *cs43130; + int ret; + unsigned int devid = 0; + unsigned int reg; + int i; + + cs43130 = devm_kzalloc(&client->dev, sizeof(*cs43130), GFP_KERNEL); + if (!cs43130) + return -ENOMEM; + + i2c_set_clientdata(client, cs43130); + + cs43130->regmap = devm_regmap_init_i2c(client, &cs43130_regmap); + if (IS_ERR(cs43130->regmap)) { + ret = PTR_ERR(cs43130->regmap); + return ret; + } + + if (client->dev.of_node) { + ret = cs43130_handle_device_data(client, cs43130); + if (ret != 0) + return ret; + } + for (i = 0; i < ARRAY_SIZE(cs43130->supplies); i++) + cs43130->supplies[i].supply = cs43130_supply_names[i]; + + ret = devm_regulator_bulk_get(&client->dev, + ARRAY_SIZE(cs43130->supplies), + cs43130->supplies); + if (ret != 0) { + dev_err(&client->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + ret = regulator_bulk_enable(ARRAY_SIZE(cs43130->supplies), + cs43130->supplies); + if (ret != 0) { + dev_err(&client->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + cs43130->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs43130->reset_gpio)) + return PTR_ERR(cs43130->reset_gpio); + + gpiod_set_value_cansleep(cs43130->reset_gpio, 1); + + usleep_range(2000, 2050); + + ret = regmap_read(cs43130->regmap, CS43130_DEVID_AB, &reg); + + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs43130->regmap, CS43130_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs43130->regmap, CS43130_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + switch (devid) { + case CS43130_CHIP_ID: + case CS4399_CHIP_ID: + case CS43131_CHIP_ID: + case CS43198_CHIP_ID: + break; + default: + dev_err(&client->dev, + "CS43130 Device ID %X. Expected ID %X, %X, %X or %X\n", + devid, CS43130_CHIP_ID, CS4399_CHIP_ID, + CS43131_CHIP_ID, CS43198_CHIP_ID); + ret = -ENODEV; + goto err; + } + + cs43130->dev_id = devid; + ret = regmap_read(cs43130->regmap, CS43130_REV_ID, &reg); + if (ret < 0) { + dev_err(&client->dev, "Get Revision ID failed\n"); + goto err; + } + + dev_info(&client->dev, + "Cirrus Logic CS43130 (%x), Revision: %02X\n", devid, + reg & 0xFF); + + mutex_init(&cs43130->clk_mutex); + + init_completion(&cs43130->xtal_rdy); + init_completion(&cs43130->pll_rdy); + init_completion(&cs43130->hpload_evt); + + ret = devm_request_threaded_irq(&client->dev, client->irq, + NULL, cs43130_irq_thread, + IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs43130", cs43130); + if (ret != 0) { + dev_err(&client->dev, "Failed to request IRQ: %d\n", ret); + return ret; + } + + cs43130->mclk_int_src = CS43130_MCLK_SRC_RCO; + + soc_codec_dev_cs43130.idle_bias_off = cs43130->idle_bias_off; + + if (cs43130->idle_bias_off) { + pm_runtime_set_autosuspend_delay(&client->dev, 100); + pm_runtime_use_autosuspend(&client->dev); + pm_runtime_set_active(&client->dev); + pm_runtime_enable(&client->dev); + } + + if (cs43130->dev_id == CS43130_CHIP_ID || + cs43130->dev_id == CS43131_CHIP_ID) { + memcpy(all_hp_widgets, digital_hp_widgets, + sizeof(digital_hp_widgets)); + memcpy(all_hp_widgets + ARRAY_SIZE(digital_hp_widgets), + analog_hp_widgets, sizeof(analog_hp_widgets)); + memcpy(all_hp_routes, digital_hp_routes, + sizeof(digital_hp_routes)); + memcpy(all_hp_routes + ARRAY_SIZE(digital_hp_routes), + analog_hp_routes, sizeof(analog_hp_routes)); + + soc_codec_dev_cs43130.component_driver.dapm_widgets = + all_hp_widgets; + soc_codec_dev_cs43130.component_driver.num_dapm_widgets = + ARRAY_SIZE(all_hp_widgets); + soc_codec_dev_cs43130.component_driver.dapm_routes = + all_hp_routes; + soc_codec_dev_cs43130.component_driver.num_dapm_routes = + ARRAY_SIZE(all_hp_routes); + } else { + /* CS43198 or CS4399 */ + soc_codec_dev_cs43130.component_driver.dapm_widgets = + digital_hp_widgets; + soc_codec_dev_cs43130.component_driver.num_dapm_widgets = + ARRAY_SIZE(digital_hp_widgets); + soc_codec_dev_cs43130.component_driver.dapm_routes = + digital_hp_routes; + soc_codec_dev_cs43130.component_driver.num_dapm_routes = + ARRAY_SIZE(digital_hp_routes); + } + ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_cs43130, + cs43130_dai, ARRAY_SIZE(cs43130_dai)); + + if (ret < 0) { + dev_err(&client->dev, + "snd_soc_register_codec failed with ret = %d\n", ret); + goto err; + } + + regmap_update_bits(cs43130->regmap, CS43130_PAD_INT_CFG, + CS43130_ASP_3ST_MASK, 0); + regmap_update_bits(cs43130->regmap, CS43130_PAD_INT_CFG, + CS43130_XSP_3ST_MASK, 0); + + return 0; +err: + return ret; +} + +static int cs43130_i2c_remove(struct i2c_client *client) +{ + struct cs43130_private *cs43130 = i2c_get_clientdata(client); + + if (cs43130->xtal_ibias != CS43130_XTAL_UNUSED) + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_ERR_INT, + 1 << CS43130_XTAL_ERR_INT_SHIFT); + + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_HP_PLUG_INT | CS43130_HP_UNPLUG_INT, + CS43130_HP_PLUG_INT | CS43130_HP_UNPLUG_INT); + + if (cs43130->dc_meas && !cs43130->idle_bias_off) { + cancel_work_sync(&cs43130->work); + flush_workqueue(cs43130->wq); + + device_remove_file(&client->dev, &dev_attr_hpload); + } + + if (cs43130->reset_gpio) + gpiod_set_value_cansleep(cs43130->reset_gpio, 0); + + pm_runtime_disable(&client->dev); + regulator_bulk_disable(CS43130_NUM_SUPPLIES, cs43130->supplies); + + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static int cs43130_runtime_suspend(struct device *dev) +{ + struct cs43130_private *cs43130 = dev_get_drvdata(dev); + + if (cs43130->xtal_ibias != CS43130_XTAL_UNUSED) + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_ERR_INT, + 1 << CS43130_XTAL_ERR_INT_SHIFT); + + regcache_cache_only(cs43130->regmap, true); + regcache_mark_dirty(cs43130->regmap); + + gpiod_set_value_cansleep(cs43130->reset_gpio, 0); + + regulator_bulk_disable(CS43130_NUM_SUPPLIES, cs43130->supplies); + + return 0; +} + +static int cs43130_runtime_resume(struct device *dev) +{ + struct cs43130_private *cs43130 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(CS43130_NUM_SUPPLIES, cs43130->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(cs43130->regmap, false); + + gpiod_set_value_cansleep(cs43130->reset_gpio, 1); + + usleep_range(2000, 2050); + + ret = regcache_sync(cs43130->regmap); + if (ret != 0) { + dev_err(dev, "Failed to restore register cache\n"); + goto err; + } + + if (cs43130->xtal_ibias != CS43130_XTAL_UNUSED) + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_ERR_INT, 0); + + return 0; +err: + regcache_cache_only(cs43130->regmap, true); + regulator_bulk_disable(CS43130_NUM_SUPPLIES, cs43130->supplies); + + return ret; +} + +static const struct dev_pm_ops cs43130_runtime_pm = { + SET_RUNTIME_PM_OPS(cs43130_runtime_suspend, cs43130_runtime_resume, + NULL) +}; + +static const struct of_device_id cs43130_of_match[] = { + {.compatible = "cirrus,cs43130",}, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs43130_of_match); + +static const struct i2c_device_id cs43130_i2c_id[] = { + {"cs43130", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs43130_i2c_id); + +static struct i2c_driver cs43130_i2c_driver = { + .driver = { + .name = "cs43130", + .of_match_table = cs43130_of_match, + .pm = &cs43130_runtime_pm, + }, + .id_table = cs43130_i2c_id, + .probe = cs43130_i2c_probe, + .remove = cs43130_i2c_remove, +}; + +module_i2c_driver(cs43130_i2c_driver); + +MODULE_AUTHOR("Li Xu <li.xu(a)cirrus.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS43130 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h new file mode 100644 index 0000000..5e7a852 --- /dev/null +++ b/sound/soc/codecs/cs43130.h @@ -0,0 +1,545 @@ +/* + * ALSA SoC CS43130 codec driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Li Xu <li.xu(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#ifndef __CS43130_H__ +#define __CS43130_H__ + +/* CS43130 registers addresses */ +/* all reg address is shifted by a byte for control byte to be LSB */ +#define CS43130_FIRSTREG 0x010000 +#define CS43130_LASTREG 0x190000 +#define CS43130_CHIP_ID 0x00043130 +#define CS4399_CHIP_ID 0x00043990 +#define CS43131_CHIP_ID 0x00043131 +#define CS43198_CHIP_ID 0x00043198 +#define CS43130_DEVID_AB 0x010000 /* Device ID A & B [RO] */ +#define CS43130_DEVID_CD 0x010001 /* Device ID C & D [RO] */ +#define CS43130_DEVID_E 0x010002 /* Device ID E [RO] */ +#define CS43130_FAB_ID 0x010003 /* Fab ID [RO] */ +#define CS43130_REV_ID 0x010004 /* Revision ID [RO] */ +#define CS43130_SUBREV_ID 0x010005 /* Subrevision ID */ +#define CS43130_SYS_CLK_CTL_1 0x010006 /* System Clocking Ctl 1 */ +#define CS43130_SP_SRATE 0x01000B /* Serial Port Sample Rate */ +#define CS43130_SP_BITSIZE 0x01000C /* Serial Port Bit Size */ +#define CS43130_PAD_INT_CFG 0x01000D /* Pad Interface Config */ +#define CS43130_DXD1 0x010010 /* DXD1 */ +#define CS43130_DXD7 0x010025 /* DXD7 */ +#define CS43130_DXD19 0x010026 /* DXD19 */ +#define CS43130_DXD17 0x010027 /* DXD17 */ +#define CS43130_DXD18 0x010028 /* DXD18 */ +#define CS43130_DXD12 0x01002C /* DXD12 */ +#define CS43130_DXD8 0x01002E /* DXD8 */ +#define CS43130_PWDN_CTL 0x020000 /* Power Down Ctl */ +#define CS43130_DXD2 0x020019 /* DXD2 */ +#define CS43130_CRYSTAL_SET 0x020052 /* Crystal Setting */ +#define CS43130_PLL_SET_1 0x030001 /* PLL Setting 1 */ +#define CS43130_PLL_SET_2 0x030002 /* PLL Setting 2 */ +#define CS43130_PLL_SET_3 0x030003 /* PLL Setting 3 */ +#define CS43130_PLL_SET_4 0x030004 /* PLL Setting 4 */ +#define CS43130_PLL_SET_5 0x030005 /* PLL Setting 5 */ +#define CS43130_PLL_SET_6 0x030008 /* PLL Setting 6 */ +#define CS43130_PLL_SET_7 0x03000A /* PLL Setting 7 */ +#define CS43130_PLL_SET_8 0x03001B /* PLL Setting 8 */ +#define CS43130_PLL_SET_9 0x040002 /* PLL Setting 9 */ +#define CS43130_PLL_SET_10 0x040003 /* PLL Setting 10 */ +#define CS43130_CLKOUT_CTL 0x040004 /* CLKOUT Ctl */ +#define CS43130_ASP_NUM_1 0x040010 /* ASP Numerator 1 */ +#define CS43130_ASP_NUM_2 0x040011 /* ASP Numerator 2 */ +#define CS43130_ASP_DEN_1 0x040012 /* ASP Denominator 1 */ +#define CS43130_ASP_DEN_2 0x040013 /* ASP Denominator 2 */ +#define CS43130_ASP_LRCK_HI_TIME_1 0x040014 /* ASP LRCK High Time 1 */ +#define CS43130_ASP_LRCK_HI_TIME_2 0x040015 /* ASP LRCK High Time 2 */ +#define CS43130_ASP_LRCK_PERIOD_1 0x040016 /* ASP LRCK Period 1 */ +#define CS43130_ASP_LRCK_PERIOD_2 0x040017 /* ASP LRCK Period 2 */ +#define CS43130_ASP_CLOCK_CONF 0x040018 /* ASP Clock Config */ +#define CS43130_ASP_FRAME_CONF 0x040019 /* ASP Frame Config */ +#define CS43130_XSP_NUM_1 0x040020 /* XSP Numerator 1 */ +#define CS43130_XSP_NUM_2 0x040021 /* XSP Numerator 2 */ +#define CS43130_XSP_DEN_1 0x040022 /* XSP Denominator 1 */ +#define CS43130_XSP_DEN_2 0x040023 /* XSP Denominator 2 */ +#define CS43130_XSP_LRCK_HI_TIME_1 0x040024 /* XSP LRCK High Time 1 */ +#define CS43130_XSP_LRCK_HI_TIME_2 0x040025 /* XSP LRCK High Time 2 */ +#define CS43130_XSP_LRCK_PERIOD_1 0x040026 /* XSP LRCK Period 1 */ +#define CS43130_XSP_LRCK_PERIOD_2 0x040027 /* XSP LRCK Period 2 */ +#define CS43130_XSP_CLOCK_CONF 0x040028 /* XSP Clock Config */ +#define CS43130_XSP_FRAME_CONF 0x040029 /* XSP Frame Config */ +#define CS43130_ASP_CH_1_LOC 0x050000 /* ASP Chan 1 Location */ +#define CS43130_ASP_CH_2_LOC 0x050001 /* ASP Chan 2 Location */ +#define CS43130_ASP_CH_1_SZ_EN 0x05000A /* ASP Chan 1 Size, Enable */ +#define CS43130_ASP_CH_2_SZ_EN 0x05000B /* ASP Chan 2 Size, Enable */ +#define CS43130_XSP_CH_1_LOC 0x060000 /* XSP Chan 1 Location */ +#define CS43130_XSP_CH_2_LOC 0x060001 /* XSP Chan 2 Location */ +#define CS43130_XSP_CH_1_SZ_EN 0x06000A /* XSP Chan 1 Size, Enable */ +#define CS43130_XSP_CH_2_SZ_EN 0x06000B /* XSP Chan 2 Size, Enable */ +#define CS43130_DSD_VOL_B 0x070000 /* DSD Volume B */ +#define CS43130_DSD_VOL_A 0x070001 /* DSD Volume A */ +#define CS43130_DSD_PATH_CTL_1 0x070002 /* DSD Proc Path Sig Ctl 1 */ +#define CS43130_DSD_INT_CFG 0x070003 /* DSD Interface Config */ +#define CS43130_DSD_PATH_CTL_2 0x070004 /* DSD Proc Path Sig Ctl 2 */ +#define CS43130_DSD_PCM_MIX_CTL 0x070005 /* DSD and PCM Mixing Ctl */ +#define CS43130_DSD_PATH_CTL_3 0x070006 /* DSD Proc Path Sig Ctl 3 */ +#define CS43130_HP_OUT_CTL_1 0x080000 /* HP Output Ctl 1 */ +#define CS43130_DXD16 0x080024 /* DXD16 */ +#define CS43130_DXD13 0x080032 /* DXD13 */ +#define CS43130_PCM_FILT_OPT 0x090000 /* PCM Filter Option */ +#define CS43130_PCM_VOL_B 0x090001 /* PCM Volume B */ +#define CS43130_PCM_VOL_A 0x090002 /* PCM Volume A */ +#define CS43130_PCM_PATH_CTL_1 0x090003 /* PCM Path Signal Ctl 1 */ +#define CS43130_PCM_PATH_CTL_2 0x090004 /* PCM Path Signal Ctl 2 */ +#define CS43130_DXD6 0x090097 /* DXD6 */ +#define CS43130_CLASS_H_CTL 0x0B0000 /* Class H Ctl */ +#define CS43130_DXD15 0x0B0005 /* DXD15 */ +#define CS43130_DXD14 0x0B0006 /* DXD14 */ +#define CS43130_DXD3 0x0C0002 /* DXD3 */ +#define CS43130_DXD10 0x0C0003 /* DXD10 */ +#define CS43130_DXD11 0x0C0005 /* DXD11 */ +#define CS43130_DXD9 0x0C0006 /* DXD9 */ +#define CS43130_DXD4 0x0C0009 /* DXD4 */ +#define CS43130_DXD5 0x0C000E /* DXD5 */ +#define CS43130_HP_DETECT 0x0D0000 /* HP Detect */ +#define CS43130_HP_STATUS 0x0D0001 /* HP Status [RO] */ +#define CS43130_HP_LOAD_1 0x0E0000 /* HP Load 1 */ +#define CS43130_HP_MEAS_LOAD_1 0x0E0003 /* HP Load Measurement 1 */ +#define CS43130_HP_MEAS_LOAD_2 0x0E0004 /* HP Load Measurement 2 */ +#define CS43130_HP_DC_STAT_1 0x0E000D /* HP DC Load Status 0 [RO] */ +#define CS43130_HP_DC_STAT_2 0x0E000E /* HP DC Load Status 1 [RO] */ +#define CS43130_HP_AC_STAT_1 0x0E0010 /* HP AC Load Status 0 [RO] */ +#define CS43130_HP_AC_STAT_2 0x0E0011 /* HP AC Load Status 1 [RO] */ +#define CS43130_HP_LOAD_STAT 0x0E001A /* HP Load Status [RO] */ +#define CS43130_INT_STATUS_1 0x0F0000 /* Interrupt Status 1 */ +#define CS43130_INT_STATUS_2 0x0F0001 /* Interrupt Status 2 */ +#define CS43130_INT_STATUS_3 0x0F0002 /* Interrupt Status 3 */ +#define CS43130_INT_STATUS_4 0x0F0003 /* Interrupt Status 4 */ +#define CS43130_INT_STATUS_5 0x0F0004 /* Interrupt Status 5 */ +#define CS43130_INT_MASK_1 0x0F0010 /* Interrupt Mask 1 */ +#define CS43130_INT_MASK_2 0x0F0011 /* Interrupt Mask 2 */ +#define CS43130_INT_MASK_3 0x0F0012 /* Interrupt Mask 3 */ +#define CS43130_INT_MASK_4 0x0F0013 /* Interrupt Mask 4 */ +#define CS43130_INT_MASK_5 0x0F0014 /* Interrupt Mask 5 */ + +#define CS43130_MCLK_SRC_SEL_MASK 0x03 +#define CS43130_MCLK_SRC_SEL_SHIFT 0 +#define CS43130_MCLK_INT_MASK 0x04 +#define CS43130_MCLK_INT_SHIFT 2 +#define CS43130_CH_BITSIZE_MASK 0x03 +#define CS43130_CH_EN_MASK 0x04 +#define CS43130_CH_EN_SHIFT 2 +#define CS43130_ASP_BITSIZE_MASK 0x03 +#define CS43130_XSP_BITSIZE_MASK 0x0C +#define CS43130_XSP_BITSIZE_SHIFT 2 +#define CS43130_SP_BITSIZE_ASP_SHIFT 0 +#define CS43130_HP_DETECT_CTRL_SHIFT 6 +#define CS43130_HP_DETECT_CTRL_MASK (0x03 << CS43130_HP_DETECT_CTRL_SHIFT) +#define CS43130_HP_DETECT_INV_SHIFT 5 +#define CS43130_HP_DETECT_INV_MASK (1 << CS43130_HP_DETECT_INV_SHIFT) + +/* CS43130_INT_MASK_1 */ +#define CS43130_HP_PLUG_INT_SHIFT 6 +#define CS43130_HP_PLUG_INT (1 << CS43130_HP_PLUG_INT_SHIFT) +#define CS43130_HP_UNPLUG_INT_SHIFT 5 +#define CS43130_HP_UNPLUG_INT (1 << CS43130_HP_UNPLUG_INT_SHIFT) +#define CS43130_XTAL_RDY_INT_SHIFT 4 +#define CS43130_XTAL_RDY_INT_MASK 0x10 +#define CS43130_XTAL_RDY_INT (1 << CS43130_XTAL_RDY_INT_SHIFT) +#define CS43130_XTAL_ERR_INT_SHIFT 3 +#define CS43130_XTAL_ERR_INT (1 << CS43130_XTAL_ERR_INT_SHIFT) +#define CS43130_PLL_RDY_INT_MASK 0x04 +#define CS43130_PLL_RDY_INT_SHIFT 2 +#define CS43130_PLL_RDY_INT (1 << CS43130_PLL_RDY_INT_SHIFT) + +/* CS43130_INT_MASK_4 */ +#define CS43130_INT_MASK_ALL 0xFF +#define CS43130_HPLOAD_NO_DC_INT_SHIFT 7 +#define CS43130_HPLOAD_NO_DC_INT (1 << CS43130_HPLOAD_NO_DC_INT_SHIFT) +#define CS43130_HPLOAD_UNPLUG_INT_SHIFT 6 +#define CS43130_HPLOAD_UNPLUG_INT (1 << CS43130_HPLOAD_UNPLUG_INT_SHIFT) +#define CS43130_HPLOAD_OOR_INT_SHIFT 4 +#define CS43130_HPLOAD_OOR_INT (1 << CS43130_HPLOAD_OOR_INT_SHIFT) +#define CS43130_HPLOAD_AC_INT_SHIFT 3 +#define CS43130_HPLOAD_AC_INT (1 << CS43130_HPLOAD_AC_INT_SHIFT) +#define CS43130_HPLOAD_DC_INT_SHIFT 2 +#define CS43130_HPLOAD_DC_INT (1 << CS43130_HPLOAD_DC_INT_SHIFT) +#define CS43130_HPLOAD_OFF_INT_SHIFT 1 +#define CS43130_HPLOAD_OFF_INT (1 << CS43130_HPLOAD_OFF_INT_SHIFT) +#define CS43130_HPLOAD_ON_INT 1 + +/* CS43130_HP_LOAD_1 */ +#define CS43130_HPLOAD_EN_SHIFT 7 +#define CS43130_HPLOAD_EN (1 << CS43130_HPLOAD_EN_SHIFT) +#define CS43130_HPLOAD_CHN_SEL_SHIFT 4 +#define CS43130_HPLOAD_CHN_SEL (1 << CS43130_HPLOAD_CHN_SEL_SHIFT) +#define CS43130_HPLOAD_AC_START_SHIFT 1 +#define CS43130_HPLOAD_AC_START (1 << CS43130_HPLOAD_AC_START_SHIFT) +#define CS43130_HPLOAD_DC_START 1 + +/* Reg CS43130_SP_BITSIZE */ +#define CS43130_SP_BIT_SIZE_8 0x03 +#define CS43130_SP_BIT_SIZE_16 0x02 +#define CS43130_SP_BIT_SIZE_24 0x01 +#define CS43130_SP_BIT_SIZE_32 0x00 + +/* Reg CS43130_SP_CH_SZ_EN */ +#define CS43130_CH_BIT_SIZE_8 0x00 +#define CS43130_CH_BIT_SIZE_16 0x01 +#define CS43130_CH_BIT_SIZE_24 0x02 +#define CS43130_CH_BIT_SIZE_32 0x03 + +/* PLL */ +#define CS43130_PLL_START_MASK 0x01 +#define CS43130_PLL_MODE_MASK 0x02 +#define CS43130_PLL_MODE_SHIFT 1 + +#define CS43130_PLL_REF_PREDIV_MASK 0x3 + +#define CS43130_SP_STP_MASK 0x10 +#define CS43130_SP_STP_SHIFT 4 +#define CS43130_SP_5050_MASK 0x08 +#define CS43130_SP_5050_SHIFT 3 +#define CS43130_SP_FSD_MASK 0x07 + +#define CS43130_SP_MODE_MASK 0x10 +#define CS43130_SP_MODE_SHIFT 4 +#define CS43130_SP_SCPOL_OUT_MASK 0x08 +#define CS43130_SP_SCPOL_OUT_SHIFT 3 +#define CS43130_SP_SCPOL_IN_MASK 0x04 +#define CS43130_SP_SCPOL_IN_SHIFT 2 +#define CS43130_SP_LCPOL_OUT_MASK 0x02 +#define CS43130_SP_LCPOL_OUT_SHIFT 1 +#define CS43130_SP_LCPOL_IN_MASK 0x01 +#define CS43130_SP_LCPOL_IN_SHIFT 0 + +/* Reg CS43130_PWDN_CTL */ +#define CS43130_PDN_XSP_MASK 0x80 +#define CS43130_PDN_XSP_SHIFT 7 +#define CS43130_PDN_ASP_MASK 0x40 +#define CS43130_PDN_ASP_SHIFT 6 +#define CS43130_PDN_DSPIF_MASK 0x20 +#define CS43130_PDN_DSDIF_SHIFT 5 +#define CS43130_PDN_HP_MASK 0x10 +#define CS43130_PDN_HP_SHIFT 4 +#define CS43130_PDN_XTAL_MASK 0x08 +#define CS43130_PDN_XTAL_SHIFT 3 +#define CS43130_PDN_PLL_MASK 0x04 +#define CS43130_PDN_PLL_SHIFT 2 +#define CS43130_PDN_CLKOUT_MASK 0x02 +#define CS43130_PDN_CLKOUT_SHIFT 1 + +/* Reg CS43130_HP_OUT_CTL_1 */ +#define CS43130_HP_IN_EN_SHIFT 3 +#define CS43130_HP_IN_EN_MASK 0x08 + +/* Reg CS43130_PAD_INT_CFG */ +#define CS43130_ASP_3ST_MASK 0x01 +#define CS43130_XSP_3ST_MASK 0x02 + +/* Reg CS43130_PLL_SET_2 */ +#define CS43130_PLL_DIV_DATA_MASK 0x000000FF +#define CS43130_PLL_DIV_FRAC_0_DATA_SHIFT 0 + +/* Reg CS43130_PLL_SET_3 */ +#define CS43130_PLL_DIV_FRAC_1_DATA_SHIFT 8 + +/* Reg CS43130_PLL_SET_4 */ +#define CS43130_PLL_DIV_FRAC_2_DATA_SHIFT 16 + +/* Reg CS43130_SP_DEN_1 */ +#define CS43130_SP_M_LSB_DATA_MASK 0x00FF +#define CS43130_SP_M_LSB_DATA_SHIFT 0 + +/* Reg CS43130_SP_DEN_2 */ +#define CS43130_SP_M_MSB_DATA_MASK 0xFF00 +#define CS43130_SP_M_MSB_DATA_SHIFT 8 + +/* Reg CS43130_SP_NUM_1 */ +#define CS43130_SP_N_LSB_DATA_MASK 0x00FF +#define CS43130_SP_N_LSB_DATA_SHIFT 0 + +/* Reg CS43130_SP_NUM_2 */ +#define CS43130_SP_N_MSB_DATA_MASK 0xFF00 +#define CS43130_SP_N_MSB_DATA_SHIFT 8 + +/* Reg CS43130_SP_LRCK_HI_TIME_1 */ +#define CS43130_SP_LCHI_DATA_MASK 0x00FF +#define CS43130_SP_LCHI_LSB_DATA_SHIFT 0 + +/* Reg CS43130_SP_LRCK_HI_TIME_2 */ +#define CS43130_SP_LCHI_MSB_DATA_SHIFT 8 + +/* Reg CS43130_SP_LRCK_PERIOD_1 */ +#define CS43130_SP_LCPR_DATA_MASK 0x00FF +#define CS43130_SP_LCPR_LSB_DATA_SHIFT 0 + +/* Reg CS43130_SP_LRCK_PERIOD_2 */ +#define CS43130_SP_LCPR_MSB_DATA_SHIFT 8 + +#define CS43130_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define CS43130_DOP_FORMATS (SNDRV_PCM_FMTBIT_DSD_U16_LE | \ + SNDRV_PCM_FMTBIT_DSD_U16_BE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/* Reg CS43130_CRYSTAL_SET */ +#define CS43130_XTAL_IBIAS_MASK 0x07 + +/* Reg CS43130_PATH_CTL_1 */ +#define CS43130_MUTE_MASK 0x03 +#define CS43130_MUTE_EN 0x03 + +/* Reg CS43130_DSD_INT_CFG */ +#define CS43130_DSD_MASTER 0x04 + +/* Reg CS43130_DSD_PATH_CTL_2 */ +#define CS43130_DSD_SRC_MASK 0x60 +#define CS43130_DSD_SRC_SHIFT 5 +#define CS43130_DSD_EN_SHIFT 4 +#define CS43130_DSD_SPEED_MASK 0x04 +#define CS43130_DSD_SPEED_SHIFT 2 + +/* Reg CS43130_DSD_PCM_MIX_CTL */ +#define CS43130_MIX_PCM_PREP_SHIFT 1 +#define CS43130_MIX_PCM_PREP_MASK 0x02 + +#define CS43130_MIX_PCM_DSD_SHIFT 0 +#define CS43130_MIX_PCM_DSD_MASK 0x01 + +/* Reg CS43130_HP_MEAS_LOAD */ +#define CS43130_HP_MEAS_LOAD_MASK 0x000000FF +#define CS43130_HP_MEAS_LOAD_1_SHIFT 0 +#define CS43130_HP_MEAS_LOAD_2_SHIFT 8 + +#define CS43130_MCLK_22M 22579200 +#define CS43130_MCLK_24M 24576000 + +#define CS43130_LINEOUT_LOAD 5000 +#define CS43130_JACK_LINEOUT (SND_JACK_MECHANICAL | SND_JACK_LINEOUT) +#define CS43130_JACK_HEADPHONE (SND_JACK_MECHANICAL | \ + SND_JACK_HEADPHONE) +#define CS43130_JACK_MASK (SND_JACK_MECHANICAL | \ + SND_JACK_LINEOUT | \ + SND_JACK_HEADPHONE) + +enum cs43130_dsd_src { + CS43130_DSD_SRC_DSD = 0, + CS43130_DSD_SRC_ASP = 2, + CS43130_DSD_SRC_XSP = 3, +}; + +enum cs43130_asp_rate { + CS43130_ASP_SPRATE_32K = 0, + CS43130_ASP_SPRATE_44_1K, + CS43130_ASP_SPRATE_48K, + CS43130_ASP_SPRATE_88_2K, + CS43130_ASP_SPRATE_96K, + CS43130_ASP_SPRATE_176_4K, + CS43130_ASP_SPRATE_192K, + CS43130_ASP_SPRATE_352_8K, + CS43130_ASP_SPRATE_384K, +}; + +enum cs43130_mclk_src_sel { + CS43130_MCLK_SRC_EXT = 0, + CS43130_MCLK_SRC_PLL, + CS43130_MCLK_SRC_RCO +}; + +enum cs43130_mclk_int_freq { + CS43130_MCLK_24P5 = 0, + CS43130_MCLK_22P5, +}; + +enum cs43130_xtal_ibias { + CS43130_XTAL_UNUSED = -1, + CS43130_XTAL_IBIAS_15UA = 2, + CS43130_XTAL_IBIAS_12_5UA = 4, + CS43130_XTAL_IBIAS_7_5UA = 6, +}; + +enum cs43130_dai_id { + CS43130_ASP_PCM_DAI = 0, + CS43130_ASP_DOP_DAI, + CS43130_XSP_DOP_DAI, + CS43130_XSP_DSD_DAI, + CS43130_DAI_ID_MAX, +}; + +struct cs43130_clk_gen { + unsigned int mclk_int; + int fs; + u16 den; + u16 num; +}; + +/* frm_size = 16 */ +static const struct cs43130_clk_gen cs43130_16_clk_gen[] = { + {22579200, 32000, 441, 10,}, + {22579200, 44100, 32, 1,}, + {22579200, 48000, 147, 5,}, + {22579200, 88200, 16, 1,}, + {22579200, 96000, 147, 10,}, + {22579200, 176400, 8, 1,}, + {22579200, 192000, 147, 20,}, + {22579200, 352800, 4, 1,}, + {22579200, 384000, 147, 40,}, + {24576000, 32000, 48, 1,}, + {24576000, 44100, 5120, 147,}, + {24576000, 48000, 32, 1,}, + {24576000, 88200, 2560, 147,}, + {24576000, 96000, 16, 1,}, + {24576000, 176400, 1280, 147,}, + {24576000, 192000, 8, 1,}, + {24576000, 352800, 640, 147,}, + {24576000, 384000, 4, 1,}, +}; + +/* frm_size = 32 */ +static const struct cs43130_clk_gen cs43130_32_clk_gen[] = { + {22579200, 32000, 441, 20,}, + {22579200, 44100, 16, 1,}, + {22579200, 48000, 147, 10,}, + {22579200, 88200, 8, 1,}, + {22579200, 96000, 147, 20,}, + {22579200, 176400, 4, 1,}, + {22579200, 192000, 147, 40,}, + {22579200, 352800, 2, 1,}, + {22579200, 384000, 147, 80,}, + {24576000, 32000, 24, 1,}, + {24576000, 44100, 2560, 147,}, + {24576000, 48000, 16, 1,}, + {24576000, 88200, 1280, 147,}, + {24576000, 96000, 8, 1,}, + {24576000, 176400, 640, 147,}, + {24576000, 192000, 4, 1,}, + {24576000, 352800, 320, 147,}, + {24576000, 384000, 2, 1,}, +}; + +/* frm_size = 48 */ +static const struct cs43130_clk_gen cs43130_48_clk_gen[] = { + {22579200, 32000, 147, 100,}, + {22579200, 44100, 32, 3,}, + {22579200, 48000, 49, 5,}, + {22579200, 88200, 16, 3,}, + {22579200, 96000, 49, 10,}, + {22579200, 176400, 8, 3,}, + {22579200, 192000, 49, 20,}, + {22579200, 352800, 4, 3,}, + {22579200, 384000, 49, 40,}, + {24576000, 32000, 16, 1,}, + {24576000, 44100, 5120, 441,}, + {24576000, 48000, 32, 3,}, + {24576000, 88200, 2560, 441,}, + {24576000, 96000, 16, 3,}, + {24576000, 176400, 1280, 441,}, + {24576000, 192000, 8, 3,}, + {24576000, 352800, 640, 441,}, + {24576000, 384000, 4, 3,}, +}; + +/* frm_size = 64 */ +static const struct cs43130_clk_gen cs43130_64_clk_gen[] = { + {22579200, 32000, 441, 40,}, + {22579200, 44100, 8, 1,}, + {22579200, 48000, 147, 20,}, + {22579200, 88200, 4, 1,}, + {22579200, 96000, 147, 40,}, + {22579200, 176400, 2, 1,}, + {22579200, 192000, 147, 80,}, + {22579200, 352800, 1, 1,}, + {24576000, 32000, 12, 1,}, + {24576000, 44100, 1280, 147,}, + {24576000, 48000, 8, 1,}, + {24576000, 88200, 640, 147,}, + {24576000, 96000, 4, 1,}, + {24576000, 176400, 320, 147,}, + {24576000, 192000, 2, 1,}, + {24576000, 352800, 160, 147,}, + {24576000, 384000, 1, 1,}, +}; + +struct cs43130_bitwidth_map { + unsigned int bitwidth; + u8 sp_bit; + u8 ch_bit; +}; + +struct cs43130_rate_map { + int fs; + int val; +}; + +#define CS43130_AC_FREQ 10 +#define CS43130_DC_THRESHOLD 2 + +#define CS43130_NUM_SUPPLIES 5 +static const char *const cs43130_supply_names[CS43130_NUM_SUPPLIES] = { + "VA", + "VP", + "VCP", + "VD", + "VL", +}; + +#define CS43130_NUM_INT 5 /* number of interrupt status reg */ + +struct cs43130_dai { + unsigned int sclk; + unsigned int dai_format; + unsigned int dai_mode; +}; + +struct cs43130_private { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct regulator_bulk_data supplies[CS43130_NUM_SUPPLIES]; + struct gpio_desc *reset_gpio; + unsigned int dev_id; /* codec device ID */ + int xtal_ibias; + bool idle_bias_off; + + /* shared by both DAIs */ + struct mutex clk_mutex; + int clk_req; + bool pll_bypass; + struct completion xtal_rdy; + struct completion pll_rdy; + unsigned int mclk; + unsigned int mclk_int; + int mclk_int_src; + + /* DAI specific */ + struct cs43130_dai dais[CS43130_DAI_ID_MAX]; + + /* HP load specific */ + bool dc_meas; + bool ac_meas; + bool hpload_done; + struct completion hpload_evt; + unsigned int hpload_stat; + u16 hpload_dc[2]; + u16 dc_threshold[CS43130_DC_THRESHOLD]; + u16 ac_freq[CS43130_AC_FREQ]; + u16 hpload_ac[CS43130_AC_FREQ][2]; + struct workqueue_struct *wq; + struct work_struct work; + struct snd_soc_jack jack; +}; + +#endif /* __CS43130_H__ */ -- 1.9.1

2 2

[alsa-devel] [PATCH] sound: fix some checkpatch errors and warnings.
by Peng Donglin 18 Aug '17

18 Aug '17

ERROR: trailing whitespace WARNING: please, no spaces at the start of a line ERROR: open brace '{' following struct go on the same line ERROR: do not initialise statics to 0 ERROR: "foo * bar" should be "foo *bar" ERROR: spaces required around that '=' ERROR: switch and case should be at the same indent WARNING: Statements should start on a tabstop WARNING: Block comments use * on subsequent lines Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/last.c | 2 +- sound/sound_core.c | 163 ++++++++++++++++++++++++----------------------------- 2 files changed, 76 insertions(+), 89 deletions(-) diff --git a/sound/last.c b/sound/last.c index 43f2228..a47bbe0 100644 --- a/sound/last.c +++ b/sound/last.c @@ -25,7 +25,7 @@ static int __init alsa_sound_last_init(void) { int idx, ok = 0; - + printk(KERN_INFO "ALSA device list:\n"); for (idx = 0; idx < SNDRV_CARDS; idx++) if (snd_cards[idx] != NULL) { diff --git a/sound/sound_core.c b/sound/sound_core.c index 99b73c6..f99bb7a 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -68,7 +68,7 @@ static void __exit cleanup_soundcore(void) #ifdef CONFIG_SOUND_OSS_CORE /* * OSS sound core handling. Breaks out sound functions to submodules - * + * * Author: Alan Cox <alan(a)lxorguk.ukuu.org.uk> * * Fixes: @@ -80,9 +80,9 @@ static void __exit cleanup_soundcore(void) * 2 of the License, or (at your option) any later version. * * -------------------- - * + * * Top level handler for the sound subsystem. Various devices can - * plug into this. The fact they don't all go via OSS doesn't mean + * plug into this. The fact they don't all go via OSS doesn't mean * they don't have to implement the OSS API. There is a lot of logic * to keeping much of the OSS weight out of the code in a compatibility * module, but it's up to the driver to rember to load it... @@ -111,8 +111,7 @@ static void __exit cleanup_soundcore(void) #define SOUND_STEP 16 -struct sound_unit -{ +struct sound_unit { int unit_minor; const struct file_operations *unit_fops; struct sound_unit *next; @@ -151,15 +150,14 @@ struct sound_unit #ifdef CONFIG_SOUND_OSS_CORE_PRECLAIM static int preclaim_oss = 1; #else -static int preclaim_oss = 0; +static int preclaim_oss; #endif module_param(preclaim_oss, int, 0444); static int soundcore_open(struct inode *, struct file *); -static const struct file_operations soundcore_fops = -{ +static const struct file_operations soundcore_fops = { /* We must have an owner or the module locking fails */ .owner = THIS_MODULE, .open = soundcore_open, @@ -171,70 +169,68 @@ struct sound_unit * join into it. Called with the lock asserted */ -static int __sound_insert_unit(struct sound_unit * s, struct sound_unit **list, const struct file_operations *fops, int index, int low, int top) +static int __sound_insert_unit(struct sound_unit *s, struct sound_unit **list, + const struct file_operations *fops, int index, + int low, int top) { - int n=low; + int n = low; if (index < 0) { /* first free */ - while (*list && (*list)->unit_minor<n) - list=&((*list)->next); + while (*list && (*list)->unit_minor < n) + list = &((*list)->next); - while(n<top) - { + while (n < top) { /* Found a hole ? */ - if(*list==NULL || (*list)->unit_minor>n) + if (*list == NULL || (*list)->unit_minor > n) break; - list=&((*list)->next); - n+=SOUND_STEP; + list = &((*list)->next); + n += SOUND_STEP; } - if(n>=top) + if (n >= top) return -ENOENT; } else { n = low+(index*16); while (*list) { - if ((*list)->unit_minor==n) + if ((*list)->unit_minor == n) return -EBUSY; - if ((*list)->unit_minor>n) + if ((*list)->unit_minor > n) break; - list=&((*list)->next); + list = &((*list)->next); } - } - + } + /* * Fill it in */ - - s->unit_minor=n; - s->unit_fops=fops; - + + s->unit_minor = n; + s->unit_fops = fops; + /* * Link it */ - - s->next=*list; - *list=s; - - + + s->next = *list; + *list = s; + return n; } /* * Remove a node from the chain. Called with the lock asserted */ - + static struct sound_unit *__sound_remove_unit(struct sound_unit **list, int unit) { - while(*list) - { - struct sound_unit *p=*list; - if(p->unit_minor==unit) - { - *list=p->next; + while (*list) { + struct sound_unit *p = *list; + if (p->unit_minor == unit) { + *list = p->next; return p; } - list=&(p->next); + list = &(p->next); } printk(KERN_ERR "Sound device %d went missing!\n", unit); return NULL; @@ -251,7 +247,9 @@ static struct sound_unit *__sound_remove_unit(struct sound_unit **list, int unit * list. Acquires locks as needed */ -static int sound_insert_unit(struct sound_unit **list, const struct file_operations *fops, int index, int low, int top, const char *name, umode_t mode, struct device *dev) +static int sound_insert_unit(struct sound_unit **list, const struct file_operations *fops, + int index, int low, int top, const char *name, + umode_t mode, struct device *dev) { struct sound_unit *s = kmalloc(sizeof(*s), GFP_KERNEL); int r; @@ -263,7 +261,7 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati retry: r = __sound_insert_unit(s, list, fops, index, low, top); spin_unlock(&sound_loader_lock); - + if (r < 0) goto fail; else if (r < SOUND_STEP) @@ -305,7 +303,7 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati * completed the removal before their file operations become * invalid. */ - + static void sound_remove_unit(struct sound_unit **list, int unit) { struct sound_unit *p; @@ -357,7 +355,7 @@ static void sound_remove_unit(struct sound_unit **list, int unit) * Return: The allocated number is returned on success. On failure, * a negative error code is returned. */ - + int register_sound_special_device(const struct file_operations *fops, int unit, struct device *dev) { @@ -367,69 +365,67 @@ int register_sound_special_device(const struct file_operations *fops, int unit, char _name[16]; switch (chain) { - case 0: + case 0: name = "mixer"; break; - case 1: + case 1: name = "sequencer"; if (unit >= SOUND_STEP) goto __unknown; max_unit = unit + 1; break; - case 2: + case 2: name = "midi"; break; - case 3: + case 3: name = "dsp"; break; - case 4: + case 4: name = "audio"; break; - case 5: + case 5: name = "dspW"; break; - case 8: + case 8: name = "sequencer2"; if (unit >= SOUND_STEP) goto __unknown; max_unit = unit + 1; break; - case 9: + case 9: name = "dmmidi"; break; - case 10: + case 10: name = "dmfm"; break; - case 12: + case 12: name = "adsp"; break; - case 13: + case 13: name = "amidi"; break; - case 14: + case 14: name = "admmidi"; break; - default: - { - __unknown: + default: + { + __unknown: sprintf(_name, "unknown%d", chain); - if (unit >= SOUND_STEP) - strcat(_name, "-"); - name = _name; + if (unit >= SOUND_STEP) + strcat(_name, "-"); + name = _name; } break; } return sound_insert_unit(&chains[chain], fops, -1, unit, max_unit, name, S_IRUSR | S_IWUSR, dev); } - EXPORT_SYMBOL(register_sound_special_device); int register_sound_special(const struct file_operations *fops, int unit) { return register_sound_special_device(fops, unit, NULL); } - EXPORT_SYMBOL(register_sound_special); /** @@ -449,7 +445,6 @@ int register_sound_mixer(const struct file_operations *fops, int dev) return sound_insert_unit(&chains[0], fops, dev, 0, 128, "mixer", S_IRUSR | S_IWUSR, NULL); } - EXPORT_SYMBOL(register_sound_mixer); /** @@ -469,14 +464,13 @@ int register_sound_midi(const struct file_operations *fops, int dev) return sound_insert_unit(&chains[2], fops, dev, 2, 130, "midi", S_IRUSR | S_IWUSR, NULL); } - EXPORT_SYMBOL(register_sound_midi); /* * DSP's are registered as a triple. Register only one and cheat * in open - see below. */ - + /** * register_sound_dsp - register a DSP device * @fops: File operations for the driver @@ -497,7 +491,6 @@ int register_sound_dsp(const struct file_operations *fops, int dev) return sound_insert_unit(&chains[3], fops, dev, 3, 131, "dsp", S_IWUSR | S_IRUSR, NULL); } - EXPORT_SYMBOL(register_sound_dsp); /** @@ -514,7 +507,6 @@ void unregister_sound_special(int unit) { sound_remove_unit(&chains[unit % SOUND_STEP], unit); } - EXPORT_SYMBOL(unregister_sound_special); /** @@ -529,7 +521,6 @@ void unregister_sound_mixer(int unit) { sound_remove_unit(&chains[0], unit); } - EXPORT_SYMBOL(unregister_sound_mixer); /** @@ -544,7 +535,6 @@ void unregister_sound_midi(int unit) { sound_remove_unit(&chains[2], unit); } - EXPORT_SYMBOL(unregister_sound_midi); /** @@ -561,20 +551,17 @@ void unregister_sound_dsp(int unit) { sound_remove_unit(&chains[3], unit); } - - EXPORT_SYMBOL(unregister_sound_dsp); static struct sound_unit *__look_for_unit(int chain, int unit) { struct sound_unit *s; - - s=chains[chain]; - while(s && s->unit_minor <= unit) - { - if(s->unit_minor==unit) + + s = chains[chain]; + while (s && s->unit_minor <= unit) { + if (s->unit_minor == unit) return s; - s=s->next; + s = s->next; } return NULL; } @@ -586,14 +573,13 @@ static int soundcore_open(struct inode *inode, struct file *file) struct sound_unit *s; const struct file_operations *new_fops = NULL; - chain=unit&0x0F; - if(chain==4 || chain==5) /* dsp/audio/dsp16 */ - { - unit&=0xF0; - unit|=3; - chain=3; + chain = unit&0x0F; + if (chain == 4 || chain == 5) { /* dsp/audio/dsp16 */ + unit &= 0xF0; + unit |= 3; + chain = 3; } - + spin_lock(&sound_loader_lock); s = __look_for_unit(chain, unit); if (s) @@ -636,7 +622,7 @@ static int soundcore_open(struct inode *inode, struct file *file) replace_fops(file, new_fops); if (file->f_op->open) - err = file->f_op->open(inode,file); + err = file->f_op->open(inode, file); return err; } @@ -648,7 +634,8 @@ static int soundcore_open(struct inode *inode, struct file *file) static void cleanup_oss_soundcore(void) { /* We have nothing to really do here - we know the lists must be - empty */ + * empty + */ unregister_chrdev(SOUND_MAJOR, "sound"); } -- 1.9.1

1 0

[alsa-devel] non-blocking mode example
by ThomasJF 18 Aug '17

18 Aug '17

Hi, could anyone please link me to or show me a small demo example, like "pcm_min.c" with working snd_pcm_open in SND_PCM_NONBLOCK - mode? Standard I/O transfers or Event waiting routines. I get both working - but only in blocking mode. I want to stop/pause with commands inside a tcp-server. Regards Thomas

2 14

[alsa-devel] [PATCH 00/14] constify asoc snd_soc_dai_ops structures
by Arvind Yadav 18 Aug '17

18 Aug '17

snd_soc_dai_ops are not supposed to change at runtime. All functions working with snd_soc_dai_ops provided by <sound/soc-dai.h> work with const snd_soc_dai_ops. So mark the non-const structs as const. Arvind Yadav (14): [PATCH 01/14] ASoC: blackfin: constify snd_soc_dai_ops structures [PATCH 02/14] ASoC: codecs: constify snd_soc_dai_ops structures [PATCH 03/14] ASoC: dwc: constify snd_soc_dai_ops structures [PATCH 04/14] ASoC: fsl: constify snd_soc_dai_ops structures [PATCH 05/14] ASoC: hisilicon: constify snd_soc_dai_ops structures [PATCH 06/14] ASoC: intel: constify snd_soc_dai_ops structures [PATCH 07/14] ASoC: mediatek: constify snd_soc_dai_ops structures [PATCH 08/14] ASoC: pxa: constify snd_soc_dai_ops structures [PATCH 09/14] ASoC: rockchip: constify snd_soc_dai_ops structures [PATCH 10/14] ASoC: spear: constify snd_soc_dai_ops structures [PATCH 11/14] ASoC: sunxi: constify snd_soc_dai_ops structures [PATCH 12/14] ASoC: tegra: constify snd_soc_dai_ops structures [PATCH 13/14] ASoC: ux500: constify snd_soc_dai_ops structures [PATCH 14/14] ASoC: zte: constify snd_soc_dai_ops structures sound/soc/blackfin/bf6xx-i2s.c | 2 +- sound/soc/codecs/cs42l42.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/hdac_hdmi.c | 2 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/max98926.c | 2 +- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- sound/soc/codecs/msm8916-wcd-digital.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/tas5720.c | 2 +- sound/soc/codecs/zx_aud96p22.c | 2 +- sound/soc/dwc/dwc-i2s.c | 2 +- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_spdif.c | 2 +- sound/soc/hisilicon/hi6210-i2s.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +++--- sound/soc/intel/skylake/skl-pcm.c | 8 ++++---- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/spear/spdif_in.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 2 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/zte/zx-i2s.c | 2 +- sound/soc/zte/zx-spdif.c | 2 +- sound/soc/zte/zx-tdm.c | 2 +- 30 files changed, 35 insertions(+), 35 deletions(-) -- 1.9.1

2 21

[alsa-devel] Applied "ASoC: codecs: constify snd_soc_dai_ops structures" to the asoc tree
by Mark Brown 18 Aug '17

18 Aug '17

The patch ASoC: codecs: constify snd_soc_dai_ops structures has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From eb59d73cb535ba32df928f210fb9a8529bf465c0 Mon Sep 17 00:00:00 2001 From: Arvind Yadav <arvind.yadav.cs(a)gmail.com> Date: Fri, 18 Aug 2017 17:35:59 +0530 Subject: [PATCH] ASoC: codecs: constify snd_soc_dai_ops structures snd_soc_dai_ops are not supposed to change at runtime. All functions working with snd_soc_dai_ops provided by <sound/soc-dai.h> work with const snd_soc_dai_ops. So mark the non-const structs as const. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/max98926.c | 2 +- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/tas5720.c | 2 +- sound/soc/codecs/zx_aud96p22.c | 2 +- 10 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 55e4520cdcaf..9e860dfa0163 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -911,7 +911,7 @@ static int cs42l42_digital_mute(struct snd_soc_dai *dai, int mute) SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops cs42l42_ops = { +static const struct snd_soc_dai_ops cs42l42_ops = { .hw_params = cs42l42_pcm_hw_params, .set_fmt = cs42l42_set_dai_fmt, .set_sysclk = cs42l42_set_sysclk, diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 4f35af6a5d3a..da2d353af5ba 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -502,7 +502,7 @@ static int es8316_mute(struct snd_soc_dai *dai, int mute) #define ES8316_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops es8316_ops = { +static const struct snd_soc_dai_ops es8316_ops = { .startup = es8316_pcm_startup, .hw_params = es8316_pcm_hw_params, .set_fmt = es8316_set_dai_fmt, diff --git a/sound/soc/codecs/inno_rk3036.c b/sound/soc/codecs/inno_rk3036.c index 64b0be94bea3..6b59b6f08298 100644 --- a/sound/soc/codecs/inno_rk3036.c +++ b/sound/soc/codecs/inno_rk3036.c @@ -310,7 +310,7 @@ static int rk3036_codec_dai_hw_params(struct snd_pcm_substream *substream, SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops rk3036_codec_dai_ops = { +static const struct snd_soc_dai_ops rk3036_codec_dai_ops = { .set_fmt = rk3036_codec_dai_set_fmt, .hw_params = rk3036_codec_dai_hw_params, }; diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 6c0c0d6e8f3c..2f60924fe919 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -350,7 +350,7 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, return 0; } -static struct snd_soc_dai_ops max9867_dai_ops = { +static const struct snd_soc_dai_ops max9867_dai_ops = { .set_fmt = max9867_dai_set_fmt, .set_sysclk = max9867_set_dai_sysclk, .prepare = max9867_prepare, diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c index 7a39bfb9e0f9..59d03d036945 100644 --- a/sound/soc/codecs/max98926.c +++ b/sound/soc/codecs/max98926.c @@ -459,7 +459,7 @@ static int max98926_dai_hw_params(struct snd_pcm_substream *substream, #define MAX98926_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops max98926_dai_ops = { +static const struct snd_soc_dai_ops max98926_dai_ops = { .set_fmt = max98926_dai_set_fmt, .hw_params = max98926_dai_hw_params, }; diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 4cc52cecf3b7..f07674eddc52 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -753,7 +753,7 @@ static void pm8916_wcd_analog_shutdown(struct snd_pcm_substream *substream, RST_CTL_DIG_SW_RST_N_MASK, 0); } -static struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = { +static const struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = { .startup = pm8916_wcd_analog_startup, .shutdown = pm8916_wcd_analog_shutdown, }; diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index 3c5f555ae6d0..c94e94fe8297 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1265,7 +1265,7 @@ static int rt5616_resume(struct snd_soc_codec *codec) #define RT5616_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5616_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5616_aif_dai_ops = { .hw_params = rt5616_hw_params, .set_fmt = rt5616_set_dai_fmt, .set_sysclk = rt5616_set_dai_sysclk, diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index ebc5e3b2b62b..1bbe36338cd3 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -2860,7 +2860,7 @@ static int rt5663_resume(struct snd_soc_codec *codec) #define RT5663_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5663_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5663_aif_dai_ops = { .hw_params = rt5663_hw_params, .set_fmt = rt5663_set_dai_fmt, .set_sysclk = rt5663_set_dai_sysclk, diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index d14b78b1917b..a736a2a6976c 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -507,7 +507,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_tas5720 = { #define TAS5720_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE |\ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops tas5720_speaker_dai_ops = { +static const struct snd_soc_dai_ops tas5720_speaker_dai_ops = { .hw_params = tas5720_hw_params, .set_fmt = tas5720_set_dai_fmt, .set_tdm_slot = tas5720_set_dai_tdm_slot, diff --git a/sound/soc/codecs/zx_aud96p22.c b/sound/soc/codecs/zx_aud96p22.c index 36d0a7087a54..51fad48892d3 100644 --- a/sound/soc/codecs/zx_aud96p22.c +++ b/sound/soc/codecs/zx_aud96p22.c @@ -312,7 +312,7 @@ static int aud96p22_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static struct snd_soc_dai_ops aud96p22_dai_ops = { +static const struct snd_soc_dai_ops aud96p22_dai_ops = { .set_fmt = aud96p22_set_fmt, }; -- 2.13.2

1 0

[alsa-devel] [PATCH 1/3] sound: ASoC: tegra: Remove SoC-specific Kconfig depends and selects
by Paul Kocialkowski 18 Aug '17

18 Aug '17

This removes the SoC-specific dependencies on the platform drivers, as well as SoC-specific selections of platform drivers for the machine drivers. The rationale behind this change is that the dependencies are not actual build dependencies but run-time ones. The previously listed SoCs were also incomplete: for instance, tegra124 uses the tegra30 platform drivers, which could not be built without ARCH_TEGRA_3x_SOC set. Descriptions of the tristates are also added to allow these options to be set via defconfig/menuconfig. Signed-off-by: Paul Kocialkowski <contact(a)paulk.fr> --- sound/soc/tegra/Kconfig | 42 ++++++++++++++---------------------------- 1 file changed, 14 insertions(+), 28 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index efbe8d4c019e..6875fc39a575 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -9,8 +9,8 @@ config SND_SOC_TEGRA Say Y or M here if you want support for SoC audio on Tegra. config SND_SOC_TEGRA20_AC97 - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 AC97 interface" + depends on SND_SOC_TEGRA select SND_SOC_AC97_BUS select SND_SOC_TEGRA20_DAS help @@ -19,16 +19,16 @@ config SND_SOC_TEGRA20_AC97 machine drivers to support below. config SND_SOC_TEGRA20_DAS - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 DAS module" + depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 DAS module. You will also need to select the individual machine drivers to support below. config SND_SOC_TEGRA20_I2S - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 I2S interface" + depends on SND_SOC_TEGRA select SND_SOC_TEGRA20_DAS help Say Y or M if you want to add support for codecs attached to the @@ -36,8 +36,8 @@ config SND_SOC_TEGRA20_I2S machine drivers to support below. config SND_SOC_TEGRA20_SPDIF - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 SPDIF interface" + depends on SND_SOC_TEGRA default m help Say Y or M if you want to add support for the Tegra20 SPDIF interface. @@ -45,16 +45,16 @@ config SND_SOC_TEGRA20_SPDIF below. config SND_SOC_TEGRA30_AHUB - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + tristate "Tegra30 AHUB module" + depends on SND_SOC_TEGRA help - Say Y or M if you want to add support for the Tegra20 AHUB module. + Say Y or M if you want to add support for the Tegra30 AHUB module. You will also need to select the individual machine drivers to support below. config SND_SOC_TEGRA30_I2S - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + tristate "Tegra30 I2S interface" + depends on SND_SOC_TEGRA select SND_SOC_TEGRA30_AHUB help Say Y or M if you want to add support for codecs attached to the @@ -64,8 +64,6 @@ config SND_SOC_TEGRA30_I2S config SND_SOC_TEGRA_RT5640 tristate "SoC Audio support for Tegra boards using an RT5640 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_RT5640 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -74,8 +72,6 @@ config SND_SOC_TEGRA_RT5640 config SND_SOC_TEGRA_WM8753 tristate "SoC Audio support for Tegra boards using a WM8753 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8753 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -84,8 +80,6 @@ config SND_SOC_TEGRA_WM8753 config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -94,7 +88,7 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_WM9712 tristate "SoC Audio support for Tegra boards using a WM9712 codec" - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC && GPIOLIB + depends on SND_SOC_TEGRA && GPIOLIB select SND_SOC_TEGRA20_AC97 select SND_SOC_WM9712 help @@ -104,7 +98,6 @@ config SND_SOC_TEGRA_WM9712 config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" depends on SND_SOC_TEGRA && I2C - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TLV320AIC23_I2C help Say Y or M here if you want to add support for SoC audio on the @@ -113,7 +106,6 @@ config SND_SOC_TEGRA_TRIMSLICE config SND_SOC_TEGRA_ALC5632 tristate "SoC Audio support for Tegra boards using an ALC5632 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_ALC5632 help Say Y or M here if you want to add support for SoC audio on the @@ -122,8 +114,6 @@ config SND_SOC_TEGRA_ALC5632 config SND_SOC_TEGRA_MAX98090 tristate "SoC Audio support for Tegra boards using a MAX98090 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_MAX98090 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -132,8 +122,6 @@ config SND_SOC_TEGRA_MAX98090 config SND_SOC_TEGRA_RT5677 tristate "SoC Audio support for Tegra boards using a RT5677 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_RT5677 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -142,8 +130,6 @@ config SND_SOC_TEGRA_RT5677 config SND_SOC_TEGRA_SGTL5000 tristate "SoC Audio support for Tegra boards using a SGTL5000 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_SGTL5000 help Say Y or M here if you want to add support for SoC audio on Tegra -- 2.13.3

3 9

[alsa-devel] [PATCH v1 1/2] ASoC: mediatek: Add interrupt-names property in binding text
by Ryder Lee 18 Aug '17

18 Aug '17

Add interrupt-names property in DT - if future revisions of the IP have different sets of interrupts the binding can adapt gracefully. Signed-off-by: Ryder Lee <ryder.lee(a)mediatek.com> CC: Garlic Tseng <garlic.tseng(a)mediatek.com> --- Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt index 9800a56..77a57f8 100644 --- a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt +++ b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt @@ -3,7 +3,8 @@ Mediatek AFE PCM controller for mt2701 Required properties: - compatible = "mediatek,mt2701-audio"; - reg: register location and size -- interrupts: Should contain AFE interrupt +- interrupts: should contain AFE and ASYS interrupts +- interrupt-names: should be "afe" and "asys" - power-domains: should define the power domain - clock-names: should have these clock names: "infra_sys_audio_clk", @@ -59,6 +60,7 @@ Example: <0 0x112A0000 0 0x20000>; interrupts = <GIC_SPI 104 IRQ_TYPE_LEVEL_LOW>, <GIC_SPI 132 IRQ_TYPE_LEVEL_LOW>; + interrupt-names = "afe", "asys"; power-domains = <&scpsys MT2701_POWER_DOMAIN_IFR_MSC>; clocks = <&infracfg CLK_INFRA_AUDIO>, <&topckgen CLK_TOP_AUD_MUX1_SEL>, -- 1.9.1

2 3

[alsa-devel] [PATCHv3 0/6] Motorola Droid 4 Audio Support
by Sebastian Reichel 18 Aug '17

18 Aug '17

Hi, This adds audio support to Motorola Droid 4. Tested: - Playing via both DACs using Speaker, Earpiece, Headphone - Recording using internal Mic - Volume Controls Known not to work: - 3.5mm detection support (requires closed source firmware, needs further investigation) - Modem / Bluetooth Audio (actually untested, but probably needs some quirks) Changes since PATCHv2: * Fix a whitespace issue * Fix const notes Takashi provided * Fix a DAPM route issue I accidently introduced in v2 Changes since PATCHv1: * Add patch from Tony exporting soc_dpcm_runtime_update * Integrate DT patch for vaudio initial mode * Split dt-binding from codec patch and add Ack from Rob * Fix CPCAP position in Kconfig/Makefile * Avoid "err +=" constructs * Simplify reset function * Drop cpcap_audio_read/write helpers * Do not use tertiary operator for mute register value * Update Input Mux logic * Switch from simple-audio-card to audio-graph-card -- Sebastian Sebastian Reichel (5): dt-bindings: sound: add motorola,cpcap-audio-codec ASoC: codec: cpcap: new codec ARM: dts: motorola-cpcap-mapphone: add audio-codec ARM: dts: motorola-cpcap-mapphone: set initial mode for vaudio ARM: dts: omap4-droid4: add soundcard Tony Lindgren (1): ALSA: pcm: Export soc_dpcm_runtime_update .../bindings/sound/motorola,cpcap-audio-codec.txt | 19 + arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 7 + arch/arm/boot/dts/omap4-droid4-xt894.dts | 84 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cpcap.c | 1594 ++++++++++++++++++++ sound/soc/soc-pcm.c | 2 + 7 files changed, 1713 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt create mode 100644 sound/soc/codecs/cpcap.c -- 2.13.2

4 14

[alsa-devel] [PATCH] ALSA: emu10k1: Fix forgotten user-copy conversion in init code
by Takashi Iwai 18 Aug '17

18 Aug '17

The commit d42fe63d5839 ("ALSA: emu10k1: Get rid of set_fs() usage") converted the user-space copy hack with set_fs() to the direct memcpy(), but one place was forgotten. This resulted in the error from snd_emu10k1_init_efx(), eventually failed to load the driver. Fix the missing piece. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196687 Fixes: d42fe63d5839 ("ALSA: emu10k1: Get rid of set_fs() usage") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/emu10k1/emufx.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index dc585959ca32..a2b56b188be4 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -698,10 +698,18 @@ static int copy_gctl(struct snd_emu10k1 *emu, { struct snd_emu10k1_fx8010_control_old_gpr __user *octl; - if (emu->support_tlv) - return copy_from_user(gctl, &_gctl[idx], sizeof(*gctl)); + if (emu->support_tlv) { + if (in_kernel) + memcpy(gctl, (void *)&_gctl[idx], sizeof(*gctl)); + else if (copy_from_user(gctl, &_gctl[idx], sizeof(*gctl))) + return -EFAULT; + return 0; + } + octl = (struct snd_emu10k1_fx8010_control_old_gpr __user *)_gctl; - if (copy_from_user(gctl, &octl[idx], sizeof(*octl))) + if (in_kernel) + memcpy(gctl, (void *)&octl[idx], sizeof(*octl)); + else if (copy_from_user(gctl, &octl[idx], sizeof(*octl))) return -EFAULT; gctl->tlv = NULL; return 0; -- 2.14.1

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: add DSD support for new Amanero PID
by Jussi Laako 18 Aug '17

18 Aug '17

Add DSD support for new Amanero Combo384 firmware version with a new PID. This firmware uses DSD_U32_BE. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2eb6e0d..c09a28b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1374,6 +1374,10 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, } } break; + case USB_ID(0x16d0, 0x0a23): + if (fp->altsetting == 2) + return SNDRV_PCM_FMTBIT_DSD_U32_BE; + break; default: break; -- 2.7.4

2 1

[alsa-devel] [PATCH 0/6] ASoC: make snd_pcm_hardware const
by Bhumika Goyal 18 Aug '17

18 Aug '17

Make these const. Done using Coccinelle Bhumika Goyal (6): ASoC: fsl: make snd_pcm_hardware const ASoC: Intel: Atom: make snd_pcm_hardware const ASoC: Intel: Skylake: make snd_pcm_hardware const ASoC: kirkwood: make snd_pcm_hardware const ASoC: sh: make snd_pcm_hardware const ASoC: qcom: make snd_pcm_hardware const sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/imx-pcm-fiq.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/sh/dma-sh7760.c | 2 +- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/siu_dai.c | 2 +- 10 files changed, 10 insertions(+), 10 deletions(-) -- 1.9.1

3 13

[alsa-devel] [PATCH] ALSA: usb-audio: don't retry snd_usb_ctl_msg after timeout
by Stephen Barber 18 Aug '17

18 Aug '17

A few calls to snd_usb_ctl_msg wrap the function in a retry loop. In the worst case, the timeout for snd_usb_ctl_msg is 5 seconds, which when retried 10 times (for example, if a device is removed) could cause a probe to hang for ~50 seconds. Example stack trace from 3.14 which triggered a hung task timeout: Call Trace: [<ffffffffa2c1f720>] ? inet6_set_link_af.part.35+0x12/0x12 [<ffffffffa2c20309>] schedule+0x6e/0x70 [<ffffffffa2c1f81c>] schedule_timeout+0xfc/0x13c [<ffffffffa2667bbc>] ? rcu_read_unlock_sched_notrace+0x17/0x17 [<ffffffffa2c20d68>] __wait_for_common+0x153/0x190 [<ffffffffa2c20d68>] ? __wait_for_common+0x153/0x190 [<ffffffffa26890e5>] ? wake_up_state+0x12/0x12 [<ffffffffa2c20e0e>] wait_for_completion_timeout+0x1d/0x1f [<ffffffffa2a07c70>] usb_start_wait_urb+0x93/0xf1 [<ffffffffa2a07daf>] usb_control_msg+0xe1/0x11d [<ffffffffc02cd254>] snd_usb_ctl_msg+0x9c/0xf1 [snd_usb_audio] [<ffffffffc02ce191>] snd_usb_mixer_set_ctl_value+0x124/0xab1 [snd_usb_audio] [<ffffffffc02ce230>] snd_usb_mixer_set_ctl_value+0x1c3/0xab1 [snd_usb_audio] [<ffffffffc02ce58e>] snd_usb_mixer_set_ctl_value+0x521/0xab1 [snd_usb_audio] [<ffffffffc02cee88>] snd_usb_mixer_add_control+0x36a/0x1264 [snd_usb_audio] [<ffffffffc02cf323>] snd_usb_mixer_add_control+0x805/0x1264 [snd_usb_audio] [<ffffffffa2a06e11>] ? usb_free_urb+0x1a/0x1c [<ffffffffc02cfcf7>] snd_usb_mixer_add_control+0x11d9/0x1264 [snd_usb_audio] [<ffffffffc02d000f>] snd_usb_create_mixer+0xbc/0x286 [snd_usb_audio] [<ffffffffc02cac18>] 0xffffffffc02cac17 [<ffffffffa2a0aaf1>] usb_probe_interface+0x17c/0x21c [<ffffffffa29a65bc>] driver_probe_device+0xae/0x1fa [<ffffffffa29a6767>] __device_attach_driver+0x5f/0x66 [<ffffffffa29a6708>] ? driver_probe_device+0x1fa/0x1fa [<ffffffffa29a4a60>] bus_for_each_drv+0x87/0xaa [<ffffffffa29a688a>] __device_attach+0x9d/0x101 [<ffffffffa29a6913>] device_initial_probe+0x13/0x15 [<ffffffffa29a5ae6>] bus_probe_device+0x33/0x96 [<ffffffffa29a3d19>] device_add+0x328/0x547 [<ffffffffa2a09355>] usb_set_configuration+0x624/0x674 [<ffffffffa2a11949>] generic_probe+0x45/0x77 [<ffffffffa2a0a962>] usb_probe_device+0x2d/0x40 [<ffffffffa29a65bc>] driver_probe_device+0xae/0x1fa [<ffffffffa29a6767>] __device_attach_driver+0x5f/0x66 [<ffffffffa29a6708>] ? driver_probe_device+0x1fa/0x1fa [<ffffffffa29a4a60>] bus_for_each_drv+0x87/0xaa [<ffffffffa29a688a>] __device_attach+0x9d/0x101 [<ffffffffa29a6913>] device_initial_probe+0x13/0x15 [<ffffffffa29a5ae6>] bus_probe_device+0x33/0x96 [<ffffffffa29a3d19>] device_add+0x328/0x547 [<ffffffffa29030bc>] ? add_device_randomness+0x111/0x130 [<ffffffffa2a00967>] usb_new_device+0x2a2/0x3c0 [<ffffffffa2a02ddc>] hub_thread+0xa3d/0xeed [<ffffffffa2c2010d>] ? __schedule+0x41e/0x5ac [<ffffffffa26957ce>] ? finish_wait+0x62/0x62 [<ffffffffa2a0239f>] ? usb_reset_device+0x16a/0x16a [<ffffffffa267b255>] kthread+0x108/0x110 [<ffffffffa267b14d>] ? __kthread_parkme+0x67/0x67 [<ffffffffa2c23b2c>] ret_from_fork+0x7c/0xb0 [<ffffffffa267b14d>] ? __kthread_parkme+0x67/0x67 Signed-off-by: Stephen Barber <smbarber(a)chromium.org> --- sound/usb/mixer.c | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 082736c539bc..253fc946db29 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -318,12 +318,15 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, while (timeout-- > 0) { idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); - if (snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request, - USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, - validx, idx, buf, val_len) >= val_len) { + err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + validx, idx, buf, val_len); + if (err >= val_len) { *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len)); err = 0; goto out; + } else if (err == -ETIMEDOUT) { + goto out; } } usb_audio_dbg(chip, @@ -483,12 +486,15 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, while (timeout-- > 0) { idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); - if (snd_usb_ctl_msg(chip->dev, - usb_sndctrlpipe(chip->dev, 0), request, - USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, - validx, idx, buf, val_len) >= 0) { + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), request, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + validx, idx, buf, val_len); + if (err >= 0) { err = 0; goto out; + } else if (err == -ETIMEDOUT) { + goto out; } } usb_audio_dbg(chip, "cannot set ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d, data = %#x/%#x\n", -- 2.14.1.480.gb18f417b89-goog

2 1

[alsa-devel] Digigram VX222e not seen as card
by Sam Mulvey 18 Aug '17

18 Aug '17

A radio station I'm working on has a machine with a Digigram VX222e connected to it. According to the Digigram site and the ALSA website, the sound card is supported by ALSA. Unfortunately, it doesn't appear to be detected as a sound card. Reading online, it requires some firmware, but it appears that my system has the proper firmware installed. The VX222e is a PCIe card, but it appears to be PCI on the card with a PCIe to PCI bridge. According to Digigram's support site, they made a hardware change in March of 2017, and I suspect that this might be the problem. I'm guessing it might be as simple enough as an ID match that I need to add to a module, but I've never worked with the card before, and I'm a little stuck. There's nothing regarding the card in dmesg, but it is visible in lspci: 03:00.0 PCI bridge: PLX Technology, Inc. PEX 8111 PCI Express-to-PCI Bridge (rev 21) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 3 NUMA node: 0 Bus: primary=03, secondary=04, subordinate=04, sec-latency=32 I/O behind bridge: 0000e000-0000efff [size=4K] Memory behind bridge: fde00000-fdefffff [size=1M] Prefetchable memory behind bridge: fdd00000-fddfffff [size=1M] Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA- VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [50] MSI: Enable- Count=1/1 Maskable- 64bit+ Address: 0000000000000000 Data: 0000 Capabilities: [60] Express (v1) PCI-Express to PCI/PCI-X Bridge, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- AttnBtn- AttnInd- PwrInd- RBE- SlotPowerLimit 25.000W DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- BrConfRtry- MaxPayload 128 bytes, MaxReadReq 512 bytes DevSta: CorrErr- UncorrErr+ FatalErr- UnsuppReq+ AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x1, ASPM L0s L1, Exit Latency L0s <1us, L1 <16us ClockPM- Surprise- LLActRep- BwNot- ASPMOptComp- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100 v1] Power Budgeting <?> Kernel modules: shpchp 04:04.0 Multimedia audio controller: PLX Technology, Inc. PCI9056 32-bit 66MHz PCI <-> IOBus Bridge Subsystem: Digigram Device ba41 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 32, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 3 NUMA node: 0 Region 0: Memory at fdeff000 (32-bit, non-prefetchable) [size=512] Region 1: I/O ports at ec00 [size=256] Region 2: I/O ports at ef00 [size=128] Capabilities: [40] Power Management version 0 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [48] #00 [0000] Capabilities: [4c] Vital Product Data pcilib: sysfs_read_vpd: read failed: Input/output error I'm also documenting what I've got and what I've tried here: https://vis.nu/blog/digigram_vx222e_isn_t_seen_in_arch_linux Any pointers would be helpful. Thanks! -Sam

1 0

[alsa-devel] [PATCH] ASoC: mediatek: Correct the interrupt property
by Ryder Lee 18 Aug '17

18 Aug '17

This patch update interrupt property since we only use ASYS IRQ in the driver, and this may confuse people. Signed-off-by: Ryder Lee <ryder.lee(a)mediatek.com> CC: Garlic Tseng <garlic.tseng(a)mediatek.com> --- Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt index 9800a56..9d7b6dc 100644 --- a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt +++ b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt @@ -3,7 +3,7 @@ Mediatek AFE PCM controller for mt2701 Required properties: - compatible = "mediatek,mt2701-audio"; - reg: register location and size -- interrupts: Should contain AFE interrupt +- interrupts: Should contain ASYS interrupt - power-domains: should define the power domain - clock-names: should have these clock names: "infra_sys_audio_clk", @@ -57,8 +57,7 @@ Example: compatible = "mediatek,mt2701-audio"; reg = <0 0x11220000 0 0x2000>, <0 0x112A0000 0 0x20000>; - interrupts = <GIC_SPI 104 IRQ_TYPE_LEVEL_LOW>, - <GIC_SPI 132 IRQ_TYPE_LEVEL_LOW>; + interrupts = <GIC_SPI 132 IRQ_TYPE_LEVEL_LOW>; power-domains = <&scpsys MT2701_POWER_DOMAIN_IFR_MSC>; clocks = <&infracfg CLK_INFRA_AUDIO>, <&topckgen CLK_TOP_AUD_MUX1_SEL>, -- 1.9.1

3 4

[alsa-devel] [PATCH] ASoC: tegra: Remove SoC-specific Kconfig depends and selects
by Paul Kocialkowski 17 Aug '17

17 Aug '17

This removes the SoC-specific dependencies on the platform drivers, as well as SoC-specific selections of platform drivers for the machine drivers. The rationale behind this change is that the dependencies are not actual build dependencies but run-time ones. The previously listed SoCs were also incomplete: for instance, tegra124 uses the tegra30 platform drivers, which could not be built without ARCH_TEGRA_3x_SOC set. Descriptions of the tristates are also added to allow these options to be set via defconfig/menuconfig. Signed-off-by: Paul Kocialkowski <contact(a)paulk.fr> Acked-by: Thierry Reding <treding(a)nvidia.com> --- sound/soc/tegra/Kconfig | 42 ++++++++++++++---------------------------- 1 file changed, 14 insertions(+), 28 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index efbe8d4c019e..6875fc39a575 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -9,8 +9,8 @@ config SND_SOC_TEGRA Say Y or M here if you want support for SoC audio on Tegra. config SND_SOC_TEGRA20_AC97 - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 AC97 interface" + depends on SND_SOC_TEGRA select SND_SOC_AC97_BUS select SND_SOC_TEGRA20_DAS help @@ -19,16 +19,16 @@ config SND_SOC_TEGRA20_AC97 machine drivers to support below. config SND_SOC_TEGRA20_DAS - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 DAS module" + depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 DAS module. You will also need to select the individual machine drivers to support below. config SND_SOC_TEGRA20_I2S - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 I2S interface" + depends on SND_SOC_TEGRA select SND_SOC_TEGRA20_DAS help Say Y or M if you want to add support for codecs attached to the @@ -36,8 +36,8 @@ config SND_SOC_TEGRA20_I2S machine drivers to support below. config SND_SOC_TEGRA20_SPDIF - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 SPDIF interface" + depends on SND_SOC_TEGRA default m help Say Y or M if you want to add support for the Tegra20 SPDIF interface. @@ -45,16 +45,16 @@ config SND_SOC_TEGRA20_SPDIF below. config SND_SOC_TEGRA30_AHUB - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + tristate "Tegra30 AHUB module" + depends on SND_SOC_TEGRA help - Say Y or M if you want to add support for the Tegra20 AHUB module. + Say Y or M if you want to add support for the Tegra30 AHUB module. You will also need to select the individual machine drivers to support below. config SND_SOC_TEGRA30_I2S - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + tristate "Tegra30 I2S interface" + depends on SND_SOC_TEGRA select SND_SOC_TEGRA30_AHUB help Say Y or M if you want to add support for codecs attached to the @@ -64,8 +64,6 @@ config SND_SOC_TEGRA30_I2S config SND_SOC_TEGRA_RT5640 tristate "SoC Audio support for Tegra boards using an RT5640 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_RT5640 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -74,8 +72,6 @@ config SND_SOC_TEGRA_RT5640 config SND_SOC_TEGRA_WM8753 tristate "SoC Audio support for Tegra boards using a WM8753 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8753 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -84,8 +80,6 @@ config SND_SOC_TEGRA_WM8753 config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -94,7 +88,7 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_WM9712 tristate "SoC Audio support for Tegra boards using a WM9712 codec" - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC && GPIOLIB + depends on SND_SOC_TEGRA && GPIOLIB select SND_SOC_TEGRA20_AC97 select SND_SOC_WM9712 help @@ -104,7 +98,6 @@ config SND_SOC_TEGRA_WM9712 config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" depends on SND_SOC_TEGRA && I2C - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TLV320AIC23_I2C help Say Y or M here if you want to add support for SoC audio on the @@ -113,7 +106,6 @@ config SND_SOC_TEGRA_TRIMSLICE config SND_SOC_TEGRA_ALC5632 tristate "SoC Audio support for Tegra boards using an ALC5632 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_ALC5632 help Say Y or M here if you want to add support for SoC audio on the @@ -122,8 +114,6 @@ config SND_SOC_TEGRA_ALC5632 config SND_SOC_TEGRA_MAX98090 tristate "SoC Audio support for Tegra boards using a MAX98090 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_MAX98090 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -132,8 +122,6 @@ config SND_SOC_TEGRA_MAX98090 config SND_SOC_TEGRA_RT5677 tristate "SoC Audio support for Tegra boards using a RT5677 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_RT5677 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -142,8 +130,6 @@ config SND_SOC_TEGRA_RT5677 config SND_SOC_TEGRA_SGTL5000 tristate "SoC Audio support for Tegra boards using a SGTL5000 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_SGTL5000 help Say Y or M here if you want to add support for SoC audio on Tegra -- 2.13.4

1 0

[alsa-devel] [PATCH v2] ASoC: Intel: kbl_rt5663_rt5514_max98927: Add rt5514 spi dailink
by Harsha Priya 17 Aug '17

17 Aug '17

This patch adds a dai to rt5514-spi driver for wake on voice functionality. This v2 patch adds the dependency on SPI in Kconfig Signed-off-by: Hsin-yu Chao <hychao(a)google.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- sound/soc/intel/Kconfig | 3 ++- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 9 +++++++++ 2 files changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b301bff..b3c7f55 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -255,11 +255,12 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH tristate "ASoC Audio driver for KBL with RT5663, RT5514 and MAX98927 in I2S Mode" - depends on X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C && SPI select SND_SOC_INTEL_SST select SND_SOC_INTEL_SKYLAKE select SND_SOC_RT5663 select SND_SOC_RT5514 + select SND_SOC_RT5514_SPI select SND_SOC_MAX98927 select SND_SOC_HDAC_HDMI help diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 0fab2df..c2ef538 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -65,6 +65,7 @@ enum { KBL_DPCM_AUDIO_HS_PB, KBL_DPCM_AUDIO_ECHO_REF_CP, KBL_DPCM_AUDIO_DMIC_CP, + KBL_DPCM_AUDIO_RT5514_DSP, KBL_DPCM_AUDIO_HDMI1_PB, KBL_DPCM_AUDIO_HDMI2_PB, }; @@ -473,6 +474,14 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) .capture_only = 1, .nonatomic = 1, }, + [KBL_DPCM_AUDIO_RT5514_DSP] = { + .name = "rt5514 dsp", + .stream_name = "Wake on Voice", + .cpu_dai_name = "spi-PRP0001:00", + .platform_name = "spi-PRP0001:00", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", .stream_name = "dmiccap", -- 1.9.1

3 2

[alsa-devel] [PATCH 1/2] ASoC: codec: use enable pin to control dmic start and stop
by Lin Huang 17 Aug '17

17 Aug '17

on some board use enable pin to control dmic start and stop, so add this feature in dmic driver. Signed-off-by: Lin Huang <hl(a)rock-chips.com> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/dmic.c | 61 ++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 62 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ccc5814..dff3586 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -61,7 +61,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7219 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C - select SND_SOC_DMIC + select SND_SOC_DMIC if GPIOLIB select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 32b7a48..c4e5f47 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -19,6 +19,8 @@ * */ +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/module.h> @@ -27,6 +29,39 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> +struct dmic_priv { + struct gpio_desc *dmic_en; +}; + +static int dmic_daiops_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct dmic_priv *dmic_data = snd_soc_codec_get_drvdata(codec); + + if (!dmic_data->dmic_en) + return 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + gpiod_set_value(dmic_data->dmic_en, 1); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + gpiod_set_value(dmic_data->dmic_en, 0); + break; + } + + return 0; +} + +static const struct snd_soc_dai_ops dmic_dai_ops = { + .trigger = dmic_daiops_trigger, +}; + static struct snd_soc_dai_driver dmic_dai = { .name = "dmic-hifi", .capture = { @@ -38,8 +73,24 @@ static struct snd_soc_dai_driver dmic_dai = { | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, + .ops = &dmic_dai_ops, }; +static int dmic_codec_probe(struct snd_soc_codec *codec) +{ + struct dmic_priv *dmic_data = snd_soc_codec_get_drvdata(codec); + + dmic_data->dmic_en = devm_gpiod_get_optional(codec->dev, + "dmicen", GPIOD_OUT_LOW); + + if (IS_ERR(dmic_data->dmic_en)) + return PTR_ERR(dmic_data->dmic_en); + + snd_soc_codec_set_drvdata(codec, dmic_data); + + return 0; +} + static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0, SND_SOC_NOPM, 0, 0), @@ -51,6 +102,7 @@ static const struct snd_soc_dapm_route intercon[] = { }; static struct snd_soc_codec_driver soc_dmic = { + .probe = dmic_codec_probe, .dapm_widgets = dmic_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets), .dapm_routes = intercon, @@ -59,6 +111,15 @@ static struct snd_soc_codec_driver soc_dmic = { static int dmic_dev_probe(struct platform_device *pdev) { + struct dmic_priv *dmic_data; + + dmic_data = devm_kzalloc(&pdev->dev, sizeof(*dmic_data), GFP_KERNEL); + + if (!dmic_data) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, dmic_data); + return snd_soc_register_codec(&pdev->dev, &soc_dmic, &dmic_dai, 1); } -- 2.7.4

3 4

[alsa-devel] Applied "ASoC: codec: use enable pin to control dmic start and stop" to the asoc tree
by Mark Brown 17 Aug '17

17 Aug '17

The patch ASoC: codec: use enable pin to control dmic start and stop has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 23c7159a799f3364720b3d250f57bc8e26b17287 Mon Sep 17 00:00:00 2001 From: huang lin <hl(a)rock-chips.com> Date: Thu, 17 Aug 2017 10:24:44 +0800 Subject: [PATCH] ASoC: codec: use enable pin to control dmic start and stop on some board use enable pin to control dmic start and stop, so add this feature in dmic driver. Signed-off-by: Lin Huang <hl(a)rock-chips.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/dmic.c | 48 +++++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 48 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6c78b0b49b81..952c44dd5e90 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -71,7 +71,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DIO2125 - select SND_SOC_DMIC + select SND_SOC_DMIC if GPIOLIB select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index d50f142e2972..b88a1ee66f80 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -19,6 +19,8 @@ * */ +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/module.h> @@ -27,6 +29,34 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> +static int dmic_daiops_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct gpio_desc *dmic_en = snd_soc_dai_get_drvdata(dai); + + if (!dmic_en) + return 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + gpiod_set_value(dmic_en, 1); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + gpiod_set_value(dmic_en, 0); + break; + } + + return 0; +} + +static const struct snd_soc_dai_ops dmic_dai_ops = { + .trigger = dmic_daiops_trigger, +}; + static struct snd_soc_dai_driver dmic_dai = { .name = "dmic-hifi", .capture = { @@ -38,8 +68,23 @@ static struct snd_soc_dai_driver dmic_dai = { | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, + .ops = &dmic_dai_ops, }; +static int dmic_codec_probe(struct snd_soc_codec *codec) +{ + struct gpio_desc *dmic_en; + + dmic_en = devm_gpiod_get_optional(codec->dev, + "dmicen", GPIOD_OUT_LOW); + if (IS_ERR(dmic_en)) + return PTR_ERR(dmic_en); + + snd_soc_codec_set_drvdata(codec, dmic_en); + + return 0; +} + static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0, SND_SOC_NOPM, 0, 0), @@ -50,7 +95,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"DMIC AIF", NULL, "DMic"}, }; -static struct snd_soc_codec_driver soc_dmic = { +static const struct snd_soc_codec_driver soc_dmic = { + .probe = dmic_codec_probe, .component_driver = { .dapm_widgets = dmic_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets), -- 2.13.3

1 0

[alsa-devel] [PATCH] ASoC: Intel: kbl: Enabling ASRC for RT5663 codec on kabylake platform
by Naveen M 17 Aug '17

17 Aug '17

From: Harsha Priya N <harshapriya.n(a)intel.com> Patch fixes cracking noise in rt5663 headphones for kbl platform by calling rt5663_sel_asrc_clk_src() for RT5663_AD_STEREO_FILTER to set ASRC. The ASRC function is for asynchronous MCLK and LRCK. For RT5663 ASRC should be enabled to support special i2s clock format like Intel's 100fs. ASRC function will track i2s clock and generate corresponding system clock for codec. Calling this func helps select clock source for both RT5663_AD_STEREO_FILTER and RT5663_DA_STEREO_FILTER filters which fixes the crackling sound. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Shruthi Sudhakar <shruthi.sudhakar(a)intel.com> Signed-off-by: Naveen M <naveen.m(a)intel.com> diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index f9ba977..29b50e4 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -312,11 +312,13 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); /* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */ - rt5663_sel_asrc_clk_src(codec_dai->codec, RT5663_DA_STEREO_FILTER, 1); + rt5663_sel_asrc_clk_src(codec_dai->codec, + RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER, + RT5663_CLK_SEL_I2S1_ASRC); + ret = snd_soc_dai_set_sysclk(codec_dai, + RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); if (ret < 0) dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); -- 1.9.1

3 2

[alsa-devel] [PATCH 0/7] ASoC: make snd_soc_ops const
by Bhumika Goyal 17 Aug '17

17 Aug '17

Make these const. Done using Coccinelle. Bhumika Goyal (7): ASoC: aux1x: make snd_soc_ops const ASoC: cirrus: make snd_soc_ops const ASoC: generic: make snd_soc_ops const ASoC: pxa: make snd_soc_ops const ASoC: rockchip: make snd_soc_ops const ASoC: samsung: make snd_soc_ops const ASoC: sh: make snd_soc_ops const sound/soc/au1x/db1200.c | 2 +- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/snappercl15.c | 2 +- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/audio-graph-scu-card.c | 2 +- sound/soc/pxa/raumfeld.c | 2 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 2 +- sound/soc/samsung/jive_wm8750.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 2 +- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/sh/migor.c | 2 +- 12 files changed, 12 insertions(+), 12 deletions(-) -- 1.9.1

3 14

[alsa-devel] [PATCH 0/2] ASoC: make snd_soc_dai_driver and snd_soc_component_driver const
by Bhumika Goyal 17 Aug '17

17 Aug '17

Make these const. Bhumika Goyal (2): ASoC: bcm: make snd_soc_dai_driver const ASoC: codecs: make snd_soc_dai_driver and snd_soc_component_driver const sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/codecs/hdmi-codec.c | 2 +- sound/soc/codecs/lm4857.c | 2 +- sound/soc/codecs/max9768.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH v2 0/3] ASoC: rockchip: Parse dai links from dts
by Jeffy Chen 17 Aug '17

17 Aug '17

Currently we are using a fixed list of dai links in the driver. This serial of patches would let the driver parse dai links from dts, so that we can disable some of them for future boards in the dts. Tested on my chromebook bob(with cros 4.4 kernel), it still works after disabled rt5514 codec in the dts. Changes in v2: Let rockchip,codec-names be a required property, because we plan to add more supported codecs to the fixed dai link list in the driver. Let rockchip,codec-names be a required property. Jeffy Chen (3): ASoC: rockchip: Parse dai links from dts arm64: dts: rockchip: Add rockchip,codec-names property dt-bindings: ASoC: rockchip: Add rockchip,codec-names property .../bindings/sound/rockchip,rk3399-gru-sound.txt | 2 + arch/arm64/boot/dts/rockchip/rk3399-gru.dtsi | 1 + sound/soc/rockchip/rk3399_gru_sound.c | 125 +++++++++++++-------- 3 files changed, 84 insertions(+), 44 deletions(-) -- 2.1.4

5 9

[alsa-devel] [PATCH] ALSA: usb-audio: Add mute TLV for playback volumes on C-Media devices
by Takashi Iwai 17 Aug '17

17 Aug '17

C-Media devices (at least some models) mute the playback stream when volumes are set to the minimum value. But this isn't informed via TLV and the user-space, typically PulseAudio, gets confused as if it's still played in a low volume. This patch adds the new flag, min_mute, to struct usb_mixer_elem_info for indicating that the mixer element is with the minimum-mute volume. This flag is set for known C-Media devices in snd_usb_mixer_fu_apply_quirk() in turn. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196669 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 2 ++ sound/usb/mixer.h | 1 + sound/usb/mixer_quirks.c | 6 ++++++ 3 files changed, 9 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 082736c539bc..e630813c5008 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -542,6 +542,8 @@ int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, if (size < sizeof(scale)) return -ENOMEM; + if (cval->min_mute) + scale[0] = SNDRV_CTL_TLVT_DB_MINMAX_MUTE; scale[2] = cval->dBmin; scale[3] = cval->dBmax; if (copy_to_user(_tlv, scale, sizeof(scale))) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 3417ef347e40..2b4b067646ab 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -64,6 +64,7 @@ struct usb_mixer_elem_info { int cached; int cache_val[MAX_CHANNELS]; u8 initialized; + u8 min_mute; void *private_data; }; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index e3d1dec48ee4..e1e7ce9ab217 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1878,6 +1878,12 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, if (unitid == 7 && cval->control == UAC_FU_VOLUME) snd_dragonfly_quirk_db_scale(mixer, cval, kctl); break; + /* lowest playback value is muted on C-Media devices */ + case USB_ID(0x0d8c, 0x000c): + case USB_ID(0x0d8c, 0x0014): + if (strstr(kctl->id.name, "Playback")) + cval->min_mute = 1; + break; } } -- 2.14.1

1 0

[alsa-devel] [PATCH 00/15] constify pnp_device_id and pnp_card_device_id
by Arvind Yadav 17 Aug '17

17 Aug '17

*_device_id are not supposed to change at runtime. All functions working with *_device_id provided by <linux/pnp.h> work with const *_device_id. So mark the non-const structs as const. Arvind Yadav (15): [PATCH 01/15] ALSA: drivers: mpu401: constify pnp_device_id [PATCH 02/15] ALSA: es18xx: constify pnp_device_id and pnp_card_device_id [PATCH 03/15] ALSA: opl3sa2: constify pnp_device_id and pnp_card_device_id [PATCH 04/15] ALSA: ad1816a: constify pnp_card_device_id [PATCH 05/15] ALSA: als100: constify pnp_card_device_id [PATCH 06/15] ALSA: azt2320: constify pnp_card_device_id [PATCH 07/15] ALSA: cmi8330: constify pnp_card_device_id [PATCH 08/15] ALSA: cs4236: constify pnp_card_device_id [PATCH 09/15] ALSA: es1688: constify pnp_card_device_id [PATCH 10/15] ALSA: gus: constify pnp_card_device_id [PATCH 11/15] ALSA: msnd: constify pnp_card_device_id [PATCH 12/15] ALSA: opti9xx: constify pnp_card_device_id [PATCH 13/15] ALSA: sb16: constify pnp_card_device_id [PATCH 14/15] ALSA: sscape: constify pnp_card_device_id [PATCH 15/15] ALSA: wavefront: constify pnp_card_device_id sound/drivers/mpu401/mpu401.c | 2 +- sound/isa/ad1816a/ad1816a.c | 2 +- sound/isa/als100.c | 2 +- sound/isa/azt2320.c | 2 +- sound/isa/cmi8330.c | 2 +- sound/isa/cs423x/cs4236.c | 2 +- sound/isa/es1688/es1688.c | 2 +- sound/isa/es18xx.c | 4 ++-- sound/isa/gus/interwave.c | 2 +- sound/isa/msnd/msnd_pinnacle.c | 2 +- sound/isa/opl3sa2.c | 4 ++-- sound/isa/opti9xx/miro.c | 2 +- sound/isa/opti9xx/opti92x-ad1848.c | 2 +- sound/isa/sb/sb16.c | 2 +- sound/isa/sscape.c | 2 +- sound/isa/wavefront/wavefront.c | 2 +- 16 files changed, 18 insertions(+), 18 deletions(-) -- 2.7.4

2 16

[alsa-devel] [PATCH 00/12] ALSA: make snd_pcm_hardware const
by Bhumika Goyal 17 Aug '17

17 Aug '17

Make these const. Bhumika Goyal (12): ALSA: arm: make snd_pcm_hardware const ALSA: atmel: make snd_pcm_hardware const ALSA: drivers: make snd_pcm_hardware const ALSA: isa: make snd_pcm_hardware const ALSA: mips: make snd_pcm_hardware const ALSA: pci: make snd_pcm_hardware const ALSA: pcmcia: make snd_pcm_hardware const ALSA: ppc: make snd_pcm_hardware const ALSA: sh: make snd_pcm_hardware const ALSA: sparc: make snd_pcm_hardware const ALSA: usb: make snd_pcm_hardware const ALSA: parisc: make snd_pcm_hardware const sound/arm/aaci.c | 2 +- sound/atmel/ac97c.c | 2 +- sound/drivers/aloop.c | 2 +- sound/drivers/dummy.c | 2 +- sound/drivers/ml403-ac97cr.c | 4 ++-- sound/drivers/pcsp/pcsp_lib.c | 2 +- sound/drivers/vx/vx_pcm.c | 4 ++-- sound/isa/ad1816a/ad1816a_lib.c | 4 ++-- sound/isa/es1688/es1688_lib.c | 4 ++-- sound/isa/es18xx.c | 4 ++-- sound/isa/gus/gus_pcm.c | 4 ++-- sound/isa/msnd/msnd.c | 4 ++-- sound/isa/sb/emu8000_pcm.c | 2 +- sound/isa/sb/sb16_main.c | 4 ++-- sound/isa/sb/sb8_main.c | 4 ++-- sound/isa/wss/wss_lib.c | 4 ++-- sound/mips/hal2.c | 2 +- sound/mips/sgio2audio.c | 2 +- sound/parisc/harmony.c | 4 ++-- sound/pci/au88x0/au88x0_pcm.c | 8 ++++---- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 2 +- sound/ppc/pmac.c | 4 ++-- sound/sh/aica.c | 2 +- sound/sh/sh_dac_audio.c | 2 +- sound/sparc/amd7930.c | 2 +- sound/sparc/cs4231.c | 4 ++-- sound/sparc/dbri.c | 2 +- sound/usb/pcm.c | 2 +- 28 files changed, 44 insertions(+), 44 deletions(-) -- 1.9.1

2 13

[alsa-devel] [PATCH v3 0/5] ASoC: codecs: msm8916-wcd-analog: Add support to MBHC
by srinivas.kandagatla＠linaro.org 17 Aug '17

17 Aug '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset adds support to MBHC(Multibutton headset control) block in PM8916 analog block. MBHC support comes from 2 blocks first mechanical headset detection and second headset type, 5 button detection. This patchset adds support to: 1> Support to NC and NO type of headset Jacks. 2> Mechanical insertion and detection of headset jack. 3> Detect a 3 pole Headphone and a 4 pole Headset. 4> Detect 5 buttons. Damien sent a similar patchset to add support to mechanical detection, but that patch has issues and will not work on most usecases (for example after a playback/capture session, multicodec case). So I only picked up the BIT mask patch from that series. Tested it on DB410c with Audio Mezz board with 4 pole and 3 pole and during playback and recording use cases. Changes since v2(https://lkml.org/lkml/2017/8/2/609): - cleaned up code spotted by Damien - dropped fix snd_soc_codec_set_jack return error as it alredy applied Srinivas Kandagatla (5): ASoC: codecs: msm8916-wcd-analog: move codec reset to probe ASoC: codecs: msm8916-wcd-analog: get micbias voltage from dt ASoC: codecs: msm8916-wcd-analog: add MBHC support ASoC: qcom: apq8016-sbc: Add support to Headset JACK arm64: dts: apq8016-sbc: add mbhc buttons support .../bindings/sound/qcom,msm8916-wcd-analog.txt | 18 +- arch/arm64/boot/dts/qcom/apq8016-sbc.dtsi | 2 + sound/soc/codecs/msm8916-wcd-analog.c | 407 +++++++++++++++++++-- sound/soc/qcom/apq8016_sbc.c | 34 ++ 4 files changed, 437 insertions(+), 24 deletions(-) -- 2.9.3

4 14

[alsa-devel] [PATCH] ASoC: Intel: kbl: Add map for Maxim IV Feedback
by Naveen M 17 Aug '17

17 Aug '17

MAX98927 provides IV feedback on the capture widget. Here we are connecting the capture widget to SSP0_RX and SSP0_RX to the algorithm running on host. Signed-off-by: Naveen M <naveen.m(a)intel.com> Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 5b68ee9..c99b121 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -103,6 +103,11 @@ enum { { "codec0_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, + /* IV feedback path */ + { "codec0_fb_in", NULL, "ssp0 Rx"}, + { "ssp0 Rx", NULL, "Left HiFi Capture" }, + { "ssp0 Rx", NULL, "Right HiFi Capture" }, + /* DMIC */ { "dmic01_hifi", NULL, "DMIC01 Rx" }, { "DMIC01 Rx", NULL, "DMIC AIF" }, diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 60a7e72..60bae42 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -113,6 +113,11 @@ enum { { "codec1_in", NULL, "ssp0 Rx" }, { "ssp0 Rx", NULL, "AIF1 Capture" }, + /* IV feedback path */ + { "codec0_fb_in", NULL, "ssp0 Rx"}, + { "ssp0 Rx", NULL, "Left HiFi Capture" }, + { "ssp0 Rx", NULL, "Right HiFi Capture" }, + /* DMIC */ { "DMIC1L", NULL, "DMIC" }, { "DMIC1R", NULL, "DMIC" }, -- 1.9.1

2 1

[alsa-devel] [PATCH v3] ASoC: Intel: kbl_rt5663_rt5514_max98927: Add rt5514 spi dailink
by Harsha Priya 17 Aug '17

17 Aug '17

This patch adds a dai to rt5514-spi driver for wake on voice functionality. Signed-off-by: Hsin-yu Chao <hychao(a)google.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- v2 patch adds the dependency on SPI in Kconfig v3 patch changes the commit message to move the version specific info sound/soc/intel/Kconfig | 3 ++- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 9 +++++++++ 2 files changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b301bff..b3c7f55 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -255,11 +255,12 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH tristate "ASoC Audio driver for KBL with RT5663, RT5514 and MAX98927 in I2S Mode" - depends on X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C && SPI select SND_SOC_INTEL_SST select SND_SOC_INTEL_SKYLAKE select SND_SOC_RT5663 select SND_SOC_RT5514 + select SND_SOC_RT5514_SPI select SND_SOC_MAX98927 select SND_SOC_HDAC_HDMI help diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 5ec9428..b51dff6 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -65,6 +65,7 @@ enum { KBL_DPCM_AUDIO_HS_PB, KBL_DPCM_AUDIO_ECHO_REF_CP, KBL_DPCM_AUDIO_DMIC_CP, + KBL_DPCM_AUDIO_RT5514_DSP, KBL_DPCM_AUDIO_HDMI1_PB, KBL_DPCM_AUDIO_HDMI2_PB, }; @@ -480,6 +481,14 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) .capture_only = 1, .nonatomic = 1, }, + [KBL_DPCM_AUDIO_RT5514_DSP] = { + .name = "rt5514 dsp", + .stream_name = "Wake on Voice", + .cpu_dai_name = "spi-PRP0001:00", + .platform_name = "spi-PRP0001:00", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", .stream_name = "dmiccap", -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: Intel: kbl: Enabling ASRC for RT5663 codec on kabylake platform
by Naveen M 17 Aug '17

17 Aug '17

From: Harsha Priya N <harshapriya.n(a)intel.com> Patch fixes cracking noise in rt5663 headphones for kbl platform by calling rt5663_sel_asrc_clk_src() for RT5663_AD_STEREO_FILTER to set ASRC. The ASRC function is for asynchronous MCLK and LRCLK. For RT5663 ASRC should be enabled to support pcm format with 100fs. ASRC function will track i2s clock and generate corresponding system clock for codec. Calling this func helps select clock source for both RT5663_AD_STEREO_FILTER and RT5663_DA_STEREO_FILTER filters which fixes the crackling sound. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Shruthi Sudhakar <shruthi.sudhakar(a)intel.com> Signed-off-by: Naveen M <naveen.m(a)intel.com> diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index f9ba977..29b50e4 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -312,11 +312,13 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); /* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */ - rt5663_sel_asrc_clk_src(codec_dai->codec, RT5663_DA_STEREO_FILTER, 1); + rt5663_sel_asrc_clk_src(codec_dai->codec, + RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER, + RT5663_CLK_SEL_I2S1_ASRC); + ret = snd_soc_dai_set_sysclk(codec_dai, + RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); if (ret < 0) dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); -- 1.9.1

2 1

[alsa-devel] [PATCH v3 0/4] i2c: document DMA handling and add helpers for it
by Wolfram Sang 16 Aug '17

16 Aug '17

So, after revisiting old mail threads and taking part in a similar discussion on the USB list, here is what I cooked up to document and ease DMA handling for I2C within Linux. Please have a look at the documentation introduced in patch 2 for further details. All patches have been tested with a Renesas Salvator-X board (r8a7796/M3-W) and a Renesas Lager board (r8a7790/H2). A more detailed test description can be found here: http://elinux.org/Tests:I2C-core-DMA The branch can be found here: git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git renesas/topic/i2c-core-dma-v3 And big kudos to Renesas Electronics for funding this work, thank you very much! Regards, Wolfram Changes since v2: * rebased to v4.13-rc1 * helper functions are not inlined anymore but moved to i2c core * __must_check has been added to the buffer check helper * the release function has been renamed to contain 'dma' as well * documentation updates. Hopefully better wording now * removed the doubled Signed-offs * adding more potentially interested parties to CC Wolfram Sang (4): i2c: add helpers to ease DMA handling i2c: add docs to clarify DMA handling i2c: sh_mobile: use helper to decide if DMA is useful i2c: rcar: check for DMA-capable buffers Documentation/i2c/DMA-considerations | 38 ++++++++++++++++++++ drivers/i2c/busses/i2c-rcar.c | 18 +++++++--- drivers/i2c/busses/i2c-sh_mobile.c | 8 +++-- drivers/i2c/i2c-core-base.c | 68 ++++++++++++++++++++++++++++++++++++ include/linux/i2c.h | 5 +++ 5 files changed, 130 insertions(+), 7 deletions(-) create mode 100644 Documentation/i2c/DMA-considerations -- 2.11.0

5 13

[alsa-devel] [PATCH] conf: HdmiLpeAudio: add support for 3 devices
by Pierre-Louis Bossart 16 Aug '17

16 Aug '17

The LPE Audio mode on BYT/CHT supports up to 3 devices, and also supports IEC61937 passthrough. Add missing alsa-lib configurations so that apps can use the usual -D'hdmi:CARD=X,DEV=Y,AES0=0x[4|6]' syntax, e.g. aplay -D'hdmi:CARD=0,DEV=2,AES0=0x6' -c2 -r48000 -fs16_le ac3_surround_test.spdif Tested on Zotac PI330 with Onkyo receiver Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- src/conf/cards/HdmiLpeAudio.conf | 77 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 77 insertions(+) diff --git a/src/conf/cards/HdmiLpeAudio.conf b/src/conf/cards/HdmiLpeAudio.conf index dae71fa..9fa30da 100644 --- a/src/conf/cards/HdmiLpeAudio.conf +++ b/src/conf/cards/HdmiLpeAudio.conf @@ -56,6 +56,83 @@ HdmiLpeAudio.pcm.hdmi.0 { { interface PCM name "IEC958 Playback Default" + device 0 + lock true + preserve true + value [ $AES0 $AES1 $AES2 $AES3 ] + } + ] + } +} + +HdmiLpeAudio.pcm.hdmi.1 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { + type string + } + @args.AES0 { + type integer + } + @args.AES1 { + type integer + } + @args.AES2 { + type integer + } + @args.AES3 { + type integer + } + type hooks + slave.pcm { + type hw + card $CARD + device 1 + } + hooks.0 { + type ctl_elems + hook_args [ + { + interface PCM + name "IEC958 Playback Default" + device 1 + lock true + preserve true + value [ $AES0 $AES1 $AES2 $AES3 ] + } + ] + } +} + +HdmiLpeAudio.pcm.hdmi.2 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { + type string + } + @args.AES0 { + type integer + } + @args.AES1 { + type integer + } + @args.AES2 { + type integer + } + @args.AES3 { + type integer + } + type hooks + slave.pcm { + type hw + card $CARD + device 2 + } + hooks.0 { + type ctl_elems + hook_args [ + { + interface PCM + name "IEC958 Playback Default" + device 2 lock true preserve true value [ $AES0 $AES1 $AES2 $AES3 ] -- 2.9.3

2 1

[alsa-devel] IEC61937 passthrough with BYT/CHT HdmiLpeAudio
by Pierre-Louis Bossart 16 Aug '17

16 Aug '17

Hi Takashi, While I was looking at IEC61937, I ran a quick test on my BYT and CHT boxes, and as I suspected IEC61937 works fine since your modifications to add s16_le support. I can get AC3 to my HDMI receiver without problems just by using the hw:0,<1,2,3> device, depending on which port is used. e.g. aplay -Dhw:0,2 -c2 -r48000 -fs16le ac3_surround_test.spdif To make things more usable for applications, and maybe support receivers who actually care about the AES0 non-audio bit or reduce the initial lock time, it'd be nice to support the traditional command line, e.g. aplay -D'hdmi:CARD=0,DEV=<0,1,2>,AES0=0x6' -c2 -r48000 -fs16_le ac3_surround_test.spdif However we seem to have a mix of alsa-lib configuration issues and maybe a miss at the driver level I added the missing devices in HdmiLpeAudio.conf with the patch below, which helps select the right output, but somehow the AES0 non-audio bit doesn't seem to be set. [plb@ZOTAC PASSTHROUGH]$ amixer -Dhw:0 cget numid=12 numid=12,iface=PCM,name='IEC958 Playback Default',device=2 ; type=IEC958,access=rw------,values=1 : values=[AES0=0x00 AES1=0x00 AES2=0x00 AES3=0x00] [plb@ZOTAC PASSTHROUGH]$ aplay -D'hdmi:CARD=0,DEV=2,AES0=0x6' -c2 -r48000 -fs16_le ac3_surround_test.spdif Playing raw data 'ac3_surround_test.spdif' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo ^CAborted by signal Interrupt... [plb@ZOTAC PASSTHROUGH]$ amixer -Dhw:0 cget numid=12numid=12,iface=PCM,name='IEC958 Playback Default',device=2 ; type=IEC958,access=rw------,values=1 : values=[AES0=0x00 AES1=0x00 AES2=0x00 AES3=0x00] Could it be that we are missing the 'index' which is used in the HDaudio case to access the 'IEC958 Playback Default' controls? Thanks -Pierre diff HdmiLpeAudio.conf /usr/share/alsa/cards/HdmiLpeAudio.conf 59,135d58 < device 0 < lock true < preserve true < value [ $AES0 $AES1 $AES2 $AES3 ] < } < ] < } < } < < HdmiLpeAudio.pcm.hdmi.1 { < @args [ CARD AES0 AES1 AES2 AES3 ] < @args.CARD { < type string < } < @args.AES0 { < type integer < } < @args.AES1 { < type integer < } < @args.AES2 { < type integer < } < @args.AES3 { < type integer < } < type hooks < slave.pcm { < type hw < card $CARD < device 1 < } < hooks.0 { < type ctl_elems < hook_args [ < { < interface PCM < name "IEC958 Playback Default" < device 1 < lock true < preserve true < value [ $AES0 $AES1 $AES2 $AES3 ] < } < ] < } < } < < HdmiLpeAudio.pcm.hdmi.2 { < @args [ CARD AES0 AES1 AES2 AES3 ] < @args.CARD { < type string < } < @args.AES0 { < type integer < } < @args.AES1 { < type integer < } < @args.AES2 { < type integer < } < @args.AES3 { < type integer < } < type hooks < slave.pcm { < type hw < card $CARD < device 2 < } < hooks.0 { < type ctl_elems < hook_args [ < { < interface PCM < name "IEC958 Playback Default" < device 2

2 2

[alsa-devel] [PATCH] alsabat: fix the buffer size dependency issue
by Zhang Keqiao 16 Aug '17

16 Aug '17

In round trip latency test, the latency time is depends on the buffer size, before the test, it will playing one second 0 data to synchronize the playback thread and the recording thread, then begin playing a ~1000 HZ sine wave and record it. Stop counting and playing if the input's loudness is higher than the threshold. The issue here is before catching the sine wave, the buffer is full with 0 data need to deal with, so this latency is also counted. This patch is intends to use flag to do the thread synchronization instead of playing 0 data, once recording thread is ready, playing the ~1000 HZ sine wave and start over this process until the final test is over. Signed-off-by: Zhang Keqiao <keqiaox.k.zhang(a)linux.intel.com> --- bat/alsa.c | 26 +++++++++++++++++++++++--- bat/common.h | 4 +++- bat/latencytest.c | 47 +++++++++++++++++++++++++---------------------- 3 files changed, 51 insertions(+), 26 deletions(-) diff --git a/bat/alsa.c b/bat/alsa.c index 7613f44..5af7766 100644 --- a/bat/alsa.c +++ b/bat/alsa.c @@ -351,6 +351,14 @@ static int latencytest_process_output(struct pcm_container *sndpcm, bat->latency.is_playing = true; + /* set playback ready */ + bat->latency.state = LATENCY_STATE_PLAY_READY; + + /* wait for capture ready */ + while (bat->latency.state != LATENCY_STATE_RECORD_AND_LISTEN) { + usleep(10); + } + while (1) { /* generate output data */ err = handleoutput(bat, sndpcm->buffer, bytes, frames); @@ -365,10 +373,16 @@ static int latencytest_process_output(struct pcm_container *sndpcm, if (bat->latency.xrun_error == true) break; + /* Final test is over */ if (bat->latency.state == LATENCY_STATE_COMPLETE_SUCCESS) break; bat->periods_played++; + + /* One time success, start over */ + if (bat->latency.state == LATENCY_STATE_ONE_TIME_SUCCESS) + break; + } bat->latency.is_playing = false; @@ -441,6 +455,7 @@ void *playback_alsa(struct bat *bat) int err = 0; struct pcm_container sndpcm; +LATENCY_TEST_START: fprintf(bat->log, _("Entering playback thread (ALSA).\n")); retval_play = 0; @@ -501,6 +516,11 @@ exit3: free(sndpcm.buffer); exit2: snd_pcm_close(sndpcm.handle); + +/* Latency test is not finished, start over */ + if (bat->latency.is_capturing == true) + goto LATENCY_TEST_START; + exit1: pthread_exit(&retval_play); } @@ -641,15 +661,15 @@ static int latencytest_process_input(struct pcm_container *sndpcm, if (err != 0) break; - if (bat->latency.is_playing == false) - break; - /* write the chunk to file */ if (fwrite(sndpcm->buffer, 1, size, fp) != size) { err = -EIO; break; } + if (bat->latency.state == LATENCY_STATE_COMPLETE_SUCCESS) + break; + bytes_read += size; } diff --git a/bat/common.h b/bat/common.h index 1b07fbe..2ab3e86 100644 --- a/bat/common.h +++ b/bat/common.h @@ -158,8 +158,10 @@ enum latency_state { LATENCY_STATE_COMPLETE_FAILURE = -1, LATENCY_STATE_COMPLETE_SUCCESS = 0, LATENCY_STATE_MEASURE_FOR_1_SECOND, - LATENCY_STATE_PLAY_AND_LISTEN, + LATENCY_STATE_PLAY_READY, + LATENCY_STATE_RECORD_AND_LISTEN, LATENCY_STATE_WAITING, + LATENCY_STATE_ONE_TIME_SUCCESS, }; struct pcm { diff --git a/bat/latencytest.c b/bat/latencytest.c index fae191c..e7727e9 100644 --- a/bat/latencytest.c +++ b/bat/latencytest.c @@ -55,8 +55,6 @@ static void play_and_listen(struct bat *bat, void *buffer, int frames) int averageinput; int n = 0; float sum = 0; - float max = 0; - float min = 100000.0f; short int *input; int num = bat->latency.number; @@ -69,17 +67,17 @@ static void play_and_listen(struct bat *bat, void *buffer, int frames) /* Check the location when it became loud enough */ while (n < frames) { - if (*input++ > bat->latency.threshold) + if (*input > bat->latency.threshold) break; - *input += bat->channels; + input += bat->channels; n++; } /* Now we get the total round trip latency*/ bat->latency.samples += n; - /* Expect at least 1 buffer of round trip latency. */ - if (bat->latency.samples > frames) { + /* Round trip latency may less than one buffer */ + if (1) { bat->latency.result[num - 1] = (float) bat->latency.samples * 1000 / bat->rate; fprintf(bat->log, @@ -88,22 +86,14 @@ static void play_and_listen(struct bat *bat, void *buffer, int frames) (int) bat->latency.result[num - 1]); for (n = 0; n < num; n++) { - if (bat->latency.result[n] > max) - max = bat->latency.result[n]; - if (bat->latency.result[n] < min) - min = bat->latency.result[n]; sum += bat->latency.result[n]; } - - /* The maximum is higher than the minimum's double */ - if (max / min > 2.0f) { - bat->latency.state = - LATENCY_STATE_COMPLETE_FAILURE; - bat->latency.is_capturing = false; - return; + /* This is unnecessary, round trip lantency can be less than one + * period size, so this case is likely happen unless the period + * size is samll enough than the round trip latency. */ /* Final results */ - } else if (num == LATENCY_TEST_NUMBER) { + if (num == LATENCY_TEST_NUMBER) { bat->latency.final_result = (int) (sum / LATENCY_TEST_NUMBER); fprintf(bat->log, @@ -113,11 +103,12 @@ static void play_and_listen(struct bat *bat, void *buffer, int frames) bat->latency.state = LATENCY_STATE_COMPLETE_SUCCESS; bat->latency.is_capturing = false; + bat->latency.is_playing = false; return; /* Next step */ } else - bat->latency.state = LATENCY_STATE_WAITING; + bat->latency.state = LATENCY_STATE_ONE_TIME_SUCCESS; bat->latency.number++; @@ -194,17 +185,23 @@ int handleinput(struct bat *bat, void *buffer, int frames) /* 1 second elapsed */ if (bat->latency.samples >= bat->rate) { calculate_threshold(bat); - bat->latency.state = LATENCY_STATE_PLAY_AND_LISTEN; + /* Capture preparation is ready, start the latency test */ + bat->latency.state = LATENCY_STATE_RECORD_AND_LISTEN; bat->latency.samples = 0; bat->latency.sum = 0; } break; /* Playing sine wave and listening if it comes back */ - case LATENCY_STATE_PLAY_AND_LISTEN: + case LATENCY_STATE_RECORD_AND_LISTEN: play_and_listen(bat, buffer, frames); break; + /* playback is ready, trigger recording */ + case LATENCY_STATE_PLAY_READY: + bat->latency.state = LATENCY_STATE_MEASURE_FOR_1_SECOND; + break; + /* Waiting 1 second */ case LATENCY_STATE_WAITING: bat->latency.samples += frames; @@ -216,6 +213,12 @@ int handleinput(struct bat *bat, void *buffer, int frames) } break; + /* wait for next round playback ready */ + case LATENCY_STATE_ONE_TIME_SUCCESS: + if (bat->latency.state == LATENCY_STATE_PLAY_READY) + bat->latency.state = LATENCY_STATE_WAITING; + break; + default: return 0; } @@ -232,7 +235,7 @@ int handleoutput(struct bat *bat, void *buffer, int bytes, int frames) && bat->latency.is_capturing == false) return bat->latency.state; - if (bat->latency.state == LATENCY_STATE_PLAY_AND_LISTEN) + if (bat->latency.state == LATENCY_STATE_RECORD_AND_LISTEN) err = generate_sine_wave(bat, frames, buffer); else /* Output silence */ -- 2.9.3

2 1

[alsa-devel] [PATCH v2] ASoC: samsung: i2s: Null pointer dereference on samsung_i2s_remove
by Anton Vasilyev 16 Aug '17

16 Aug '17

If (quirks & QUIRK_SEC_DAI == 0) then samsung_i2s_probe() doesn't allocate sec_dai and pri_dai->sec_dai remains Null, but samsung_i2s_remove() permorms pri_dai->sec_dai dereference in any case. The patch adds sec_dai check on Null before derefence at samsung_i2s_remove(). Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Anton Vasilyev <vasilyev(a)ispras.ru> --- v2: Drop initialization of sec_dai at samsung_i2s_remove as Sylwester Nawrocki suggest. --- sound/soc/samsung/i2s.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index af3ba4d..6f896e3 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1376,13 +1376,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) static int samsung_i2s_remove(struct platform_device *pdev) { - struct i2s_dai *pri_dai, *sec_dai; + struct i2s_dai *pri_dai; pri_dai = dev_get_drvdata(&pdev->dev); - sec_dai = pri_dai->sec_dai; - - pri_dai->sec_dai = NULL; - sec_dai->pri_dai = NULL; pm_runtime_get_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: compress: Set reasonable compress id string
by Peng Donglin 16 Aug '17

16 Aug '17

For dynamic compress rtd, the codec_dai may not have any actual meaning(like snd-soc-dummy-dai), so compress id can just use the value of dai_link->stream_name. But for others, its codec_dai has actual meaning, so compress id can include codec_dai name. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/soc-compress.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 206f36b..b6a0714 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -737,9 +737,6 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* check client and interface hw capabilities */ - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); - if (codec_dai->driver->playback.channels_min) playback = 1; if (codec_dai->driver->capture.channels_min) @@ -758,7 +755,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) return -EINVAL; } - if(playback) + if (playback) direction = SND_COMPRESS_PLAYBACK; else direction = SND_COMPRESS_CAPTURE; @@ -797,19 +794,18 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) else if (rtd->dai_link->dpcm_capture) be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); - } else + } else { + snprintf(new_name, sizeof(new_name), "%s %s-%d", + rtd->dai_link->stream_name, codec_dai->name, num); + memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); + } /* Add copy callback for not memory mapped DSPs */ if (platform->driver->compr_ops && platform->driver->compr_ops->copy) compr->ops->copy = soc_compr_copy; mutex_init(&compr->lock); - - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, - rtd->codec_dai->name, num); - ret = snd_compress_new(rtd->card->snd_card, num, direction, new_name, compr); if (ret < 0) { -- 1.9.1

2 1

[alsa-devel] [PATCH 0/3] ALSA: make snd_kcontrol_new const
by Bhumika Goyal 16 Aug '17

16 Aug '17

Make these structures const. Done using Coccinelle. Bhumika Goyal (3): ALSA: aoa: make snd_kcontrol_new const ALSA: pcxhr: make snd_kcontrol_new const ALSA: hda: make snd_kcontrol_new const sound/aoa/codecs/onyx.c | 12 ++++++------ sound/pci/hda/patch_analog.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/pcxhr/pcxhr_mixer.c | 4 ++-- 4 files changed, 11 insertions(+), 11 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC: Medfield: Delete an error message for a failed memory allocation in snd_mfld_mc_probe()
by SF Markus Elfring 16 Aug '17

16 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 11 Aug 2017 11:25:41 +0200 Omit an extra message for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/intel/boards/mfld_machine.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c index 4e08885f37aa..6f44acfb4aae 100644 --- a/sound/soc/intel/boards/mfld_machine.c +++ b/sound/soc/intel/boards/mfld_machine.c @@ -376,10 +376,8 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) /* audio interrupt base of SRAM location where * interrupts are stored by System FW */ mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC); - if (!mc_drv_ctx) { - pr_err("allocation failed\n"); + if (!mc_drv_ctx) return -ENOMEM; - } irq_mem = platform_get_resource_byname( pdev, IORESOURCE_MEM, "IRQ_BASE"); -- 2.14.0

3 2

[alsa-devel] [PATCH] ASoC: codecs: make snd_compr_ops const
by Bhumika Goyal 16 Aug '17

16 Aug '17

Make these const as they are only stored in the compr_ops field of a snd_soc_platform_driver structure, which is of type const. Done using Coccinelle Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 505dbc9..e09fc8f 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1203,7 +1203,7 @@ static struct regmap *cs47l24_get_regmap(struct device *dev) }, }; -static struct snd_compr_ops cs47l24_compr_ops = { +static const struct snd_compr_ops cs47l24_compr_ops = { .open = cs47l24_open, .free = wm_adsp_compr_free, .set_params = wm_adsp_compr_set_params, diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index f500692..72486bf 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2017,7 +2017,7 @@ static struct regmap *wm5102_get_regmap(struct device *dev) }, }; -static struct snd_compr_ops wm5102_compr_ops = { +static const struct snd_compr_ops wm5102_compr_ops = { .open = wm5102_open, .free = wm_adsp_compr_free, .set_params = wm_adsp_compr_set_params, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index d6fae139..858a24f 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2372,7 +2372,7 @@ static struct regmap *wm5110_get_regmap(struct device *dev) }, }; -static struct snd_compr_ops wm5110_compr_ops = { +static const struct snd_compr_ops wm5110_compr_ops = { .open = wm5110_open, .free = wm_adsp_compr_free, .set_params = wm_adsp_compr_set_params, -- 1.9.1

2 1

[alsa-devel] [PATCH 0/3] ASoC: soc-core: tidyup snd_soc_unregister_component()
by Kuninori Morimoto 16 Aug '17

16 Aug '17

Hi Mark Mainly these are for platform replace to component. 1), 2) are cosmetic patch. 3) is a littile bit aggressive patch which unregister all component by 1 call. Previous my patch-set tried to add new remove component function, but if we can unregister all component by 1 call, it is no longer needed. Kuninori Morimoto (3): 1) ASoC: soc-core: rename "cmpnt" to "component" 2) ASoC: soc-core: remove duplicate mutex_unlock from snd_soc_unregister_component() 3) ASoC: soc-core: snd_soc_unregister_component() unregister all component include/sound/soc.h | 4 ++-- sound/soc/soc-core.c | 60 +++++++++++++++++++++++++++++++--------------------- 2 files changed, 38 insertions(+), 26 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH v2] ASoC: compress: Set reasonable compress id string
by Donglin Peng 16 Aug '17

16 Aug '17

For dynamic compress rtd, the codec_dai may not have any actual meaning(like snd-soc-dummy-dai), so compress id can just use the value of dai_link->stream_name. But for others, its codec_dai has actual meaning, so compress id can include codec_dai name. Signed-off-by: pengdonglin <pengdonglin137(a)163.com> --- sound/soc/soc-compress.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 206f36bf43e8..b6a071461b10 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -737,9 +737,6 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* check client and interface hw capabilities */ - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); - if (codec_dai->driver->playback.channels_min) playback = 1; if (codec_dai->driver->capture.channels_min) @@ -758,7 +755,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) return -EINVAL; } - if(playback) + if (playback) direction = SND_COMPRESS_PLAYBACK; else direction = SND_COMPRESS_CAPTURE; @@ -797,19 +794,18 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) else if (rtd->dai_link->dpcm_capture) be_pcm->streams[SNDRV_PCM_STRE AM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); - } else + } else { + snprintf(new_name, sizeof(new_name), "%s %s-%d", + rtd->dai_link->stream_name, codec_dai->name, num); + memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); + } /* Add copy callback for not memory mapped DSPs */ if (platform->driver->compr_ops && platform->driver->compr_ops->c opy) compr->ops->copy = soc_compr_copy; mutex_init(&compr->lock); - - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, - rtd->codec_dai->name, num); - ret = snd_compress_new(rtd->card->snd_card, num, direction, new_name, compr); if (ret < 0) { -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: Intel: kbl: Enabling ASRC for RT5663 codec on kabylake platform
by Naveen M 16 Aug '17

16 Aug '17

From: Harsha Priya N <harshapriya.n(a)intel.com> Patch fixes cracking noise in rt5663 headphones for kbl platform by calling rt5663_sel_asrc_clk_src() for RT5663_AD_STEREO_FILTER to set ASRC. The ASRC function is for asynchronous MCLK and LRCLK. For RT5663 ASRC should be enabled to support pcm format with 100fs. ASRC function will track i2s clock and generate corresponding system clock for codec. Calling this func helps select clock source for both RT5663_AD_STEREO_FILTER and RT5663_DA_STEREO_FILTER filters which fixes the crackling sound. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Shruthi Sudhakar <shruthi.sudhakar(a)intel.com> Signed-off-by: Naveen M <naveen.m(a)intel.com> diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index f9ba977..29b50e4 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -312,11 +312,13 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); /* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */ - rt5663_sel_asrc_clk_src(codec_dai->codec, RT5663_DA_STEREO_FILTER, 1); + rt5663_sel_asrc_clk_src(codec_dai->codec, + RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER, + RT5663_CLK_SEL_I2S1_ASRC); + ret = snd_soc_dai_set_sysclk(codec_dai, + RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); if (ret < 0) dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); -- 1.9.1

1 1

[alsa-devel] alsa settings for sound device Intel cherry trail chipset
by fulgor fulgor 16 Aug '17

16 Aug '17

hello from europe I can provide logging informations http://www.alsa-project.org/db/?f=6fb68d0201f421b41fe65677fde2ec5b1c9d4429 Thanks you -- Lionel Van Winnendaele

1 0

[alsa-devel] [PATCH v4] ASoC: samsung: i2s: Null pointer dereference on samsung_i2s_remove
by Anton Vasilyev 15 Aug '17

15 Aug '17

If (quirks & QUIRK_SEC_DAI == 0) then samsung_i2s_probe() doesn't allocate sec_dai and pri_dai->sec_dai remains Null, but samsung_i2s_remove() performs pri_dai->sec_dai dereference in any case. The patch removes useless reinitialization of sec_dai at samsung_i2s_remove(), because resources are under devm control. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Anton Vasilyev <vasilyev(a)ispras.ru> --- v2: Drop initialization of sec_dai by NULL at samsung_i2s_remove as Sylwester Nawrocki suggest. --- v3: Fix typo in the comment --- v4: Change comment based on Krzysztof review. --- sound/soc/samsung/i2s.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index af3ba4d..6f896e3 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1376,13 +1376,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) static int samsung_i2s_remove(struct platform_device *pdev) { - struct i2s_dai *pri_dai, *sec_dai; + struct i2s_dai *pri_dai; pri_dai = dev_get_drvdata(&pdev->dev); - sec_dai = pri_dai->sec_dai; - - pri_dai->sec_dai = NULL; - sec_dai->pri_dai = NULL; pm_runtime_get_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: Freescale: Delete an error message for a failed memory allocation in three functions" to the asoc tree
by Mark Brown 15 Aug '17

15 Aug '17

The patch ASoC: Freescale: Delete an error message for a failed memory allocation in three functions has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 443be77e4b09d98c28196dca7132a1191ecf742b Mon Sep 17 00:00:00 2001 From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 11 Aug 2017 15:10:43 +0200 Subject: [PATCH] ASoC: Freescale: Delete an error message for a failed memory allocation in three functions Omit an extra message for a memory allocation failure in these functions. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_asrc_dma.c | 4 +--- sound/soc/fsl/fsl_dma.c | 1 - sound/soc/fsl/fsl_ssi.c | 4 +--- 3 files changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 282d841840b1..2baf19608bd0 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -279,10 +279,8 @@ static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream) struct fsl_asrc_pair *pair; pair = kzalloc(sizeof(struct fsl_asrc_pair), GFP_KERNEL); - if (!pair) { - dev_err(dev, "failed to allocate pair\n"); + if (!pair) return -ENOMEM; - } pair->asrc_priv = asrc_priv; diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 979ed1b3ecc4..c17359648915 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -905,7 +905,6 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL); if (!dma) { - dev_err(&pdev->dev, "could not allocate dma object\n"); of_node_put(ssi_np); return -ENOMEM; } diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 173cb8496641..64598d1183f8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1432,10 +1432,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) ssi_private = devm_kzalloc(&pdev->dev, sizeof(*ssi_private), GFP_KERNEL); - if (!ssi_private) { - dev_err(&pdev->dev, "could not allocate DAI object\n"); + if (!ssi_private) return -ENOMEM; - } ssi_private->soc = of_id->data; ssi_private->dev = &pdev->dev; -- 2.13.3

1 0

[alsa-devel] [PATCH 0/5] constify snd_soc_dai_ops structures
by Julia Lawall 15 Aug '17

15 Aug '17

These snd_soc_dai_ops structures are only stored in the ops field of a snd_soc_dai_driver structure, which is const. Thus, the snd_soc_dai_ops structures can be const too. Done with the help of Coccinelle. --- sound/soc/blackfin/bf6xx-i2s.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/zx_aud96p22.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/spear/spdif_in.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-)

2 9

[alsa-devel] [PATCH] ASoC: spear: spdif_in: constify snd_soc_dai_ops structure
by Gustavo A. R. Silva 15 Aug '17

15 Aug '17

This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/spear/spdif_in.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 7f32527..d9368e1 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -151,7 +151,7 @@ static int spdif_in_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static struct snd_soc_dai_ops spdif_in_dai_ops = { +static const struct snd_soc_dai_ops spdif_in_dai_ops = { .shutdown = spdif_in_shutdown, .trigger = spdif_in_trigger, .hw_params = spdif_in_hw_params, -- 2.5.0

3 2

[alsa-devel] [PATCH] arizona: anc: Correct setting of FCx_MIC_MODE_SEL
by Charles Keepax 15 Aug '17

15 Aug '17

From: Sapthagiri Baratam <sapthagiri.baratam(a)cirrus.com> The mask rather than the shift is accidentally passed to the SOC_ENUM_SINGLE macro which results in the wrong bits being set for this control, change to using the shift to correct this. Signed-off-by: Sapthagiri Baratam <sapthagiri.baratam(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0a734d910850..a1149f6a8450 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -710,7 +710,7 @@ const struct soc_enum arizona_anc_input_src[] = { ARRAY_SIZE(arizona_anc_input_src_text), arizona_anc_input_src_text), SOC_ENUM_SINGLE(ARIZONA_FCL_ADC_REFORMATTER_CONTROL, - ARIZONA_FCL_MIC_MODE_SEL, + ARIZONA_FCL_MIC_MODE_SEL_SHIFT, ARRAY_SIZE(arizona_anc_channel_src_text), arizona_anc_channel_src_text), SOC_ENUM_SINGLE(ARIZONA_ANC_SRC, @@ -718,7 +718,7 @@ const struct soc_enum arizona_anc_input_src[] = { ARRAY_SIZE(arizona_anc_input_src_text), arizona_anc_input_src_text), SOC_ENUM_SINGLE(ARIZONA_FCR_ADC_REFORMATTER_CONTROL, - ARIZONA_FCR_MIC_MODE_SEL, + ARIZONA_FCR_MIC_MODE_SEL_SHIFT, ARRAY_SIZE(arizona_anc_channel_src_text), arizona_anc_channel_src_text), }; -- 2.11.0

2 2

[alsa-devel] [PATCH 3/3] ALSA: firewire-motu: constify snd_rawmidi_ops structures
by Julia Lawall 15 Aug '17

15 Aug '17

These snd_rawmidi_ops structures are only passed as the third argument of snd_rawmidi_set_ops. This argument is const, so the snd_rawmidi_ops structures can be const too. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/firewire/motu/motu-midi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/motu/motu-midi.c b/sound/firewire/motu/motu-midi.c index e3acfcc..e55cab6 100644 --- a/sound/firewire/motu/motu-midi.c +++ b/sound/firewire/motu/motu-midi.c @@ -128,12 +128,12 @@ static void set_midi_substream_names(struct snd_motu *motu, int snd_motu_create_midi_devices(struct snd_motu *motu) { - static struct snd_rawmidi_ops capture_ops = { + static const struct snd_rawmidi_ops capture_ops = { .open = midi_capture_open, .close = midi_capture_close, .trigger = midi_capture_trigger, }; - static struct snd_rawmidi_ops playback_ops = { + static const struct snd_rawmidi_ops playback_ops = { .open = midi_playback_open, .close = midi_playback_close, .trigger = midi_playback_trigger,

3 5

[alsa-devel] [PATCH 0/3] constify snd_rawmidi_ops structures
by Julia Lawall 15 Aug '17

15 Aug '17

These snd_rawmidi_ops structures are only passed as the third argument of snd_rawmidi_set_ops. This argument is const, so the snd_rawmidi_ops structures can be const too. Done with the help of Coccinelle. --- drivers/hid/hid-prodikeys.c | 2 +- drivers/usb/gadget/function/f_midi.c | 4 ++-- sound/firewire/motu/motu-midi.c | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-)

1 0

[alsa-devel] Headphone and headset jack control naming
by Tanu Kaskinen 15 Aug '17

15 Aug '17

A user reported various audio problems with the Alienware 14 laptop in the PulseAudio bugzilla. One of the problems is that the jack control names that alsa reports are incorrect. However, I'm not sure what the correct names would be, so I'd like to get that clarified (I will then file a proper bug report in the kernel bugzilla). The laptop has three physical jacks: headphone, headset and microphone. In general, headset jacks have been so far represented by two jack controls: "Headphone" + "Headset Mic" (or is "Headphone" + "Mic" more common? I think I've seen both being used). However, in this case there's also a headphone-only jack. Should we invent a new name for headset outputs, so that the headphone and headset outputs can be distinguished, or should there be two "Headphone" controls, or should one of the controls (which one?) use the "Headphone2" name that is already used by some hardware? FWIW, currently the output jack control names on Alienware 14 are "Headphone Front" for the headset jack and "Headphone Surround" for the headphone jack, which I assume we all agree is incorrect. Furthermore, there's no "Headset Mic" jack control, or any other control corresponding to the headset mic. "Mic" control exists, but it corresponds to the separate microphone jack. -- Tanu https://www.patreon.com/tanuk

1 0

[alsa-devel] [PATCH v3 00/11] ASoC: Add I2S support for Allwinner H3 SoCs
by codekipper＠gmail.com 15 Aug '17

15 Aug '17

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, please find attached a series of patches to bring i2s support to the Allwinner H3 SoC. This has been tested with the following setups: A20 Olimex EVB connected to a pcm5102 Orange Pi 2 connected to a uda1380 Orange Pi 2 hdmi audio playback Pine 64 connected to the audio DAC board To get i2s working some additional patches are required which will be delivered later. For now they have been pushed here https://github.com/codekipper/linux-sunxi/commits/sunxi-audio-h3 I don't own a A33 device which uses the i2s block for the audio codec so if someone could test against that it would be much appreciated. I'm also wondering if there is a preferred way of setting the lrclk size in the dts?..currently it is set to the sample width but for example the pcm5102a wants it to be 32 bits whatever the sample rate. Thanks in advance, CK --- v4 changes compared to v3 are: - moved clkdiv variant adjustment out of function - used PTR_ERR_OR_ZERO for checks - tidy up of extra lines and lines over 80 chars. - reduced names of polarity, wss and sr reg fields. - added reviewed-by to commit messages - added comments for functionality that hasn't been implemented yet. v3 changes compared to v2 are: - initial changes to prepare driver for newer SoCs has been broken down into smaller patches - reduce use of regmap fields to where just needed. - clkdiv expansion will be delivered later. - defines for H3 variant segregated. - fixed regmap config issue with SUN8I_I2S_FIFO_TX_REG. v2 changes compared to v1 are: - massive refactoring to remove duplicate code making use of regmap_fields. - extending the clock divisors. - removed code that should be delivered when we support 20/24bits --- Marcus Cooper (11): ASoC: sun4i-i2s: Add clkdiv offsets to quirks ASoC: sun4i-i2s: Add regmap config to quirks ASoC: sun4i-i2s: Add TX FIFO offset to quirks ASoC: sun4i-i2s: Add regmap fields for channels ASoC: sun4i-i2s: Add regfields for word size select and sample resolution ASoC: sun4i-i2s: bclk and lrclk polarity tidyup ASoC: sun4i-i2s: Add mclk enable regmap field ASoC: sun4i-i2s: Add regmap field to set DAI format ASoC: sun4i-i2s: Check for slave select bit ASoC: sun4i-i2s: Update global enable with bitmask ASoC: sun4i-i2s: Add support for H3 .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + sound/soc/sunxi/sun4i-i2s.c | 444 ++++++++++++++++++--- 2 files changed, 391 insertions(+), 55 deletions(-) -- 2.14.1

3 15

[alsa-devel] [PATCH] ALSA: usb-audio: Apply sample rate quirk to Sennheiser headset
by Takashi Iwai 15 Aug '17

15 Aug '17

A Senheisser headset requires the typical sample-rate quirk for avoiding spurious errors from inquiring the current sample rate like: usb 1-1: 2:1: cannot get freq at ep 0x4 usb 1-1: 3:1: cannot get freq at ep 0x83 The USB ID 1395:740a has to be added to the entries in snd_usb_get_sample_rate_quirk(). Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1052580 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d7b0b0a3a2db..f3e9e30172f3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1142,6 +1142,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */ case USB_ID(0x05A3, 0x9420): /* ELP HD USB Camera */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ + case USB_ID(0x1395, 0x740a): /* Sennheiser DECT */ case USB_ID(0x1901, 0x0191): /* GE B850V3 CP2114 audio interface */ case USB_ID(0x1de7, 0x0013): /* Phoenix Audio MT202exe */ case USB_ID(0x1de7, 0x0014): /* Phoenix Audio TMX320 */ -- 2.13.3

1 0

[alsa-devel] [PATCH] ALSA: rme9652: Use common error handling code in two functions
by SF Markus Elfring 15 Aug '17

15 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 14 Aug 2017 22:00:40 +0200 Add a jump target so that a bit of exception handling can be better reused in these functions. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/rme9652/hdsp.c | 13 ++++++------- sound/pci/rme9652/rme9652.c | 14 ++++++-------- 2 files changed, 12 insertions(+), 15 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index fe36d44d16c6..4750d14555bc 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5381,17 +5381,16 @@ static int snd_hdsp_probe(struct pci_dev *pci, card->private_free = snd_hdsp_card_free; hdsp->dev = dev; hdsp->pci = pci; - - if ((err = snd_hdsp_create(card, hdsp)) < 0) { - snd_card_free(card); - return err; - } + err = snd_hdsp_create(card, hdsp); + if (err) + goto free_card; strcpy(card->shortname, "Hammerfall DSP"); sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, hdsp->port, hdsp->irq); - - if ((err = snd_card_register(card)) < 0) { + err = snd_card_register(card); + if (err) { +free_card: snd_card_free(card); return err; } diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 150d08898db8..b36ba7f9a7bf 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -2618,19 +2618,17 @@ static int snd_rme9652_probe(struct pci_dev *pci, card->private_free = snd_rme9652_card_free; rme9652->dev = dev; rme9652->pci = pci; - - if ((err = snd_rme9652_create(card, rme9652, precise_ptr[dev])) < 0) { - snd_card_free(card); - return err; - } + err = snd_rme9652_create(card, rme9652, precise_ptr[dev]); + if (err) + goto free_card; strcpy(card->shortname, rme9652->card_name); sprintf(card->longname, "%s at 0x%lx, irq %d", card->shortname, rme9652->port, rme9652->irq); - - - if ((err = snd_card_register(card)) < 0) { + err = snd_card_register(card); + if (err) { +free_card: snd_card_free(card); return err; } -- 2.14.0

2 1

[alsa-devel] [PATCH] sound: emu8000: constify emu8000_ops
by Julia Lawall 15 Aug '17

15 Aug '17

The structure emu8000_ops is only copied into another structure, so it can be const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/isa/sb/emu8000_callback.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/sb/emu8000_callback.c b/sound/isa/sb/emu8000_callback.c index d28d712..5a48550 100644 --- a/sound/isa/sb/emu8000_callback.c +++ b/sound/isa/sb/emu8000_callback.c @@ -62,7 +62,7 @@ static int load_fx(struct snd_emux *emu, int type, int mode, /* * set up operators */ -static struct snd_emux_operators emu8000_ops = { +static const struct snd_emux_operators emu8000_ops = { .owner = THIS_MODULE, .get_voice = get_voice, .prepare = start_voice,

2 1

[alsa-devel] [PATCH] ALSA: seq: 2nd attempt at fixing race creating a q
by Daniel Mentz 15 Aug '17

15 Aug '17

commit 4842e98f26dd80be3623c4714a244ba52ea096a8 ("ALSA: seq: Fix race at creating a queue") attempted to fix a race reported by syzkaller. That fix has been described as follows: " When a sequencer queue is created in snd_seq_queue_alloc(),it adds the new queue element to the public list before referencing it. Thus the queue might be deleted before the call of snd_seq_queue_use(), and it results in the use-after-free error, as spotted by syzkaller. The fix is to reference the queue object at the right time. " The last phrase in that commit message refers to calling queue_use(q, client, 1) which increments queue->clients, but that does not prevent the DELETE_QUEUE ioctl() and queue_delete() from kfree()ing the queue. Hence, the commit is not effective at preventing the race. This commit adds code to effectively prevent the removal of the queue by calling snd_use_lock_use(). Reported-by: Dmitry Vyukov <dvyukov(a)google.com> Cc: <stable(a)vger.kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Daniel Mentz <danielmentz(a)google.com> --- sound/core/seq/seq_clientmgr.c | 13 ++++--------- sound/core/seq/seq_queue.c | 14 +++++++++----- sound/core/seq/seq_queue.h | 2 +- 3 files changed, 14 insertions(+), 15 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 272c55fe17c8..ea2d0ae85bd3 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1502,16 +1502,11 @@ static int snd_seq_ioctl_unsubscribe_port(struct snd_seq_client *client, static int snd_seq_ioctl_create_queue(struct snd_seq_client *client, void *arg) { struct snd_seq_queue_info *info = arg; - int result; struct snd_seq_queue *q; - result = snd_seq_queue_alloc(client->number, info->locked, info->flags); - if (result < 0) - return result; - - q = queueptr(result); - if (q == NULL) - return -EINVAL; + q = snd_seq_queue_alloc(client->number, info->locked, info->flags); + if (IS_ERR(q)) + return PTR_ERR(q); info->queue = q->queue; info->locked = q->locked; @@ -1521,7 +1516,7 @@ static int snd_seq_ioctl_create_queue(struct snd_seq_client *client, void *arg) if (!info->name[0]) snprintf(info->name, sizeof(info->name), "Queue-%d", q->queue); strlcpy(q->name, info->name, sizeof(q->name)); - queuefree(q); + snd_use_lock_free(&q->use_lock); return 0; } diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index 450c5187eecb..79e0c5604ef8 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -184,22 +184,26 @@ void __exit snd_seq_queues_delete(void) static void queue_use(struct snd_seq_queue *queue, int client, int use); /* allocate a new queue - - * return queue index value or negative value for error + * return pointer to new queue or ERR_PTR(-errno) for error + * The new queue's use_lock is set to 1. It is the caller's responsibility to + * call snd_use_lock_free(&q->use_lock). */ -int snd_seq_queue_alloc(int client, int locked, unsigned int info_flags) +struct snd_seq_queue *snd_seq_queue_alloc(int client, int locked, unsigned int info_flags) { struct snd_seq_queue *q; q = queue_new(client, locked); if (q == NULL) - return -ENOMEM; + return ERR_PTR(-ENOMEM); q->info_flags = info_flags; queue_use(q, client, 1); + snd_use_lock_use(&q->use_lock); if (queue_list_add(q) < 0) { + snd_use_lock_free(&q->use_lock); queue_delete(q); - return -ENOMEM; + return ERR_PTR(-ENOMEM); } - return q->queue; + return q; } /* delete a queue - queue must be owned by the client */ diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h index 30c8111477f6..719093489a2c 100644 --- a/sound/core/seq/seq_queue.h +++ b/sound/core/seq/seq_queue.h @@ -71,7 +71,7 @@ void snd_seq_queues_delete(void); /* create new queue (constructor) */ -int snd_seq_queue_alloc(int client, int locked, unsigned int flags); +struct snd_seq_queue *snd_seq_queue_alloc(int client, int locked, unsigned int flags); /* delete queue (destructor) */ int snd_seq_queue_delete(int client, int queueid); -- 2.14.0.434.g98096fd7a8-goog

2 5

[alsa-devel] Detailed information about asoc framework
by Oleksandr Müller 14 Aug '17

14 Aug '17

Hello Alsa developers! I am developing an asoc driver for a hat sound card for a raspberry pi 3. It has a pcm5142 dac (which has already a codec driver in the linux kernel code) and a pcm1863 adc which isnt supported by the codec repo. Can I find detailed information about the asoc layer, the components and accesses from the os, in which order what builds up etc. Right now I have a device tree overlay with 2 sound links and a minimal piece of code as the codec driver which even maps and shows (with aplay and arecord command) the soundcard as a capture and playback device. The overlay connects the codec driver through i2c. Do you need more information? Hope you can help me with some sort of guides or explanations to better understand. Oleksandr

1 0

Re: [alsa-devel] [PATCH 3/3] mfd: twl: move header file out of I2C realm
by Lee Jones 14 Aug '17

14 Aug '17

On Sun, 13 Aug 2017, Wolfram Sang wrote: > On Thu, Jul 06, 2017 at 08:03:52AM +0100, Lee Jones wrote: > > On Thu, 06 Jul 2017, Thierry Reding wrote: > > > > > On Mon, May 22, 2017 at 12:02:10AM +0200, Wolfram Sang wrote: > > > > include/linux/i2c is not for client devices. Move the header file to a > > > > more appropriate location. > > > > > > > > Signed-off-by: Wolfram Sang <wsa(a)the-dreams.de> > > > > --- > > > > arch/arm/mach-omap2/common.h | 2 +- > > > > arch/arm/mach-omap2/omap_twl.c | 2 +- > > > > drivers/gpio/gpio-twl4030.c | 2 +- > > > > drivers/iio/adc/twl4030-madc.c | 2 +- > > > > drivers/iio/adc/twl6030-gpadc.c | 2 +- > > > > drivers/input/keyboard/twl4030_keypad.c | 2 +- > > > > drivers/input/misc/twl4030-pwrbutton.c | 2 +- > > > > drivers/input/misc/twl4030-vibra.c | 2 +- > > > > drivers/mfd/twl-core.c | 6 +++--- > > > > drivers/mfd/twl4030-audio.c | 2 +- > > > > drivers/mfd/twl4030-irq.c | 2 +- > > > > drivers/mfd/twl4030-power.c | 2 +- > > > > drivers/mfd/twl6030-irq.c | 2 +- > > > > drivers/phy/phy-twl4030-usb.c | 2 +- > > > > drivers/power/supply/twl4030_charger.c | 2 +- > > > > drivers/pwm/pwm-twl-led.c | 2 +- > > > > drivers/pwm/pwm-twl.c | 2 +- > > > > drivers/regulator/twl-regulator.c | 2 +- > > > > drivers/regulator/twl6030-regulator.c | 2 +- > > > > drivers/rtc/rtc-twl.c | 2 +- > > > > drivers/usb/phy/phy-twl6030-usb.c | 2 +- > > > > drivers/video/backlight/pandora_bl.c | 2 +- > > > > drivers/watchdog/twl4030_wdt.c | 2 +- > > > > include/linux/{i2c => mfd}/twl.h | 0 > > > > sound/soc/codecs/twl4030.c | 2 +- > > > > 25 files changed, 26 insertions(+), 26 deletions(-) > > > > rename include/linux/{i2c => mfd}/twl.h (100%) > > > > > > I didn't see this get applied yet, so just in case anyone was waiting > > > for me (this is trivial, so I don't think there's a need): > > > > You're not the last. :) > > Given the triviality of the change for non-MFD subsystems, can we apply > this for 4.14? I can't apply anything without Acks for *all* of the subsystems above. As you well know, it is the submitters responsibility to obtain them. My suggestion would be to collect the Acks you've received up until this point and identify which SS's are still lacking in change log section of a RESEND. -- Lee Jones Linaro STMicroelectronics Landing Team Lead Linaro.org │ Open source software for ARM SoCs Follow Linaro: Facebook | Twitter | Blog

4 4

[alsa-devel] [PATCH] arizona: anc: Correct setting of FCx_MIC_MODE_SEL
by Charles Keepax 14 Aug '17

14 Aug '17

From: Sapthagiri Baratam <sapthagiri.baratam(a)cirrus.com> The mask rather than the shift is accidentally passed to the SOC_ENUM_SINGLE macro which results in the wrong bits being set for this control, change to using the shift to correct this. Signed-off-by: Sapthagiri Baratam <sapthagiri.baratam(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0a734d910850..a1149f6a8450 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -710,7 +710,7 @@ const struct soc_enum arizona_anc_input_src[] = { ARRAY_SIZE(arizona_anc_input_src_text), arizona_anc_input_src_text), SOC_ENUM_SINGLE(ARIZONA_FCL_ADC_REFORMATTER_CONTROL, - ARIZONA_FCL_MIC_MODE_SEL, + ARIZONA_FCL_MIC_MODE_SEL_SHIFT, ARRAY_SIZE(arizona_anc_channel_src_text), arizona_anc_channel_src_text), SOC_ENUM_SINGLE(ARIZONA_ANC_SRC, @@ -718,7 +718,7 @@ const struct soc_enum arizona_anc_input_src[] = { ARRAY_SIZE(arizona_anc_input_src_text), arizona_anc_input_src_text), SOC_ENUM_SINGLE(ARIZONA_FCR_ADC_REFORMATTER_CONTROL, - ARIZONA_FCR_MIC_MODE_SEL, + ARIZONA_FCR_MIC_MODE_SEL_SHIFT, ARRAY_SIZE(arizona_anc_channel_src_text), arizona_anc_channel_src_text), }; -- 2.11.0

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix pincfg for Dell quirk
by Shih-Yuan Lee (FourDollars) 14 Aug '17

14 Aug '17

The initial pin configs for Dell headset mode of ALC3271 has changed. Fixes: b4576de87243 ("ALSA: hda/realtek - Fix typo of pincfg for Dell quirk") Signed-off-by: Shih-Yuan Lee (FourDollars) <sylee(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a91a9ef..217bb58 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6647,7 +6647,6 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0299, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x12, 0xb7a60130}, - {0x13, 0xb8a61140}, {0x17, 0x90170110}), {} }; -- 2.7.4

2 3

[alsa-devel] No Headphone Jack Audio from Lenovo Docking Station with a T470 connected
by Dennis Wassenberg 14 Aug '17

14 Aug '17

Hi, are there any news on this? I tested pin cfg 0x17 0x21211010 too. In general it looks good, but I can not hear anything. Does anyone have some hints what can I do or test to get the sound working on Dock HP? Thanks, Best regards, Dennis > Hello, > > I'm unable to get any sound from the Headphone Jack of a Lenovo Ultra > Docking Station with a new Lenovo T470 connected. With a T440p however I > get sound from the Docking Station Headphone Jack. > > I did some research on my alsa configuration and found out that there is > no pin configuration supplied from the Laptop BIOS that connects to the > Docking Headphone Jack. > > The hdajacksensetest tool only shows this when the Laptop is docked and > a Headphone is pluged in: > Pin 0x18 (Black Mic, Left side): present = No > Pin 0x21 (Black Headphone, Left side): present = No > > So I did some patchingOn 14.08.2017 11:36, Stefan Mark (JIRA) wrote: > using a early firmware load patch floating around the web targeting the > same docking station for my older T440p laptop: > echo 0x17 0x21211010 > /sys/class/sound/hwC0D0/user_pin_configs > echo 1 > /sys/class/sound/hwC0D0/reconfig > > Now my Docking Station Headphone Jack is being detected! :) > > With the Headphone Connected: > Pin 0x17 (Black Headphone, Docking station, Rear side): present = Yes > Pin 0x18 (Black Mic, Left side): present = No > Pin 0x21 (Black Headphone, Left side): present = No > > And disconnected: > Pin 0x17 (Black Headphone, Docking station, Rear side): present = No > Pin 0x18 (Black Mic, Left side): present = No > Pin 0x21 (Black Headphone, Left side): present = No > > However there is STILL no sound even after unmuting everything in alsamixer! > I'm pretty sure 0x17 is the right pincfg but I'm not sure about > 0x21211010 and how I obtain such an address for my laptop. > > Could someone please help me out here or give me a hint? > > Here is a alsa-info.sh dump when the Laptop is not connected to the Dock: > https://pastebin.com/eTNnfnzL > > And here when the Laptop is connected to the Dock (No changes): > https://pastebin.com/ySBsKUeH > > And here after setting the pin config 0x17 0x21211010: > https://pastebin.com/xmDfsNtQ > > I'm sure I'm very close. But something is still missing. Any help is > much apprechiated! > > Thanks in advance > Dieter >

1 0

[alsa-devel] [PATCH 000/102] Convert drivers to explicit reset API
by Philipp Zabel 14 Aug '17

14 Aug '17

The reset control API has two modes: exclusive access, where the driver expects to have full and immediate control over the state of the reset line, and shared (clock-like) access, where drivers only request reset deassertion while active, but don't care about the state of the reset line while inactive. Commit a53e35db70d1 ("reset: Ensure drivers are explicit when requesting reset lines") started to transition the reset control request API calls to explicitly state whether the driver needs exclusive or shared reset control behavior. This series converts all drivers that currently implicitly request exclusive reset controls to the corresponding explicit API call. It is, for the most part, generated from the following semantic patch: @@ expression rstc, dev, id; @@ -rstc = reset_control_get(dev, id); +rstc = reset_control_get_exclusive(dev, id); @@ expression rstc, dev, id; @@ -rstc = reset_control_get_optional(dev, id); +rstc = reset_control_get_optional_exclusive(dev, id); @@ expression rstc, node, id; @@ -rstc = of_reset_control_get(node, id); +rstc = of_reset_control_get_exclusive(node, id); @@ expression rstc, node, index; @@ -rstc = of_reset_control_get_by_index(node, index); +rstc = of_reset_control_get_exclusive_by_index(node, index); @@ expression rstc, dev, id; @@ -rstc = devm_reset_control_get(dev, id); +rstc = devm_reset_control_get_exclusive(dev, id); @@ expression rstc, dev, id; @@ -rstc = devm_reset_control_get_optional(dev, id); +rstc = devm_reset_control_get_optional_exclusive(dev, id); @@ expression rstc, dev, index; @@ -rstc = devm_reset_control_get_by_index(dev, index); +rstc = devm_reset_control_get_exclusive_by_index(dev, index); After all driver patches are applied, the temporary transition helpers can be removed. regards Philipp Philipp Zabel (102): ARM: rockchip: explicitly request exclusive reset control ARM: socfpga: explicitly request exclusive reset control MIPS: pci-mt7620: explicitly request exclusive reset control ahci: st: explicitly request exclusive reset control ata: sata_gemini: explicitly request exclusive reset control ata: ahci_tegra: explicitly request exclusive reset control bus: sunxi-rsb: explicitly request exclusive reset control bus: tegra-gmi: explicitly request exclusive reset control clk: sunxi: explicitly request exclusive reset control clk: tegra: explicitly request exclusive reset control clocksource/drivers/timer-stm32: explicitly request exclusive reset control clocksource/drivers/sun5i: explicitly request exclusive reset control crypto: rockchip: explicitly request exclusive reset control crypto: sun4i-ss - request exclusive reset control PM / devfreq: tegra: explicitly request exclusive reset control dmaengine: stm32-dma: explicitly request exclusive reset control dmaengine: sun6i: explicitly request exclusive reset control dmaengine: tegra-apb: explicitly request exclusive reset control drm: kirin: explicitly request exclusive reset control drm/nouveau/tegra: explicitly request exclusive reset control drm/rockchip: explicitly request exclusive reset control drm/sti: explicitly request exclusive reset control drm/stm: explicitly request exclusive reset control drm/sun4i: explicitly request exclusive reset control drm/tegra: explicitly request exclusive reset control gpu: host1x: explicitly request exclusive reset control i2c: mv64xxx: explicitly request exclusive reset control i2c: stm32f4: explicitly request exclusive reset control i2c: sun6i-pw2i: explicitly request exclusive reset control i2c: tegra: explicitly request exclusive reset control iio: adc: rockchip_saradc: explicitly request exclusive reset control iio: dac: stm32-dac-core: explicitly request exclusive reset control Input: tegra-kbc - request exclusive reset control coda: explicitly request exclusive reset control st-rc: explicitly request exclusive reset control stm32-dcmi: explicitly request exclusive reset control rc: sunxi-cir: explicitly request exclusive reset control mmc: dw_mmc: explicitly request exclusive reset control mmc: sdhci-st: explicitly request exclusive reset control mmc: sunxi: explicitly request exclusive reset control mmc: tegra: explicitly request exclusive reset control mtd: nand: sunxi: explicitly request exclusive reset control mtd: spi-nor: stm32-quadspi: explicitly request exclusive reset control net: dsa: mt7530: explicitly request exclusive reset control net: ethernet: hisi_femac: explicitly request exclusive reset control net: ethernet: hix5hd2_gmac: explicitly request exclusive reset control net: stmmac: explicitly request exclusive reset control net: stmmac: dwc-qos: explicitly request exclusive reset control ath10k: explicitly request exclusive reset control nvmem: lpc18xx-eeprom: explicitly request exclusive reset control PCI: dwc: pcie-qcom: explicitly request exclusive reset control PCI: imx6: explicitly request exclusive reset control PCI: tegra: explicitly request exclusive reset control PCI: rockchip: explicitly request exclusive reset control phy: berlin-usb: explicitly request exclusive reset control PCI: mediatek: explicitly request exclusive reset control phy: qcom-usb-hs: explicitly request exclusive reset control phy: rockchip-pcie: explicitly request exclusive reset control phy: rockchip-typec: explicitly request exclusive reset control phy: rockchip-usb: explicitly request exclusive reset control phy: sun4i-usb: explicitly request exclusive reset control phy: sun9i-usb: explicitly request exclusive reset control phy: tegra: explicitly request exclusive reset control phy: qcom-qmp: explicitly request exclusive reset control phy: qcom-qusb2: explicitly request exclusive reset control pinctrl: stm32: explicitly request exclusive reset control pinctrl: sunxi: explicitly request exclusive reset control pinctrl: tegra: explicitly request exclusive reset control pwm: hibvt: explicitly request exclusive reset control pwm: tegra: explicitly request exclusive reset control remoteproc/keystone: explicitly request exclusive reset control remoteproc: qcom: explicitly request exclusive reset control remoteproc: st: explicitly request exclusive reset control soc: mediatek: PMIC wrap: explicitly request exclusive reset control soc/tegra: pmc: explicitly request exclusive reset control spi: stm32: explicitly request exclusive reset control spi: sun6i: explicitly request exclusive reset control spi: tegra20-slink: explicitly request exclusive reset control spi: tegra114: explicitly request exclusive reset control spi: tegra20-sflash: explicitly request exclusive reset control staging: nvec: explicitly request exclusive reset control thermal: rockchip: explicitly request exclusive reset control thermal: tegra: explicitly request exclusive reset control serial: 8250_dw: explicitly request exclusive reset control serial: tegra: explicitly request exclusive reset control usb: chipidea: msm: explicitly request exclusive reset control usb: dwc2: explicitly request exclusive reset control usb: host: ehci-tegra: explicitly request exclusive reset control usb: host: xhci-tegra: explicitly request exclusive reset control usb: musb: sunxi: explicitly request exclusive reset control usb: phy: msm: explicitly request exclusive reset control usb: phy: qcom-8x16-usb: explicitly request exclusive reset control watchdog: asm9260: explicitly request exclusive reset control watchdog: mt7621: explicitly request exclusive reset control watchdog: rt2880: explicitly request exclusive reset control watchdog: zx2967: explicitly request exclusive reset control ASoC: img: explicitly request exclusive reset control ASoC: stm32: explicitly request exclusive reset control ASoC: sun4i: explicitly request exclusive reset control ASoC: tegra: explicitly request exclusive reset control Documentation: devres: add explicit exclusive/shared reset control request calls reset: finish transition to explicit exclusive reset control requests Documentation/driver-model/devres.txt | 7 ++- arch/arm/mach-rockchip/platsmp.c | 2 +- arch/mips/pci/pci-mt7620.c | 2 +- drivers/ata/ahci_st.c | 6 +-- drivers/ata/ahci_tegra.c | 8 ++-- drivers/ata/sata_gemini.c | 4 +- drivers/bus/sunxi-rsb.c | 2 +- drivers/bus/tegra-gmi.c | 2 +- drivers/clk/sunxi/clk-sun9i-mmc.c | 2 +- drivers/clk/tegra/clk-dfll.c | 2 +- drivers/clocksource/timer-stm32.c | 2 +- drivers/clocksource/timer-sun5i.c | 2 +- drivers/crypto/rockchip/rk3288_crypto.c | 2 +- drivers/crypto/sunxi-ss/sun4i-ss-core.c | 3 +- drivers/devfreq/tegra-devfreq.c | 2 +- drivers/dma/stm32-dma.c | 2 +- drivers/dma/sun6i-dma.c | 2 +- drivers/dma/tegra20-apb-dma.c | 2 +- drivers/fpga/altera-hps2fpga.c | 3 +- drivers/gpu/drm/hisilicon/kirin/kirin_drm_ade.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/device/tegra.c | 2 +- drivers/gpu/drm/rockchip/analogix_dp-rockchip.c | 2 +- drivers/gpu/drm/rockchip/cdn-dp-core.c | 8 ++-- drivers/gpu/drm/rockchip/dw-mipi-dsi.c | 2 +- drivers/gpu/drm/rockchip/rockchip_drm_vop.c | 4 +- drivers/gpu/drm/sti/sti_hdmi.c | 2 +- drivers/gpu/drm/sti/sti_hqvdp.c | 2 +- drivers/gpu/drm/sti/sti_tvout.c | 2 +- drivers/gpu/drm/stm/ltdc.c | 2 +- drivers/gpu/drm/sun4i/sun4i_backend.c | 4 +- drivers/gpu/drm/sun4i/sun4i_tcon.c | 2 +- drivers/gpu/drm/sun4i/sun4i_tv.c | 2 +- drivers/gpu/drm/sun4i/sun6i_drc.c | 2 +- drivers/gpu/drm/sun4i/sun8i_mixer.c | 2 +- drivers/gpu/drm/tegra/dc.c | 2 +- drivers/gpu/drm/tegra/dpaux.c | 3 +- drivers/gpu/drm/tegra/dsi.c | 2 +- drivers/gpu/drm/tegra/gr3d.c | 6 +-- drivers/gpu/drm/tegra/hdmi.c | 2 +- drivers/gpu/drm/tegra/sor.c | 2 +- drivers/gpu/host1x/dev.c | 2 +- drivers/i2c/busses/i2c-mv64xxx.c | 2 +- drivers/i2c/busses/i2c-stm32f4.c | 2 +- drivers/i2c/busses/i2c-sun6i-p2wi.c | 2 +- drivers/i2c/busses/i2c-tegra.c | 2 +- drivers/iio/adc/rockchip_saradc.c | 3 +- drivers/iio/dac/stm32-dac-core.c | 2 +- drivers/input/keyboard/tegra-kbc.c | 2 +- drivers/media/platform/coda/coda-common.c | 3 +- drivers/media/platform/stm32/stm32-dcmi.c | 2 +- drivers/media/rc/st_rc.c | 2 +- drivers/media/rc/sunxi-cir.c | 2 +- drivers/mmc/host/dw_mmc.c | 2 +- drivers/mmc/host/sdhci-st.c | 2 +- drivers/mmc/host/sdhci-tegra.c | 3 +- drivers/mmc/host/sunxi-mmc.c | 3 +- drivers/mtd/nand/sunxi_nand.c | 2 +- drivers/mtd/spi-nor/stm32-quadspi.c | 2 +- drivers/net/dsa/mt7530.c | 3 +- drivers/net/ethernet/hisilicon/hisi_femac.c | 4 +- drivers/net/ethernet/hisilicon/hix5hd2_gmac.c | 6 +-- .../ethernet/stmicro/stmmac/dwmac-dwc-qos-eth.c | 2 +- drivers/net/ethernet/stmicro/stmmac/dwmac-sun8i.c | 3 +- .../net/ethernet/stmicro/stmmac/stmmac_platform.c | 4 +- drivers/net/wireless/ath/ath10k/ahb.c | 15 ++++--- drivers/nvmem/lpc18xx_eeprom.c | 2 +- drivers/pci/dwc/pci-imx6.c | 7 +-- drivers/pci/dwc/pcie-qcom.c | 40 +++++++++-------- drivers/pci/host/pci-tegra.c | 6 +-- drivers/pci/host/pcie-mediatek.c | 2 +- drivers/pci/host/pcie-rockchip.c | 15 ++++--- drivers/phy/allwinner/phy-sun4i-usb.c | 2 +- drivers/phy/allwinner/phy-sun9i-usb.c | 4 +- drivers/phy/marvell/phy-berlin-usb.c | 2 +- drivers/phy/qualcomm/phy-qcom-qmp.c | 4 +- drivers/phy/qualcomm/phy-qcom-qusb2.c | 3 +- drivers/phy/qualcomm/phy-qcom-usb-hs.c | 3 +- drivers/phy/rockchip/phy-rockchip-pcie.c | 2 +- drivers/phy/rockchip/phy-rockchip-typec.c | 6 +-- drivers/phy/rockchip/phy-rockchip-usb.c | 2 +- drivers/phy/tegra/xusb-tegra210.c | 4 +- drivers/phy/tegra/xusb.c | 2 +- drivers/pinctrl/stm32/pinctrl-stm32.c | 2 +- drivers/pinctrl/sunxi/pinctrl-sun6i-a31-r.c | 2 +- drivers/pinctrl/sunxi/pinctrl-sun8i-a23-r.c | 2 +- drivers/pinctrl/tegra/pinctrl-tegra-xusb.c | 2 +- drivers/pwm/pwm-hibvt.c | 2 +- drivers/pwm/pwm-tegra.c | 2 +- drivers/remoteproc/keystone_remoteproc.c | 2 +- drivers/remoteproc/qcom_q6v5_pil.c | 3 +- drivers/remoteproc/st_remoteproc.c | 6 ++- drivers/reset/core.c | 2 +- drivers/soc/mediatek/mtk-pmic-wrap.c | 5 ++- drivers/soc/tegra/pmc.c | 2 +- drivers/spi/spi-stm32.c | 2 +- drivers/spi/spi-sun6i.c | 2 +- drivers/spi/spi-tegra114.c | 2 +- drivers/spi/spi-tegra20-sflash.c | 2 +- drivers/spi/spi-tegra20-slink.c | 2 +- drivers/staging/nvec/nvec.c | 2 +- drivers/thermal/rockchip_thermal.c | 3 +- drivers/thermal/tegra/soctherm.c | 3 +- drivers/tty/serial/8250/8250_dw.c | 2 +- drivers/tty/serial/serial-tegra.c | 2 +- drivers/usb/chipidea/ci_hdrc_msm.c | 2 +- drivers/usb/dwc2/platform.c | 3 +- drivers/usb/host/ehci-tegra.c | 5 ++- drivers/usb/host/xhci-tegra.c | 6 ++- drivers/usb/musb/sunxi.c | 2 +- drivers/usb/phy/phy-msm-usb.c | 4 +- drivers/usb/phy/phy-qcom-8x16-usb.c | 2 +- drivers/watchdog/asm9260_wdt.c | 2 +- drivers/watchdog/mt7621_wdt.c | 2 +- drivers/watchdog/rt2880_wdt.c | 2 +- drivers/watchdog/zx2967_wdt.c | 2 +- include/linux/reset.h | 50 ---------------------- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/img/img-parallel-out.c | 2 +- sound/soc/img/img-spdif-in.c | 2 +- sound/soc/img/img-spdif-out.c | 2 +- sound/soc/stm/stm32_i2s.c | 2 +- sound/soc/stm/stm32_sai.c | 2 +- sound/soc/stm/stm32_spdifrx.c | 2 +- sound/soc/sunxi/sun4i-codec.c | 3 +- sound/soc/sunxi/sun4i-i2s.c | 2 +- sound/soc/sunxi/sun4i-spdif.c | 3 +- sound/soc/tegra/tegra30_ahub.c | 4 +- 128 files changed, 226 insertions(+), 235 deletions(-) -- 2.11.0 Cc: "David S. Miller" <davem(a)davemloft.net> Cc: "Emilio López" <emilio(a)elopez.com.ar> Cc: Adrian Hunter <adrian.hunter(a)intel.com> Cc: Alan Stern <stern(a)rowland.harvard.edu> Cc: Alan Tull <atull(a)kernel.org> Cc: Alexandre Torgue <alexandre.torgue(a)st.com> Cc: Andrew Lunn <andrew(a)lunn.ch> Cc: Ben Skeggs <bskeggs(a)redhat.com> Cc: Benjamin Gaignard <benjamin.gaignard(a)linaro.org> Cc: Bin Liu <b-liu(a)ti.com> Cc: Bjorn Andersson <bjorn.andersson(a)linaro.org> Cc: Bjorn Helgaas <bhelgaas(a)google.com> Cc: Boris Brezillon <boris.brezillon(a)free-electrons.com> Cc: Brian Norris <computersforpeace(a)gmail.com> Cc: Chanwoo Choi <cw00.choi(a)samsung.com> Cc: Chen Feng <puck.chen(a)hisilicon.com> Cc: Chen-Yu Tsai <wens(a)csie.org> Cc: Corentin Labbe <clabbe.montjoie(a)gmail.com> Cc: Cyrille Pitchen <cyrille.pitchen(a)wedev4u.fr> Cc: Dan Williams <dan.j.williams(a)intel.com> Cc: Daniel Lezcano <daniel.lezcano(a)linaro.org> Cc: David Airlie <airlied(a)linux.ie> Cc: David Woodhouse <dwmw2(a)infradead.org> Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: Eduardo Valentin <edubezval(a)gmail.com> Cc: Felipe Balbi <balbi(a)kernel.org> Cc: Florian Fainelli <f.fainelli(a)gmail.com> Cc: Giuseppe Cavallaro <peppe.cavallaro(a)st.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Guenter Roeck <linux(a)roeck-us.net> Cc: Hartmut Knaack <knaack.h(a)gmx.de> Cc: Heiko Stuebner <heiko(a)sntech.de> Cc: Herbert Xu <herbert(a)gondor.apana.org.au> Cc: Jaehoon Chung <jh80.chung(a)samsung.com> Cc: Jiri Slaby <jslaby(a)suse.com> Cc: Joachim Eastwood <manabian(a)gmail.com> Cc: John Youn <johnyoun(a)synopsys.com> Cc: Jon Hunter <jonathanh(a)nvidia.com> Cc: Jonathan Cameron <jic23(a)kernel.org> Cc: Jonathan Corbet <corbet(a)lwn.net> Cc: Jonathan Hunter <jonathanh(a)nvidia.com> Cc: Kalle Valo <kvalo(a)qca.qualcomm.com> Cc: Kishon Vijay Abraham I <kishon(a)ti.com> Cc: Kyungmin Park <kyungmin.park(a)samsung.com> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Laxman Dewangan <ldewangan(a)nvidia.com> Cc: Lee Jones <lee.jones(a)linaro.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Linus Walleij <linus.walleij(a)linaro.org> Cc: Lucas Stach <l.stach(a)pengutronix.de> Cc: Marc Dietrich <marvin24(a)gmx.de> Cc: Marek Vasut <marek.vasut(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Mark Yao <mark.yao(a)rock-chips.com> Cc: Mathias Nyman <mathias.nyman(a)intel.com> Cc: Matthias Brugger <matthias.bgg(a)gmail.com> Cc: Maxime Coquelin <mcoquelin.stm32(a)gmail.com> Cc: Maxime Ripard <maxime.ripard(a)free-electrons.com> Cc: Michael Turquette <mturquette(a)baylibre.com> Cc: Moritz Fischer <moritz.fischer(a)ettus.com> Cc: MyungJoo Ham <myungjoo.ham(a)samsung.com> Cc: Ohad Ben-Cohen <ohad(a)wizery.com> Cc: Patrice Chotard <patrice.chotard(a)st.com> Cc: Peter Chen <Peter.Chen(a)nxp.com> Cc: Peter De Schrijver <pdeschrijver(a)nvidia.com> Cc: Peter Meerwald-Stadler <pmeerw(a)pmeerw.net> Cc: Philippe Cornu <philippe.cornu(a)st.com> Cc: Prashant Gaikwad <pgaikwad(a)nvidia.com> Cc: Rakesh Iyer <riyer(a)nvidia.com> Cc: Ralf Baechle <ralf(a)linux-mips.org> Cc: Richard Weinberger <richard(a)nod.at> Cc: Richard Zhu <hongxing.zhu(a)nxp.com> Cc: Rongrong Zou <zourongrong(a)gmail.com> Cc: Ryder Lee <ryder.lee(a)mediatek.com> Cc: Salil Mehta <salil.mehta(a)huawei.com> Cc: Shawn Lin <shawn.lin(a)rock-chips.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Stanimir Varbanov <svarbanov(a)mm-sol.com> Cc: Stephen Boyd <sboyd(a)codeaurora.org> Cc: Tejun Heo <tj(a)kernel.org> Cc: Thierry Reding <thierry.reding(a)gmail.com> Cc: Thomas Gleixner <tglx(a)linutronix.de> Cc: Ulf Hansson <ulf.hansson(a)linaro.org> Cc: Vincent Abriou <vincent.abriou(a)st.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Vivien Didelot <vivien.didelot(a)savoirfairelinux.com> Cc: Wim Van Sebroeck <wim(a)iguana.be> Cc: Wolfram Sang <wsa(a)the-dreams.de> Cc: Xinliang Liu <z.liuxinliang(a)hisilicon.com> Cc: Xinwei Kong <kong.kongxinwei(a)hisilicon.com> Cc: Yannick Fertre <yannick.fertre(a)st.com> Cc: Yisen Zhuang <yisen.zhuang(a)huawei.com> Cc: Zhang Rui <rui.zhang(a)intel.com> Cc: alsa-devel(a)alsa-project.org Cc: ath10k(a)lists.infradead.org Cc: devel(a)driverdev.osuosl.org Cc: dmaengine(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-arm-msm(a)vger.kernel.org Cc: linux-clk(a)vger.kernel.org Cc: linux-crypto(a)vger.kernel.org Cc: linux-doc(a)vger.kernel.org Cc: linux-fpga(a)vger.kernel.org Cc: linux-gpio(a)vger.kernel.org Cc: linux-i2c(a)vger.kernel.org Cc: linux-ide(a)vger.kernel.org Cc: linux-iio(a)vger.kernel.org Cc: linux-input(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: linux-media(a)vger.kernel.org Cc: linux-mediatek(a)lists.infradead.org Cc: linux-mips(a)linux-mips.org Cc: linux-mmc(a)vger.kernel.org Cc: linux-mtd(a)lists.infradead.org Cc: linux-pci(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linux-pwm(a)vger.kernel.org Cc: linux-remoteproc(a)vger.kernel.org Cc: linux-rockchip(a)lists.infradead.org Cc: linux-serial(a)vger.kernel.org Cc: linux-spi(a)vger.kernel.org Cc: linux-tegra(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: linux-watchdog(a)vger.kernel.org Cc: linux-wireless(a)vger.kernel.org Cc: netdev(a)vger.kernel.org Cc: nouveau(a)lists.freedesktop.org

9 21

[alsa-devel] [PATCH 00/10] constify ASoC snd_pcm_ops structures
by Arvind Yadav 14 Aug '17

14 Aug '17

snd_pcm_ops are not supposed to change at runtime. All functions working with snd_pcm_ops provided by <sound/soc.h> work with const snd_pcm_ops. So mark the non-const structs as const. Arvind Yadav (10): [PATCH 01/10] ASoC: au1x: constify snd_pcm_ops structures [PATCH 02/10] ASoC: blackfin: constify snd_pcm_ops structures [PATCH 03/10] ASoC: fsl: constify snd_pcm_ops structures [PATCH 04/10] ASoC: intel: constify snd_pcm_ops structures [PATCH 05/10] ASoC: nuc900: constify snd_pcm_ops structures [PATCH 06/10] ASoC: omap: constify snd_pcm_ops structures [PATCH 07/10] ASoC: pxa: constify snd_pcm_ops structures [PATCH 08/10] ASoC: samsung: constify snd_pcm_ops structures [PATCH 09/10] ASoC: sh: constify snd_pcm_ops structures [PATCH 10/10] ASoC: txx9: constify snd_pcm_ops structures sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 2 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 2 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/imx-pcm-fiq.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 2 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 2 +- sound/soc/nuc900/nuc900-pcm.c | 2 +- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 2 +- sound/soc/samsung/idma.c | 2 +- sound/soc/sh/dma-sh7760.c | 2 +- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/siu_pcm.c | 2 +- sound/soc/txx9/txx9aclc.c | 2 +- 18 files changed, 18 insertions(+), 18 deletions(-) -- 1.9.1

1 10

[alsa-devel] [PATCH] ASoC: rt5663: Fine tune for the headphone output pop sound
by Oder Chiou 14 Aug '17

14 Aug '17

The patch fine tunes for the headphone output pop sound. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 21 ++++++++++++++++----- 1 file changed, 16 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 8119819..c98a8a7 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1565,6 +1565,10 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) val = snd_soc_read(codec, RT5663_EM_JACK_TYPE_2) & 0x0003; dev_dbg(codec->dev, "%s val = %d\n", __func__, val); + snd_soc_update_bits(codec, RT5663_HP_CHARGE_PUMP_1, + RT5663_OSW_HP_L_MASK | RT5663_OSW_HP_R_MASK, + RT5663_OSW_HP_L_EN | RT5663_OSW_HP_R_EN); + switch (val) { case 1: case 2: @@ -2073,6 +2077,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, snd_soc_write(codec, RT5663_ANA_BIAS_CUR_1, 0x7766); snd_soc_write(codec, RT5663_HP_BIAS, 0xafaa); snd_soc_write(codec, RT5663_CHARGE_PUMP_2, 0x7777); + snd_soc_update_bits(codec, RT5663_STO_DRE_1, 0x8000, + 0x8000); snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x3000, 0x3000); } @@ -2105,14 +2111,19 @@ static int rt5663_charge_pump_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - if (rt5663->codec_ver == CODEC_VER_0) - snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x003b, - 0x003b); + if (rt5663->codec_ver == CODEC_VER_0) { + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030, + 0x0030); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0003, + 0x0003); + } break; case SND_SOC_DAPM_POST_PMD: - if (rt5663->codec_ver == CODEC_VER_0) - snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x003b, 0); + if (rt5663->codec_ver == CODEC_VER_0) { + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0003, 0); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030, 0); + } break; default: -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: samsung: make tm2_dapm_widgets and tm2_pm_ops static
by Colin King 13 Aug '17

13 Aug '17

From: Colin Ian King <colin.king(a)canonical.com> tm2_dapm_widgets and tm2_pm_ops are local to the source and do not need to be in global scope, so make them static. Cleans up sparse warnings: symbol 'tm2_dapm_widgets' was not declared. Should it be static? symbol 'tm2_pm_ops' was not declared. Should it be static? Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/samsung/tm2_wm5110.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 24cc9d63ce87..68698f3d72f9 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -318,7 +318,7 @@ static const struct snd_kcontrol_new tm2_controls[] = { SOC_DAPM_PIN_SWITCH("Headset Mic"), }; -const struct snd_soc_dapm_widget tm2_dapm_widgets[] = { +static const struct snd_soc_dapm_widget tm2_dapm_widgets[] = { SND_SOC_DAPM_HP("HP", NULL), SND_SOC_DAPM_SPK("SPK", NULL), SND_SOC_DAPM_SPK("RCV", NULL), @@ -521,7 +521,7 @@ static void tm2_pm_complete(struct device *dev) tm2_start_sysclk(card); } -const struct dev_pm_ops tm2_pm_ops = { +static const struct dev_pm_ops tm2_pm_ops = { .prepare = tm2_pm_prepare, .suspend = snd_soc_suspend, .resume = snd_soc_resume, -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: zx_aud96p22: make array aud96p22_dt_ids static
by Colin King 13 Aug '17

13 Aug '17

From: Colin Ian King <colin.king(a)canonical.com> The array aud96p22_dt_ids is local to the source and does not need to be in global scope, so make it static. Cleans up sparse warning: symbol 'aud96p22_dt_ids' was not declared. Should it be static? Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/zx_aud96p22.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/zx_aud96p22.c b/sound/soc/codecs/zx_aud96p22.c index 36d0a7087a54..78017fb69c04 100644 --- a/sound/soc/codecs/zx_aud96p22.c +++ b/sound/soc/codecs/zx_aud96p22.c @@ -382,7 +382,7 @@ static int aud96p22_i2c_remove(struct i2c_client *i2c) return 0; } -const struct of_device_id aud96p22_dt_ids[] = { +static const struct of_device_id aud96p22_dt_ids[] = { { .compatible = "zte,zx-aud96p22", }, { } }; -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: rt5514: make array rt5514_dai static
by Colin King 13 Aug '17

13 Aug '17

From: Colin Ian King <colin.king(a)canonical.com> The array rt5514_dai is local to the source and does not need to be in global scope, so make it static. Cleans up sparse warnings: symbol 'rt5514_dai' was not declared. Should it be static? warning: symbol 'rt5514_dai' was not declared. Should it be static? Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5514.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 7afcb57989af..d7956ababd11 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -1066,7 +1066,7 @@ static const struct snd_soc_dai_ops rt5514_aif_dai_ops = { .set_tdm_slot = rt5514_set_tdm_slot, }; -struct snd_soc_dai_driver rt5514_dai[] = { +static struct snd_soc_dai_driver rt5514_dai[] = { { .name = "rt5514-aif1", .id = 0, -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: max98926: make max98926_spk_tlv and max98926_current_tlv static
by Colin King 13 Aug '17

13 Aug '17

From: Colin Ian King <colin.king(a)canonical.com> max98926_spk_tlv and max98926_current_tlv are local to the source and do not need to be in global scope, so make them static. Cleans up sparse warnings: symbol 'max98926_spk_tlv' was not declared. Should it be static? symbol 'max98926_current_tlv' was not declared. Should it be static? Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/max98926.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c index 59d03d036945..03d07bf4d942 100644 --- a/sound/soc/codecs/max98926.c +++ b/sound/soc/codecs/max98926.c @@ -213,8 +213,8 @@ static bool max98926_readable_register(struct device *dev, unsigned int reg) } }; -DECLARE_TLV_DB_SCALE(max98926_spk_tlv, -600, 100, 0); -DECLARE_TLV_DB_RANGE(max98926_current_tlv, +static DECLARE_TLV_DB_SCALE(max98926_spk_tlv, -600, 100, 0); +static DECLARE_TLV_DB_RANGE(max98926_current_tlv, 0, 11, TLV_DB_SCALE_ITEM(20, 20, 0), 12, 15, TLV_DB_SCALE_ITEM(320, 40, 0), ); -- 2.11.0

1 0

[alsa-devel] [PATCH 0/3] ALSA: mpu401: Adjustments for three function implementations
by SF Markus Elfring 13 Aug '17

13 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 12 Aug 2017 21:05:21 +0200 A few update suggestions were taken into account from static source code analysis. Markus Elfring (3): Delete an error message for a failed memory allocation in snd_mpu401_uart_new() Use common error handling code in snd_mpu401_uart_new() Adjust four checks for null pointers sound/drivers/mpu401/mpu401_uart.c | 24 +++++++++++++----------- 1 file changed, 13 insertions(+), 11 deletions(-) -- 2.14.0

3 5

[alsa-devel] [PATCH] ASoC: hdmi-codec: make a function and two arrays static
by Colin King 13 Aug '17

13 Aug '17

From: Colin Ian King <colin.king(a)canonical.com> The function hdmi_codec_eld_chmap and arrays hdmi_codec_stereo_chmaps and hdmi_codec_8ch_chmaps are local to the source and do not need to be in global scope, so make them static. Cleans up sparse warnings: symbol 'hdmi_codec_eld_chmap' was not declared. Should it be static? symbol 'hdmi_codec_stereo_chmaps' was not declared. Should it be static? symbol 'hdmi_codec_8ch_chmaps' was not declared. Should it be static? Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/hdmi-codec.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index f288404f0a61..7aba4f312809 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -67,14 +67,14 @@ struct hdmi_codec_cea_spk_alloc { }; /* Channel maps stereo HDMI */ -const struct snd_pcm_chmap_elem hdmi_codec_stereo_chmaps[] = { +static const struct snd_pcm_chmap_elem hdmi_codec_stereo_chmaps[] = { { .channels = 2, .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, { } }; /* Channel maps for multi-channel playbacks, up to 8 n_ch */ -const struct snd_pcm_chmap_elem hdmi_codec_8ch_chmaps[] = { +static const struct snd_pcm_chmap_elem hdmi_codec_8ch_chmaps[] = { { .channels = 2, /* CA_ID 0x00 */ .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, { .channels = 4, /* CA_ID 0x01 */ @@ -340,7 +340,7 @@ static unsigned long hdmi_codec_spk_mask_from_alloc(int spk_alloc) return spk_mask; } -void hdmi_codec_eld_chmap(struct hdmi_codec_priv *hcp) +static void hdmi_codec_eld_chmap(struct hdmi_codec_priv *hcp) { u8 spk_alloc; unsigned long spk_mask; -- 2.11.0

1 0

[alsa-devel] [PATCH 0/9] constify isa snd_pcm_ops structures
by Arvind Yadav 13 Aug '17

13 Aug '17

snd_pcm_ops are not supposed to change at runtime. All functions working with snd_pcm_ops provided by <sound/pcm.h> work with const snd_pcm_ops. So mark the non-const structs as const. Arvind Yadav (9): [PATCH 1/9] ALSA: ad1816a: constify snd_pcm_ops structures [PATCH 2/9] ALSA: es1688: constify snd_pcm_ops structures [PATCH 3/9] ALSA: es18xx: constify snd_pcm_ops structures [PATCH 4/9] ALSA: gus: constify snd_pcm_ops structures [PATCH 5/9] ALSA: msnd: constify snd_pcm_ops structures [PATCH 6/9] ALSA: emu8000: constify snd_pcm_ops structures [PATCH 7/9] ALSA: sb16: constify snd_pcm_ops structures [PATCH 8/9] ALSA: sb8: constify snd_pcm_ops structures [PATCH 9/9] ALSA: wss: constify snd_pcm_ops structures sound/isa/ad1816a/ad1816a_lib.c | 4 ++-- sound/isa/es1688/es1688_lib.c | 4 ++-- sound/isa/es18xx.c | 4 ++-- sound/isa/gus/gus_pcm.c | 4 ++-- sound/isa/msnd/msnd.c | 4 ++-- sound/isa/sb/emu8000_pcm.c | 2 +- sound/isa/sb/sb16_main.c | 4 ++-- sound/isa/sb/sb8_main.c | 4 ++-- sound/isa/wss/wss_lib.c | 4 ++-- 9 files changed, 17 insertions(+), 17 deletions(-) -- 1.9.1

2 10

[alsa-devel] [PATCH alsa-lib] src/rawmidi/rawmidi_symbols.c: use rawmidi_virt only when available
by Thomas Petazzoni 13 Aug '17

13 Aug '17

src/rawmidi/Makefile.am only brings rawmidi_virt.c into the build when BUILD_SEQ is defined (i.e when --enable-seq is passed). However, rawmidi_symbols.c unconditionally refers to _snd_module_rawmidi_virt, defined in rawmidi_virt.c. This causes a link failure when BUILD_SEQ is disabled. For example when linking ffmpeg against alsa-lib: /home/thomas/projets/buildroot/output/host/arm-buildroot-linux-uclibcgnueabi/sysroot/usr/lib/libasound.a(pcm_dmix.o): In function `snd_pcm_dmix_sync_ptr': pcm_dmix.c:(.text+0x83c): warning: /home/thomas/projets/buildroot/output/host/arm-buildroot-linux-uclibcgnueabi/sysroot/usr/lib/libasound.a(rawmidi_symbols.o):(.data+0x4): undefined reference to `_snd_module_rawmidi_virt' collect2: error: ld returned 1 exit status To fix this, we make sure that rawmidi_symbols.c only uses _snd_module_rawmidi_virt when available. Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)free-electrons.com> --- src/rawmidi/rawmidi_symbols.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/src/rawmidi/rawmidi_symbols.c b/src/rawmidi/rawmidi_symbols.c index cdc06d7..6473433 100644 --- a/src/rawmidi/rawmidi_symbols.c +++ b/src/rawmidi/rawmidi_symbols.c @@ -21,11 +21,15 @@ #ifndef PIC extern const char *_snd_module_rawmidi_hw; +#ifdef BUILD_SEQ extern const char *_snd_module_rawmidi_virt; +#endif static const char **snd_rawmidi_open_objects[] = { &_snd_module_rawmidi_hw, +#ifdef BUILD_SEQ &_snd_module_rawmidi_virt +#endif }; void *snd_rawmidi_open_symbols(void) -- 2.9.4

2 1

[alsa-devel] [PATCH] ALSA: opl3: Delete an error message for a failed memory allocation in snd_opl3_new()
by SF Markus Elfring 13 Aug '17

13 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 12 Aug 2017 19:54:20 +0200 Omit an extra message for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/drivers/opl3/opl3_lib.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index d5e5b4657b4b..588963d6be28 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -355,10 +355,8 @@ int snd_opl3_new(struct snd_card *card, *ropl3 = NULL; opl3 = kzalloc(sizeof(*opl3), GFP_KERNEL); - if (opl3 == NULL) { - snd_printk(KERN_ERR "opl3: cannot allocate\n"); + if (!opl3) return -ENOMEM; - } opl3->card = card; opl3->hardware = hardware; -- 2.14.0

2 1

[alsa-devel] [PATCH] ALSA: ca0106: Delete an error message for a failed memory allocation in snd_ca0106_pcm_open_capture_channel()
by SF Markus Elfring 13 Aug '17

13 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 12 Aug 2017 19:09:23 +0200 Omit an extra message for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/ca0106/ca0106_main.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 6165a57a94ae..b7783137af04 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -660,11 +660,9 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre int err; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); - if (epcm == NULL) { - dev_err(chip->card->dev, - "open_capture_channel: failed epcm alloc\n"); + if (!epcm) return -ENOMEM; - } + epcm->emu = chip; epcm->substream = substream; epcm->channel_id=channel_id; -- 2.14.0

2 1

[alsa-devel] [PATCH] ALSA: mixart: Delete an error message for a failed memory allocation in snd_mixart_create()
by SF Markus Elfring 13 Aug '17

13 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 12 Aug 2017 18:45:42 +0200 Omit an extra message for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/mixart/mixart.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 6d7fbf30618b..84f6b2651bf3 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1052,10 +1052,8 @@ static int snd_mixart_create(struct mixart_mgr *mgr, struct snd_card *card, int }; chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (! chip) { - dev_err(card->dev, "cannot allocate chip\n"); + if (!chip) return -ENOMEM; - } chip->card = card; chip->chip_idx = idx; -- 2.14.0

2 1

[alsa-devel] [PATCH] ALSA: pcxhr: Delete an error message for a failed memory allocation in pcxhr_create()
by SF Markus Elfring 13 Aug '17

13 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 12 Aug 2017 18:14:54 +0200 Omit an extra message for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/pcxhr/pcxhr.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index fa919f52e225..376fd76caec2 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1215,10 +1215,8 @@ static int pcxhr_create(struct pcxhr_mgr *mgr, }; chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (! chip) { - dev_err(card->dev, "cannot allocate chip\n"); + if (!chip) return -ENOMEM; - } chip->card = card; chip->chip_idx = idx; -- 2.14.0

2 1

[alsa-devel] [PATCH 00/16] ALSA: PCI: constify snd_pcm_hardware const
by Bhumika Goyal 13 Aug '17

13 Aug '17

Declare snd_pcm_hardware const. Done using Coccinelle. Bhumika Goyal (16): ALSA: aw2: make snd_pcm_hardware const ALSA: ca0106: make snd_pcm_hardware const ALSA: cs46xx: make snd_pcm_hardware const ALSA: korg1212: make snd_pcm_hardware const ALSA: cs5535audio: make snd_pcm_hardware const ALSA: emu10k1: make snd_pcm_hardware const ALSA: lola: make snd_pcm_hardware const ALSA: lx6464es: make snd_pcm_hardware const ALSA: mixart: make snd_pcm_hardware const ALSA: ctxfi: make snd_pcm_hardware const ALSA: pcxhr: make snd_pcm_hardware const ALSA: riptide: make snd_pcm_hardware const ALSA: rme9652: make snd_pcm_hardware const ALSA: trident: make snd_pcm_hardware const ALSA: ymfpci: make snd_pcm_hardware const ALSA: pci: make snd_pcm_hardware const sound/pci/ad1889.c | 4 ++-- sound/pci/als300.c | 4 ++-- sound/pci/als4000.c | 4 ++-- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/aw2/aw2-alsa.c | 4 ++-- sound/pci/bt87x.c | 4 ++-- sound/pci/ca0106/ca0106_main.c | 4 ++-- sound/pci/cmipci.c | 12 ++++++------ sound/pci/cs4281.c | 4 ++-- sound/pci/cs46xx/cs46xx_lib.c | 4 ++-- sound/pci/cs5535audio/cs5535audio_pcm.c | 4 ++-- sound/pci/ctxfi/ctpcm.c | 6 +++--- sound/pci/emu10k1/emu10k1x.c | 4 ++-- sound/pci/emu10k1/emupcm.c | 10 +++++----- sound/pci/emu10k1/p16v.c | 4 ++-- sound/pci/ens1370.c | 6 +++--- sound/pci/es1938.c | 4 ++-- sound/pci/es1968.c | 4 ++-- sound/pci/fm801.c | 4 ++-- sound/pci/intel8x0.c | 2 +- sound/pci/intel8x0m.c | 2 +- sound/pci/korg1212/korg1212.c | 4 ++-- sound/pci/lola/lola_pcm.c | 2 +- sound/pci/lx6464es/lx6464es.c | 2 +- sound/pci/maestro3.c | 4 ++-- sound/pci/mixart/mixart.c | 4 ++-- sound/pci/pcxhr/pcxhr.c | 2 +- sound/pci/riptide/riptide.c | 4 ++-- sound/pci/rme32.c | 8 ++++---- sound/pci/rme96.c | 8 ++++---- sound/pci/rme9652/hdsp.c | 4 ++-- sound/pci/rme9652/rme9652.c | 4 ++-- sound/pci/sis7019.c | 4 ++-- sound/pci/sonicvibes.c | 4 ++-- sound/pci/trident/trident_main.c | 10 +++++----- sound/pci/via82xx.c | 2 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/ymfpci/ymfpci_main.c | 4 ++-- 39 files changed, 85 insertions(+), 85 deletions(-) -- 1.9.1

2 17

[alsa-devel] [PATCH 0/3] ALSA: rme9652: Adjustments for six function implementations
by SF Markus Elfring 13 Aug '17

13 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 12 Aug 2017 17:17:23 +0200 A few update suggestions were taken into account from static source code analysis. Markus Elfring (3): Delete an error message for a failed memory allocation in snd_hdspm_create() Improve eight size determinations Adjust seven checks for null pointers sound/pci/rme9652/hdspm.c | 42 ++++++++++++++++++------------------------ 1 file changed, 18 insertions(+), 24 deletions(-) -- 2.14.0

2 4

[alsa-devel] [PATCH 0/3] ALSA: rme96: Adjustments for six function implementations
by SF Markus Elfring 13 Aug '17

13 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 12 Aug 2017 15:36:54 +0200 A few update suggestions were taken into account from static source code analysis. Markus Elfring (3): Delete two error messages for a failed memory allocation in snd_rme96_probe() Use common error handling code in snd_rme96_probe() Adjust five checks for null pointers sound/pci/rme96.c | 43 ++++++++++++++++++++----------------------- 1 file changed, 20 insertions(+), 23 deletions(-) -- 2.14.0

2 4

[alsa-devel] [PATCH] ALSA: trident: Delete an error message for a failed memory allocation in snd_trident_tlb_alloc()
by SF Markus Elfring 13 Aug '17

13 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 12 Aug 2017 13:20:16 +0200 Omit an extra message for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/trident/trident_main.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 64d3b8eba4bb..0eca3c370990 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3363,11 +3363,9 @@ static int snd_trident_tlb_alloc(struct snd_trident *trident) trident->tlb.entries_dmaaddr = ALIGN(trident->tlb.buffer.addr, SNDRV_TRIDENT_MAX_PAGES * 4); /* allocate shadow TLB page table (virtual addresses) */ trident->tlb.shadow_entries = vmalloc(SNDRV_TRIDENT_MAX_PAGES*sizeof(unsigned long)); - if (trident->tlb.shadow_entries == NULL) { - dev_err(trident->card->dev, - "unable to allocate shadow TLB entries\n"); + if (!trident->tlb.shadow_entries) return -ENOMEM; - } + /* allocate and setup silent page and initialise TLB entries */ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(trident->pci), SNDRV_TRIDENT_PAGE_SIZE, &trident->tlb.silent_page) < 0) { -- 2.14.0

2 1

[alsa-devel] [PATCH] ALSA: usb: caiaq: audio: Delete two error messages for a failed memory allocation in alloc_urbs()
by SF Markus Elfring 13 Aug '17

13 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 11 Aug 2017 19:39:06 +0200 Omit extra messages for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/usb/caiaq/audio.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 8f66ba730d69..1f306155e76a 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -731,7 +731,6 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) urbs = kmalloc(N_URBS * sizeof(*urbs), GFP_KERNEL); if (!urbs) { - dev_err(dev, "unable to kmalloc() urbs, OOM!?\n"); *ret = -ENOMEM; return NULL; } @@ -746,7 +745,6 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) urbs[i]->transfer_buffer = kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL); if (!urbs[i]->transfer_buffer) { - dev_err(dev, "unable to kmalloc() transfer buffer, OOM!?\n"); *ret = -ENOMEM; return urbs; } -- 2.14.0

3 2

[alsa-devel] [PATCH] ALSA: usb: Delete an error message for a failed memory allocation in two functions
by SF Markus Elfring 13 Aug '17

13 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 11 Aug 2017 18:38:25 +0200 Omit an extra message for a memory allocation failure in these functions. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/usb/quirks.c | 5 ++--- sound/usb/stream.c | 4 +--- 2 files changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d7b0b0a3a2db..9db40d58b4e2 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -146,10 +146,9 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, unsigned *rate_table = NULL; fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL); - if (!fp) { - usb_audio_err(chip, "cannot memdup\n"); + if (!fp) return -ENOMEM; - } + INIT_LIST_HEAD(&fp->list); if (fp->nr_rates > MAX_NR_RATES) { kfree(fp); diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 8e9548bc1f1a..d1776e5517ff 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -658,10 +658,8 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) continue; fp = kzalloc(sizeof(*fp), GFP_KERNEL); - if (! fp) { - dev_err(&dev->dev, "cannot malloc\n"); + if (!fp) return -ENOMEM; - } fp->iface = iface_no; fp->altsetting = altno; -- 2.14.0

2 1

[alsa-devel] [PATCH] ALSA: usx2y: Delete an error message for a failed memory allocation in two functions
by SF Markus Elfring 13 Aug '17

13 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 11 Aug 2017 17:40:57 +0200 Omit an extra message for a memory allocation failure in these functions. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/usb/usx2y/usbusx2yaudio.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index dd40ca9d858a..612aa7ff7e18 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -422,7 +422,5 @@ static int usX2Y_urbs_allocate(struct snd_usX2Y_substream *subs) - if (NULL == subs->tmpbuf) { - snd_printk(KERN_ERR "cannot malloc tmpbuf\n"); + if (!subs->tmpbuf) return -ENOMEM; - } } /* allocate and initialize data urbs */ for (i = 0; i < NRURBS; i++) { @@ -952,7 +950,6 @@ static int usX2Y_audio_stream_new(struct snd_card *card, int playback_endpoint, - if (NULL == usX2Y_substream[i]) { - snd_printk(KERN_ERR "cannot malloc\n"); + if (!usX2Y_substream[i]) return -ENOMEM; - } + usX2Y_substream[i]->usX2Y = usX2Y(card); } -- 2.14.0

2 1

[alsa-devel] [PATCH v3] ASoC: samsung: i2s: Null pointer dereference on samsung_i2s_remove
by Anton Vasilyev 12 Aug '17

12 Aug '17

If (quirks & QUIRK_SEC_DAI == 0) then samsung_i2s_probe() doesn't allocate sec_dai and pri_dai->sec_dai remains Null, but samsung_i2s_remove() performs pri_dai->sec_dai dereference in any case. The patch adds sec_dai check on Null before derefence at samsung_i2s_remove(). Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Anton Vasilyev <vasilyev(a)ispras.ru> --- v2: Drop initialization of sec_dai by NULL at samsung_i2s_remove as Sylwester Nawrocki suggest. --- v3: Fix typo in the comment --- sound/soc/samsung/i2s.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index af3ba4d..6f896e3 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1376,13 +1376,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) static int samsung_i2s_remove(struct platform_device *pdev) { - struct i2s_dai *pri_dai, *sec_dai; + struct i2s_dai *pri_dai; pri_dai = dev_get_drvdata(&pdev->dev); - sec_dai = pri_dai->sec_dai; - - pri_dai->sec_dai = NULL; - sec_dai->pri_dai = NULL; pm_runtime_get_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: Intel: constify snd_compr_codec_caps structures
by Julia Lawall 12 Aug '17

12 Aug '17

These snd_compr_codec_caps structures are only copied into other structures, so they can be const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/soc/intel/atom/sst/sst_drv_interface.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index ce689c5..71af544 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -407,7 +407,7 @@ static int sst_cdev_caps(struct snd_compr_caps *caps) return 0; } -static struct snd_compr_codec_caps caps_mp3 = { +static const struct snd_compr_codec_caps caps_mp3 = { .num_descriptors = 1, .descriptor[0].max_ch = 2, .descriptor[0].sample_rates[0] = 48000, @@ -424,7 +424,7 @@ static int sst_cdev_caps(struct snd_compr_caps *caps) .descriptor[0].formats = 0, }; -static struct snd_compr_codec_caps caps_aac = { +static const struct snd_compr_codec_caps caps_aac = { .num_descriptors = 2, .descriptor[1].max_ch = 2, .descriptor[0].sample_rates[0] = 48000,

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Re: [alsa-devel] [PATCH] ASoC: hdmi-codec: Use different name for playback streams
by Brian Norris 12 Aug '17

12 Aug '17

Hi Jeffy, You need to be more careful about the addressee's of your patches. No one on To/CC is a maintainer or a sufficiently-targeted mailing list. I doubt any of the maintainers will read your patch. (I've added alsa-devel for you, but given my understanding of at least Mark's patchwork workflow -- and of general patch etiquette -- it's important the patch is actually sent there in the first place.) Probably worth resending. Brian On Fri, Aug 11, 2017 at 04:29:33PM +0800, Jeffy Chen wrote: > Currently the hdmi i2s playback stream and hdmi spdif playback stream > are using the same name. So when they are enabled at the same time, > kernel will print this warning: > > [ 2.201835] hdmi-audio-codec hdmi-audio-codec.1.auto: ASoC: Failed to > create Playback debugfs file > > Assign different names to them to avoid that. > > Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> > --- > > sound/soc/codecs/hdmi-codec.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c > index 509ab51..a2af440 100644 > --- a/sound/soc/codecs/hdmi-codec.c > +++ b/sound/soc/codecs/hdmi-codec.c > @@ -696,7 +696,7 @@ static struct snd_soc_dai_driver hdmi_i2s_dai = { > .name = "i2s-hifi", > .id = DAI_ID_I2S, > .playback = { > - .stream_name = "Playback", > + .stream_name = "I2S Playback", > .channels_min = 2, > .channels_max = 8, > .rates = HDMI_RATES, > @@ -711,7 +711,7 @@ static const struct snd_soc_dai_driver hdmi_spdif_dai = { > .name = "spdif-hifi", > .id = DAI_ID_SPDIF, > .playback = { > - .stream_name = "Playback", > + .stream_name = "SPDIF Playback", > .channels_min = 2, > .channels_max = 2, > .rates = HDMI_RATES, > -- > 2.1.4 > >

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[alsa-devel] [PATCH 0/3] ASoC-Blackfin: Adjustments for some function implementations
by SF Markus Elfring 11 Aug '17

11 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 11 Aug 2017 22:34:56 +0200 A few update suggestions were taken into account from static source code analysis. Markus Elfring (3): Delete an error message for a failed memory allocation in sport_create() Use common error handling code in sport_create() Add some spaces for better code readability sound/soc/blackfin/bf6xx-sport.c | 38 +++++++++++++++++--------------------- 1 file changed, 17 insertions(+), 21 deletions(-) -- 2.14.0

1 3

[alsa-devel] [PATCH 10/11] ASoC: pxa: Handle return value of clk_prepare_enable.
by Arvind Yadav 11 Aug '17

11 Aug '17

clk_prepare_enable() can fail here and we must check its return value. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/soc/pxa/pxa2xx-i2s.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 3fb60ba..2312c22 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -164,10 +164,14 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_dmaengine_dai_dma_data *dma_data; + int ret; if (WARN_ON(IS_ERR(clk_i2s))) return -EINVAL; - clk_prepare_enable(clk_i2s); + ret = clk_prepare_enable(clk_i2s); + if (ret) + return ret; + clk_ena = 1; pxa_i2s_wait(); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: make skl_dsp_fw_ops const
by Bhumika Goyal 11 Aug '17

11 Aug '17

Make these const as they are only used during a copy operation. Done using Coccinelle. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/intel/skylake/bxt-sst.c | 2 +- sound/soc/intel/skylake/cnl-sst.c | 2 +- sound/soc/intel/skylake/skl-sst.c | 4 ++-- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 71042ac..5e1ac99 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -530,7 +530,7 @@ static int bxt_set_dsp_D3(struct sst_dsp *ctx, unsigned int core_id) return 0; } -static struct skl_dsp_fw_ops bxt_fw_ops = { +static const struct skl_dsp_fw_ops bxt_fw_ops = { .set_state_D0 = bxt_set_dsp_D0, .set_state_D3 = bxt_set_dsp_D3, .set_state_D0i3 = bxt_schedule_dsp_D0i3, diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c index 7dcfe66..48ef04d 100644 --- a/sound/soc/intel/skylake/cnl-sst.c +++ b/sound/soc/intel/skylake/cnl-sst.c @@ -277,7 +277,7 @@ static unsigned int cnl_get_errno(struct sst_dsp *ctx) return sst_dsp_shim_read(ctx, CNL_ADSP_ERROR_CODE); } -static struct skl_dsp_fw_ops cnl_fw_ops = { +static const struct skl_dsp_fw_ops cnl_fw_ops = { .set_state_D0 = cnl_set_dsp_D0, .set_state_D3 = cnl_set_dsp_D3, .load_fw = cnl_load_base_firmware, diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 6cc586d..a7ad734 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -503,7 +503,7 @@ static void skl_clear_module_table(struct sst_dsp *ctx) } } -static struct skl_dsp_fw_ops skl_fw_ops = { +static const struct skl_dsp_fw_ops skl_fw_ops = { .set_state_D0 = skl_set_dsp_D0, .set_state_D3 = skl_set_dsp_D3, .load_fw = skl_load_base_firmware, @@ -512,7 +512,7 @@ static void skl_clear_module_table(struct sst_dsp *ctx) .unload_mod = skl_unload_module, }; -static struct skl_dsp_fw_ops kbl_fw_ops = { +static const struct skl_dsp_fw_ops kbl_fw_ops = { .set_state_D0 = skl_set_dsp_D0, .set_state_D3 = skl_set_dsp_D3, .load_fw = skl_load_base_firmware, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: kbl: make snd_pcm_hw_constraint_list const
by Bhumika Goyal 11 Aug '17

11 Aug '17

Make these const as they are only passed to the function snd_pcm_hw_constraint_list having the corresponding argument as const. Done using Coccinelle. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 12b34b9..8bdd613 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -286,7 +286,7 @@ static int kabylake_5663_hdmi2_init(struct snd_soc_pcm_runtime *rtd) 48000, }; -static struct snd_pcm_hw_constraint_list constraints_rates = { +static const struct snd_pcm_hw_constraint_list constraints_rates = { .count = ARRAY_SIZE(rates), .list = rates, .mask = 0, @@ -296,7 +296,7 @@ static int kabylake_5663_hdmi2_init(struct snd_soc_pcm_runtime *rtd) 2, }; -static struct snd_pcm_hw_constraint_list constraints_channels = { +static const struct snd_pcm_hw_constraint_list constraints_channels = { .count = ARRAY_SIZE(channels), .list = channels, .mask = 0, @@ -425,7 +425,7 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) 16000, }; -static struct snd_pcm_hw_constraint_list constraints_16000 = { +static const struct snd_pcm_hw_constraint_list constraints_16000 = { .count = ARRAY_SIZE(rates_16000), .list = rates_16000, }; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: DaVinci: Delete an error message for a failed memory allocation in two functions
by SF Markus Elfring 11 Aug '17

11 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 11 Aug 2017 16:51:46 +0200 Omit an extra message for a memory allocation failure in these functions. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/davinci/davinci-mcasp.c | 2 -- sound/soc/davinci/davinci-vcif.c | 5 +---- 2 files changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 56ec1d301ac2..d486908d4e0e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1604,6 +1604,4 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( if (!pdata) { - dev_err(&pdev->dev, - "Failed to allocate memory for pdata\n"); ret = -ENOMEM; return pdata; } diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index c77d9218795a..5415b72393fa 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -213,8 +213,5 @@ static int davinci_vcif_probe(struct platform_device *pdev) - if (!davinci_vcif_dev) { - dev_dbg(&pdev->dev, - "could not allocate memory for private data\n"); + if (!davinci_vcif_dev) return -ENOMEM; - } /* DMA tx params */ davinci_vcif_dev->davinci_vc = davinci_vc; -- 2.14.0

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[alsa-devel] [PATCH] ASoC: Freescale: Delete an error message for a failed memory allocation in three functions
by SF Markus Elfring 11 Aug '17

11 Aug '17

>From 885ccd6c63291dcd4854a0cbaab5188cdc3db805 Mon Sep 17 00:00:00 2001 From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 11 Aug 2017 15:10:43 +0200 Subject: [PATCH] ASoC: Freescale: Delete an error message for a failed memory allocation in three functions Omit an extra message for a memory allocation failure in these functions. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/fsl/fsl_asrc_dma.c | 4 +--- sound/soc/fsl/fsl_dma.c | 1 - sound/soc/fsl/fsl_ssi.c | 4 +--- 3 files changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 282d841840b1..2baf19608bd0 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -282,7 +282,5 @@ static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream) - if (!pair) { - dev_err(dev, "failed to allocate pair\n"); + if (!pair) return -ENOMEM; - } pair->asrc_priv = asrc_priv; diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index ccadefceeff2..0ce172f86d6c 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -907,5 +907,4 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) if (!dma) { - dev_err(&pdev->dev, "could not allocate dma object\n"); of_node_put(ssi_np); return -ENOMEM; } diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 173cb8496641..64598d1183f8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1435,7 +1435,5 @@ static int fsl_ssi_probe(struct platform_device *pdev) - if (!ssi_private) { - dev_err(&pdev->dev, "could not allocate DAI object\n"); + if (!ssi_private) return -ENOMEM; - } ssi_private->soc = of_id->data; ssi_private->dev = &pdev->dev; -- 2.14.0

2 1

[alsa-devel] [PATCH] ASoC: dwc: Delete an error message for a failed memory allocation in dw_i2s_probe()
by SF Markus Elfring 11 Aug '17

11 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 11 Aug 2017 16:04:53 +0200 Omit an extra message for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/dwc/dwc-i2s.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index eebe9d34e7a0..e27e21f8569a 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -617,10 +617,8 @@ static int dw_i2s_probe(struct platform_device *pdev) const char *clk_id; dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); - if (!dev) { - dev_warn(&pdev->dev, "kzalloc fail\n"); + if (!dev) return -ENOMEM; - } dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); if (!dw_i2s_dai) -- 2.14.0

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[alsa-devel] [RFC PATCH 0/9] ASoC: qcom: Add support to QDSP based HDMI audio
by srinivas.kandagatla＠linaro.org 11 Aug '17

11 Aug '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset aims to provide a very basic version of QCOM DSP based audio support which is available in downstream andriod kernels. This patchset only support Digital audio based for HDMI-TX and will add support to other features as we move on. Why do we need this now: On All Qualcomm SOCs the audio services are usually provided by an on chip DSP using LPASS, however LPASS is mostly tied with a Q6DSP on most of the SOCs. So Bypassing the DSP is becoming almost impossible with new generation of QCOM SOCs. QDSP has 3 modules AFE (Audio FrontEnd), ADM (Audio Device Manager), and ASM(Audio Stream Manager) to provide this audio services. All these services use APR (Asynchronous Packet Router) protocol via smd/glink transport to communicate with Application processor. More details on each module is availble in there respective patch. This patchset is tested on DB820c, with HDMI audio playback on top of mainline with few additional patches. Here is block diagram to give a quick overview of the components +---------+ +---------+ +---------+ | q6pcm | |q6routing| | q6hdmi | | ASoC(FE | <------> | ASoC(BE | <-----> | ASoC(BE | | cpu dai)| |Platform)| | cpu dai)| | | | | | | +---------+ +---------+ +---------+ ^ ^ ^ | | | | +------------------+----------------+ | | | | | | v v v v v +---------+ +---------+ +---------+ | | | | | | | q6ASM | | q6ADM | | q6AFE | | | | | | | | | | | | | +---------+ +---------+ +---------+ ^ ^ ^ ^ | | | CPU Side | ------+---------------------+-------------------+-------- | | | | |APR(smd/glink) | | | | | | | | +------------------+----------------+ | | | | | +-----+--+-----------------------------------+--+------- | | | | QDSP Side | v v v v v +---------+ +---------+ +---------+ | | | | | | | ASM | <------> | ADM | <-----> | AFE | | | | | | | | | | | | | +---------+ +---------+ +---------+ ^ | +-------------------+ | ---------------------------+-------------------------- | Audio I/O | v v +--------------------------------------------------+ | Audio devices | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | +--------------------------------------------------+ Srinivas Kandagatla (9): soc: qcom: add support APR driver ASoC: qcom: qdsp6v2: Add support to Q6AFE ASoC: qcom: qdsp6v2: Add support to Q6ADM ASoC: qcom: qdsp6v2: Add support to Q6ASM ASoC: qcom: qdsp6v2: Add support to q6 routing driver ASoC: qcom: qdsp6v2: Add support to q6 HDMI dai driver ASoC: qcom: qdsp6v2: Add support to q6 pcm driver ASoC: qcom: apq8096: Add db820c machine driver arm64: dts: msm8996: db820c: Add sound card support .../devicetree/bindings/soc/qcom/qcom,apr.txt | 66 ++ .../devicetree/bindings/sound/qcom,apq8096.txt | 85 ++ .../devicetree/bindings/sound/qcom,q6adm.txt | 15 + .../devicetree/bindings/sound/qcom,q6afe.txt | 15 + .../devicetree/bindings/sound/qcom,q6asm.txt | 15 + .../devicetree/bindings/sound/qcom,q6hdmi.txt | 27 + .../bindings/sound/qcom,q6pcm-routing.txt | 21 + .../devicetree/bindings/sound/qcom,q6pcm.txt | 21 + arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi | 37 + arch/arm64/boot/dts/qcom/msm8996.dtsi | 67 ++ drivers/soc/qcom/Kconfig | 8 + drivers/soc/qcom/Makefile | 1 + drivers/soc/qcom/apr.c | 406 ++++++++ include/dt-bindings/sound/qcom,afe.h | 8 + include/dt-bindings/sound/qcom,asm.h | 13 + include/linux/soc/qcom/apr.h | 163 ++++ sound/soc/qcom/Kconfig | 46 + sound/soc/qcom/Makefile | 5 + sound/soc/qcom/apq8096.c | 174 ++++ sound/soc/qcom/qdsp6v2/Makefile | 6 + sound/soc/qcom/qdsp6v2/common.h | 225 +++++ sound/soc/qcom/qdsp6v2/q6adm-v2.h | 76 ++ sound/soc/qcom/qdsp6v2/q6adm.c | 603 ++++++++++++ sound/soc/qcom/qdsp6v2/q6afe-v2.h | 62 ++ sound/soc/qcom/qdsp6v2/q6afe.c | 499 ++++++++++ sound/soc/qcom/qdsp6v2/q6asm-v2.h | 176 ++++ sound/soc/qcom/qdsp6v2/q6asm.c | 1008 ++++++++++++++++++++ sound/soc/qcom/qdsp6v2/q6hdmi.c | 258 +++++ sound/soc/qcom/qdsp6v2/q6pcm.c | 558 +++++++++++ sound/soc/qcom/qdsp6v2/q6routing-v2.h | 8 + sound/soc/qcom/qdsp6v2/q6routing.c | 403 ++++++++ 31 files changed, 5075 insertions(+) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6adm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6afe.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6hdmi.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6pcm-routing.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6pcm.txt create mode 100644 drivers/soc/qcom/apr.c create mode 100644 include/dt-bindings/sound/qcom,afe.h create mode 100644 include/dt-bindings/sound/qcom,asm.h create mode 100644 include/linux/soc/qcom/apr.h create mode 100644 sound/soc/qcom/apq8096.c create mode 100644 sound/soc/qcom/qdsp6v2/Makefile create mode 100644 sound/soc/qcom/qdsp6v2/common.h create mode 100644 sound/soc/qcom/qdsp6v2/q6adm-v2.h create mode 100644 sound/soc/qcom/qdsp6v2/q6adm.c create mode 100644 sound/soc/qcom/qdsp6v2/q6afe-v2.h create mode 100644 sound/soc/qcom/qdsp6v2/q6afe.c create mode 100644 sound/soc/qcom/qdsp6v2/q6asm-v2.h create mode 100644 sound/soc/qcom/qdsp6v2/q6asm.c create mode 100644 sound/soc/qcom/qdsp6v2/q6hdmi.c create mode 100644 sound/soc/qcom/qdsp6v2/q6pcm.c create mode 100644 sound/soc/qcom/qdsp6v2/q6routing-v2.h create mode 100644 sound/soc/qcom/qdsp6v2/q6routing.c -- 2.9.3

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[alsa-devel] [PATCH] ASoC: Broadwell RT5677: Delete an error message for a failed memory allocation in bdw_rt5677_probe()
by SF Markus Elfring 11 Aug '17

11 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 11 Aug 2017 13:05:40 +0200 Omit an extra message for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- 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 058b8ccedf02..4b1431c3b460 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -349,7 +349,5 @@ static int bdw_rt5677_probe(struct platform_device *pdev) - if (!bdw_rt5677) { - dev_err(&pdev->dev, "Can't allocate bdw_rt5677\n"); + if (!bdw_rt5677) return -ENOMEM; - } snd_soc_card_set_drvdata(&bdw_rt5677_card, bdw_rt5677); -- 2.14.0

1 0

[alsa-devel] [PATCH] ASoC: Baytrail SST MAX98090: Delete an error message for a failed memory allocation in byt_max98090_probe()
by SF Markus Elfring 11 Aug '17

11 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 11 Aug 2017 12:42:40 +0200 Omit an extra message for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/intel/boards/byt-max98090.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/byt-max98090.c b/sound/soc/intel/boards/byt-max98090.c index 047be7fa0ce9..fa68bb32adef 100644 --- a/sound/soc/intel/boards/byt-max98090.c +++ b/sound/soc/intel/boards/byt-max98090.c @@ -152,10 +152,8 @@ static int byt_max98090_probe(struct platform_device *pdev) int ret_val; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC); - if (!priv) { - dev_err(&pdev->dev, "allocation failed\n"); + if (!priv) return -ENOMEM; - } ret_val = devm_acpi_dev_add_driver_gpios(dev->parent, acpi_byt_max98090_gpios); if (ret_val) -- 2.14.0

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[alsa-devel] [PATCH] ALSA: seq: Fix CONFIG_SND_SEQ_MIDI dependency
by Takashi Iwai 11 Aug '17

11 Aug '17

The commit 0181307abc1d ("ALSA: seq: Reorganize kconfig and build") rewrote the dependency of each sequencer module in a standard way, but there was one change applied mistakenly: CONFIG_SND_SEQ_MIDI isn't enabled properly by CONFIG_SND_RAWMIDI. I seem to have changed the wrong one instead, CONFIG_SND_SEQ_MIDI_EMUL, which is eventually reverse-selected by CONFIG_SND_SEQ_MIDI itself. This ended up the lack of snd-seq-midi module as reported below. The fix is to put def_tristate properly to CONFIG_SND_SEQ_MIDI instead of *_MIDI_EMUL entry. Fixes: 0181307abc1d ("ALSA: seq: Reorganize kconfig and build") Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196633 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/Kconfig b/sound/core/seq/Kconfig index a536760a94c2..45c1336c6597 100644 --- a/sound/core/seq/Kconfig +++ b/sound/core/seq/Kconfig @@ -47,10 +47,10 @@ config SND_SEQ_HRTIMER_DEFAULT timer. config SND_SEQ_MIDI_EVENT - def_tristate SND_RAWMIDI + tristate config SND_SEQ_MIDI - tristate + def_tristate SND_RAWMIDI select SND_SEQ_MIDI_EVENT config SND_SEQ_MIDI_EMUL -- 2.13.3

1 0

[alsa-devel] [PATCH] ASoC: kirkwood: Delete an error message for a failed memory allocation in kirkwood_i2s_dev_probe()
by SF Markus Elfring 11 Aug '17

11 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 11 Aug 2017 09:35:33 +0200 Omit an extra message for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/kirkwood/kirkwood-i2s.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 3b09f71b9535..105a73cc5158 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -538,10 +538,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) int err; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) { - dev_err(&pdev->dev, "allocation failed\n"); + if (!priv) return -ENOMEM; - } + dev_set_drvdata(&pdev->dev, priv); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); -- 2.14.0

1 0

[alsa-devel] [PATCH] ASoC: rockchip: Delete an error message for a failed memory allocation in rockchip_i2s_probe()
by SF Markus Elfring 10 Aug '17

10 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 10 Aug 2017 18:38:09 +0200 Omit an extra message for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/rockchip/rockchip_i2s.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 199338fdeda0..b6590467fd14 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -579,10 +579,8 @@ static int rockchip_i2s_probe(struct platform_device *pdev) int val; i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); - if (!i2s) { - dev_err(&pdev->dev, "Can't allocate rk_i2s_dev\n"); + if (!i2s) return -ENOMEM; - } i2s->dev = &pdev->dev; -- 2.14.0

1 0

[alsa-devel] [PATCH] ASoC: samsung: i2s: Null pointer dereference on samsung_i2s_remove
by Anton Vasilyev 10 Aug '17

10 Aug '17

If (quirks & QUIRK_SEC_DAI == 0) then samsung_i2s_probe() doesn't allocate sec_dai and pri_dai->sec_dai remains Null, but samsung_i2s_remove() permorms pri_dai->sec_dai dereference in any case. The patch adds sec_dai check on Null before derefence at samsung_i2s_remove(). Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Anton Vasilyev <vasilyev(a)ispras.ru> --- sound/soc/samsung/i2s.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index af3ba4d..6beeea8 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1382,7 +1382,8 @@ static int samsung_i2s_remove(struct platform_device *pdev) sec_dai = pri_dai->sec_dai; pri_dai->sec_dai = NULL; - sec_dai->pri_dai = NULL; + if (sec_dai) + sec_dai->pri_dai = NULL; pm_runtime_get_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: fsi: Delete an error message for a failed memory allocation in fsi_probe()
by SF Markus Elfring 10 Aug '17

10 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 10 Aug 2017 18:18:20 +0200 Omit an extra message for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/sh/fsi.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 005b21506556..1f41cf1ff830 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1962,10 +1962,8 @@ static int fsi_probe(struct platform_device *pdev) } master = devm_kzalloc(&pdev->dev, sizeof(*master), GFP_KERNEL); - if (!master) { - dev_err(&pdev->dev, "Could not allocate master\n"); + if (!master) return -ENOMEM; - } master->base = devm_ioremap_nocache(&pdev->dev, res->start, resource_size(res)); -- 2.14.0

1 0

[alsa-devel] [PATCH] conf/ucm: DB410c-HiFi: add CIC selection
by srinivas.kandagatla＠linaro.org 10 Aug '17

10 Aug '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patch adds CIC selection controls which have been added recently to the kernel to select mic source. Without this patch user has to manually select the control to record from DMIC or AMIC. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- src/conf/ucm/DB410c/HiFi | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/src/conf/ucm/DB410c/HiFi b/src/conf/ucm/DB410c/HiFi index f9cbcf0..dcb7ac9 100644 --- a/src/conf/ucm/DB410c/HiFi +++ b/src/conf/ucm/DB410c/HiFi @@ -108,6 +108,7 @@ SectionDevice."Handset" { EnableSequence [ cdev "hw:0" cset "name='DEC1 MUX' ADC2" + cset "name='CIC1 MUX' AMIC" cset "name='ADC2 Volume' 8" cset "name='ADC2 MUX' INP2" ] @@ -130,6 +131,7 @@ SectionDevice."Primarymic" { EnableSequence [ cdev "hw:0" cset "name='DEC1 MUX' ADC1" + cset "name='CIC1 MUX' AMIC" cset "name='ADC1 Volume' 8" ] @@ -150,6 +152,7 @@ SectionDevice."Secondarymic" { EnableSequence [ cdev "hw:0" cset "name='DEC1 MUX' ADC2" + cset "name='CIC1 MUX' AMIC" cset "name='ADC2 Volume' 8" cset "name='ADC2 MUX' INP2" ] @@ -172,6 +175,7 @@ SectionDevice."DMIC" { EnableSequence [ cdev "hw:0" cset "name='DEC1 MUX' DMIC1" + cset "name='CIC1 MUX' DMIC" ] DisableSequence [ -- 2.9.3

2 1

[alsa-devel] [PATCH 0/7] constify pci snd_pcm_ops structures
by Arvind Yadav 10 Aug '17

10 Aug '17

snd_pcm_ops are not supposed to change at runtime. All functions working with snd_pcm_ops provided by <sound/pcm.h> work with const snd_pcm_ops. So mark the non-const structs as const. Arvind Yadav (7): [PATCH 1/7] ALSA: ali5451: constify snd_pcm_ops structures [PATCH 2/7] ALSA: au88x0: constify snd_pcm_ops structures [PATCH 3/7] ALSA: echoaudio: constify snd_pcm_ops structures [PATCH 4/7] ALSA: intel8x0: constify snd_pcm_ops structures [PATCH 5/7] ALSA: intel8x0m: constify snd_pcm_ops structures [PATCH 6/7] ALSA: sis7019: constify snd_pcm_ops structures [PATCH 7/7] ALSA: trident: constify snd_pcm_ops structures sound/pci/ali5451/ali5451.c | 12 ++++++------ sound/pci/au88x0/au88x0_pcm.c | 2 +- sound/pci/echoaudio/echoaudio.c | 8 ++++---- sound/pci/intel8x0.c | 24 ++++++++++++------------ sound/pci/intel8x0m.c | 8 ++++---- sound/pci/sis7019.c | 4 ++-- sound/pci/trident/trident_main.c | 4 ++-- 7 files changed, 31 insertions(+), 31 deletions(-) -- 1.9.1

2 8

[alsa-devel] [PATCH 0/4] ALSA: emux: Adjustments for four function implementations
by SF Markus Elfring 10 Aug '17

10 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 9 Aug 2017 09:50:05 +0200 A few update suggestions were taken into account from static source code analysis. Markus Elfring (4): Adjust one function call together with a variable assignment Improve a size determination in two functions Adjust four checks for null pointers Delete two error messages for a failed memory allocation in snd_emux_create_port() sound/synth/emux/emux_seq.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) -- 2.13.4

2 5

[alsa-devel] [PATCH] ASoC: spear: Remove redundant message
by dspmihai 10 Aug '17

10 Aug '17

This was reported by checkpatch.pl Signed-off-by: Despotovici Mihai <despotovici.mihai(a)gmail.com> --- sound/soc/spear/spdif_in.c | 4 +--- sound/soc/spear/spdif_out.c | 4 +--- 2 files changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 977a078..0f3b1a8 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -216,10 +216,8 @@ static int spdif_in_probe(struct platform_device *pdev) return -EINVAL; host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); - if (!host) { - dev_warn(&pdev->dev, "kzalloc fail\n"); + if (!host) return -ENOMEM; - } host->io_base = io_base; host->irq = platform_get_irq(pdev, 0); diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 0a72d52..58d5843 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -282,10 +282,8 @@ static int spdif_out_probe(struct platform_device *pdev) int ret; host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); - if (!host) { - dev_warn(&pdev->dev, "kzalloc fail\n"); + if (!host) return -ENOMEM; - } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); host->io_base = devm_ioremap_resource(&pdev->dev, res); -- 2.7.4

3 2

[alsa-devel] [PATCH 0/3] ASoC: rsnd: request_irq/free_irq
by Kuninori Morimoto 10 Aug '17

10 Aug '17

Hi Mark Renesas sound driver supports many features, thus, it has many data path. I noticed request_irq/free_irq balance are not correct when MIXer case, and PIO mode case. These patches solve these issue. Kuninori Morimoto (3): ASoC: rsnd: call free_irq() both DMA/PIO mode ASoC: rsnd: add rsnd_ssi_flags_has/add() macro ASoC: rsnd: call request_irq/free_irq once in MIX case sound/soc/sh/rcar/ssi.c | 81 ++++++++++++++++++++++++++++++------------------- 1 file changed, 50 insertions(+), 31 deletions(-) -- 1.9.1

2 6

[alsa-devel] [PATCH 0/2] ASoC: make snd_soc_codec_driver structures const
by Bhumika Goyal 10 Aug '17

10 Aug '17

2 7

[alsa-devel] [PATCH] ASoC: compress: Delete error messages for a failed memory allocation in snd_soc_new_compress()
by SF Markus Elfring 10 Aug '17

10 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 10 Aug 2017 16:21:34 +0200 Omit extra messages for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/soc-compress.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 206f36bf43e8..6c867d50e036 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -764,15 +764,12 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) direction = SND_COMPRESS_CAPTURE; compr = kzalloc(sizeof(*compr), GFP_KERNEL); - if (compr == NULL) { - snd_printk(KERN_ERR "Cannot allocate compr\n"); + if (!compr) return -ENOMEM; - } compr->ops = devm_kzalloc(rtd->card->dev, sizeof(soc_compr_ops), GFP_KERNEL); - if (compr->ops == NULL) { - dev_err(rtd->card->dev, "Cannot allocate compressed ops\n"); + if (!compr->ops) { ret = -ENOMEM; goto compr_err; } -- 2.14.0

2 1

[alsa-devel] [PATCH v1 0/6] ASoC: codecs: msm8916-wcd-analog: Add support to MBHC
by srinivas.kandagatla＠linaro.org 10 Aug '17

10 Aug '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset adds support to MBHC(Multibutton headset control) block in PM8916 analog block. MBHC support comes from2 blocks first mechanical headset detection and second headset type, 5 button detection. This patchset adds support to: 1> Support to NC and NO type of headset Jacks. 2> Mechanical insertion and detection of headset jack. 3> Detect a 3 pole Headphone and a 4 pole Headset. 4> Detect 5 buttons. First patch fixes error code returned by snd_soc_codec_set_jack() to be more precise for users. Second patch is patch from Damien, followed by few fixes to codec driver w.r.t mbhc and final patch to add mbhc support. Damien sent a similar patchset to add support to mechanical detection, but that patch has issues and will not work on most usecases (for example after a playback/capture session, multicodec case). So I only picked up the BIT mask patch from that series. Tested it on DB410c with Audio Mezz board with 4 pole and 3 pole and during playback and recording use cases. root@linaro-alip:~# evtest No device specified, trying to scan all of /dev/input/event* Available devices: /dev/input/event0: pm8941_pwrkey /dev/input/event1: DB410c Headset Jack /dev/input/event2: gpio_keys Select the device event number [0-2]: 1 Input driver version is 1.0.1 Input device ID: bus 0x0 vendor 0x0 product 0x0 version 0x0 Input device name: "DB410c Headset Jack" Supported events: Event type 0 (EV_SYN) Event type 1 (EV_KEY) Event code 114 (KEY_VOLUMEDOWN) Event code 115 (KEY_VOLUMEUP) Event code 226 (KEY_MEDIA) Event code 260 (BTN_4) Event code 582 (KEY_VOICECOMMAND) Event type 5 (EV_SW) Event code 2 (SW_HEADPHONE_INSERT) state 1 Event code 4 (SW_MICROPHONE_INSERT) state 1 Properties: Testing ... (interrupt to exit) Event: time 1501001112.059684, type 1 (EV_KEY), code 226 (KEY_MEDIA), value 1 Event: time 1501001112.059684, type 5 (EV_SW), code 2 (SW_HEADPHONE_INSERT), value 0 Event: time 1501001112.059684, type 5 (EV_SW), code 4 (SW_MICROPHONE_INSERT), value 0 Event: time 1501001112.059684, -------------- SYN_REPORT ------------ Event: time 1501001112.254116, type 1 (EV_KEY), code 226 (KEY_MEDIA), value 0 Event: time 1501001112.254116, -------------- SYN_REPORT ------------ Event: time 1501001113.154184, type 1 (EV_KEY), code 114 (KEY_VOLUMEDOWN), value 1 Event: time 1501001113.154184, -------------- SYN_REPORT ------------ Event: time 1501001113.319658, type 1 (EV_KEY), code 114 (KEY_VOLUMEDOWN), value 0 Event: time 1501001113.319658, -------------- SYN_REPORT ------------ Event: time 1501001113.781084, type 1 (EV_KEY), code 226 (KEY_MEDIA), value 1 Event: time 1501001113.781084, -------------- SYN_REPORT ------------ Event: time 1501001114.024930, type 1 (EV_KEY), code 226 (KEY_MEDIA), value 0 Event: time 1501001114.024930, -------------- SYN_REPORT ------------ Event: time 1501001114.419082, type 1 (EV_KEY), code 114 (KEY_VOLUMEDOWN), value 1 Event: time 1501001114.419082, -------------- SYN_REPORT ------------ Event: time 1501001114.576011, type 1 (EV_KEY), code 114 (KEY_VOLUMEDOWN), value 0 Event: time 1501001114.576011, -------------- SYN_REPORT ------------ Event: time 1501001114.795281, type 1 (EV_KEY), code 114 (KEY_VOLUMEDOWN), value 1 Event: time 1501001114.795281, -------------- SYN_REPORT ------------ Event: time 1501001114.863333, type 1 (EV_KEY), code 114 (KEY_VOLUMEDOWN), value 0 Event: time 1501001114.863333, -------------- SYN_REPORT ------------ Event: time 1501001116.603372, type 1 (EV_KEY), code 226 (KEY_MEDIA), value 1 Event: time 1501001116.603372, -------------- SYN_REPORT ------------ Damien Riegel (1): ASoC: codecs: msm8916-analog: fix DIG_CLK_CTL_RXD3_CLK_EN define Srinivas Kandagatla (5): ASoC: jack: fix snd_soc_codec_set_jack return error ASoC: codecs: msm8916-wcd-analog: move codec reset to probe ASoC: codecs: msm8916-wcd-analog: set default micbias voltage to 1.8 ASoC: codecs: msm8916-wcd-analog: add MBHC support ASoC: qcom: apq8016-sbc: Add support to Headset JACK .../bindings/sound/qcom,msm8916-wcd-analog.txt | 9 + sound/soc/codecs/msm8916-wcd-analog.c | 359 +++++++++++++++++++-- sound/soc/codecs/msm8916-wcd.h | 12 + sound/soc/qcom/apq8016_sbc.c | 52 +++ sound/soc/soc-jack.c | 2 +- 5 files changed, 410 insertions(+), 24 deletions(-) create mode 100644 sound/soc/codecs/msm8916-wcd.h -- 2.9.3

3 19

[alsa-devel] [PATCH] ASoC: rsnd: Delete an error message for a failed memory allocation in three functions
by SF Markus Elfring 10 Aug '17

10 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 10 Aug 2017 17:13:19 +0200 Omit an extra message for a memory allocation failure in these functions. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/sh/rcar/adg.c | 4 +--- sound/soc/sh/rcar/core.c | 4 +--- sound/soc/sh/rcar/gen.c | 4 +--- 3 files changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 5b5389e5b92b..938baff86ef2 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -586,10 +586,8 @@ int rsnd_adg_probe(struct rsnd_priv *priv) int ret; adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); - if (!adg) { - dev_err(dev, "ADG allocate failed\n"); + if (!adg) return -ENOMEM; - } ret = rsnd_mod_init(priv, &adg->mod, &adg_ops, NULL, NULL, 0, 0); diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index c34562388cfa..9e331a474a21 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1402,10 +1402,8 @@ static int rsnd_probe(struct platform_device *pdev) * init priv data */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); - if (!priv) { - dev_err(dev, "priv allocate failed\n"); + if (!priv) return -ENODEV; - } priv->pdev = pdev; priv->flags = (unsigned long)of_device_get_match_data(dev); diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index ee00e3516911..f04c4100043a 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -406,10 +406,8 @@ int rsnd_gen_probe(struct rsnd_priv *priv) int ret; gen = devm_kzalloc(dev, sizeof(*gen), GFP_KERNEL); - if (!gen) { - dev_err(dev, "GEN allocate failed\n"); + if (!gen) return -ENOMEM; - } priv->gen = gen; -- 2.14.0

2 1

[alsa-devel] [PATCH 0/3 v2][resend] ASoC: Component level pcm_new/pcm_free v2
by Kuninori Morimoto 10 Aug '17

10 Aug '17

Hi Mark I posted this patch-set 2weeks ago, and no response, no objection from community. Thus, I re-post these again. These are Component level pcm_new/free v2. v1 pcm_new/free had breaked some platform, because it called "card" connected component callback. It should call "rtd" connected one's. Special thanks to Arnaud who tested this v2 patch on his environment. These patch-set adds rtd connected component list (= rtdcom) and replace current platform level pcm_new/free into component level pcm_new/free I don't know this naming (= rtdcom) sense is nice or not ;P I can replace it to more nicer naming if someone have nice sense. Kuninori Morimoto (3): ASoC: soc-core: add snd_soc_rtdcom_xxx() ASoC: use snd_soc_rtdcom_add() and convert to consistent operation ASoC: add Component level pcm_new/pcm_free v2 include/sound/soc.h | 21 ++++++- sound/soc/soc-core.c | 153 +++++++++++++++++++++++++++++++++++++++------------ sound/soc/soc-pcm.c | 68 +++++++++++++++-------- 3 files changed, 183 insertions(+), 59 deletions(-) -- 1.9.1

2 6

[alsa-devel] [Test Request v2] Component level pcm_new/pcm_free v2
by Kuninori Morimoto 10 Aug '17

10 Aug '17

Hi Mark, Johan, Brian These are v2 of Component level pcm_new/free patches v2. Previous patch-set still had issue if 2 platform were registered. I think this version solved this issue. But, I'm not sure if you are using unexpected usage which I don't use. So, can you please test these patches ? replacing platform to component is based on this patch-set Kuninori Morimoto (3): ASoC: soc-core: add snd_soc_rtdcom_xxx() ASoC: use snd_soc_rtdcom_add() and convert to consistent operation ASoC: add Component level pcm_new/pcm_free v2 Best regards --- Kuninori Morimoto

3 6

[alsa-devel] [PATCH] ASoC: Intel: kbl_rt5663_rt5514_max98927: Add rt5514 spi dailink
by Harsha Priya 10 Aug '17

10 Aug '17

This patch adds a dai to rt5514-spi driver for wake on voice functionality. Signed-off-by: Hsin-yu Chao <hychao(a)google.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- sound/soc/intel/Kconfig | 1 + sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 9 +++++++++ 2 files changed, 10 insertions(+) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b301bff..e3ca40d 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -260,6 +260,7 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH select SND_SOC_INTEL_SKYLAKE select SND_SOC_RT5663 select SND_SOC_RT5514 + select SND_SOC_RT5514_SPI select SND_SOC_MAX98927 select SND_SOC_HDAC_HDMI help diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 5ec9428..b51dff6 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -65,6 +65,7 @@ enum { KBL_DPCM_AUDIO_HS_PB, KBL_DPCM_AUDIO_ECHO_REF_CP, KBL_DPCM_AUDIO_DMIC_CP, + KBL_DPCM_AUDIO_RT5514_DSP, KBL_DPCM_AUDIO_HDMI1_PB, KBL_DPCM_AUDIO_HDMI2_PB, }; @@ -480,6 +481,14 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) .capture_only = 1, .nonatomic = 1, }, + [KBL_DPCM_AUDIO_RT5514_DSP] = { + .name = "rt5514 dsp", + .stream_name = "Wake on Voice", + .cpu_dai_name = "spi-PRP0001:00", + .platform_name = "spi-PRP0001:00", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", .stream_name = "dmiccap", -- 1.9.1

2 2

[alsa-devel] [PATCH 0/5] ASoC: dapm: Adjustments for some function implementations
by SF Markus Elfring 10 Aug '17

10 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 10 Aug 2017 14:54:32 +0200 A few update suggestions were taken into account from static source code analysis. Markus Elfring (5): Delete error messages for a failed memory allocation in snd_soc_dapm_new_pcm() Improve a size determination in two functions Use kcalloc() in snd_soc_dapm_new_widgets() Adjust one function call together with a variable assignment Adjust seven checks for null pointers sound/soc/soc-dapm.c | 36 ++++++++++++++++-------------------- 1 file changed, 16 insertions(+), 20 deletions(-) -- 2.14.0

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[alsa-devel] [PATCH] ASoC: spear: Delete an error message for a failed memory allocation in two functions
by SF Markus Elfring 10 Aug '17

10 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 10 Aug 2017 09:50:51 +0200 Omit an extra message for a memory allocation failure in these functions. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/spear/spdif_in.c | 4 +--- sound/soc/spear/spdif_out.c | 4 +--- 2 files changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 7f32527fc3c8..01b28e59b4c0 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -216,10 +216,8 @@ static int spdif_in_probe(struct platform_device *pdev) return -EINVAL; host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); - if (!host) { - dev_warn(&pdev->dev, "kzalloc fail\n"); + if (!host) return -ENOMEM; - } host->io_base = io_base; host->irq = platform_get_irq(pdev, 0); diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 0a72d52d533e..58d5843811f9 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -282,10 +282,8 @@ static int spdif_out_probe(struct platform_device *pdev) int ret; host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); - if (!host) { - dev_warn(&pdev->dev, "kzalloc fail\n"); + if (!host) return -ENOMEM; - } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); host->io_base = devm_ioremap_resource(&pdev->dev, res); -- 2.13.4

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[alsa-devel] sidebar link to Bug Tracker is 404
by 積丹尼 Dan Jacobson 10 Aug '17

10 Aug '17

In the sidebar of https://www.alsa-project.org/ there is a link to Bug Tracker. Alas that link is broken.

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[alsa-devel] speaker-test man page should mention the proper way to exit
by 積丹尼 Dan Jacobson 10 Aug '17

10 Aug '17

The speaker-test man page should mention the proper way to exit speaker-test. ^C gives Write error: -4,Interrupted system call xrun_recovery failed: -4,Interrupted system call Transfer failed: Interrupted system call except when run with -c 2. See also https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=802935

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[alsa-devel] [PATCH v5 0/7] add support for Sama5d2 audio PLLs and enable ClassD
by Quentin Schulz 10 Aug '17

10 Aug '17

This patch series adds support for the audio PLLs and enables ClassD that can be found in ATMEL Sama5d2 SoC. There are two audio PLLs (PMC and PAD) that shares the same parent (FRAC). FRAC can output between 620 and 700MHz and only multiply the rate of its parent. The two audio PLLs then divide the FRAC rate to best match the asked rate. I basically took an old patch series posted by Nicolas on December, 6th 2016[1][2][3] and the comments Boris did on the first version[4] Nicolas sent on July, 15th 2015. I also fixed the function used to compute the divisors, removed useless spinlocks and added a range to the audio frac PLL to stay within vendor's supported range. Clocks that are children of gclk (generated-clk) are now able to propagate rate to the audio PLL clocks when needed. However, there are multiple children clocks that could technically change the rate of audio_pll (via gck). With the rate locking introduced in Jerome Brunet's patch series[5], the first consumer to enable the clock will be the one definitely setting the rate of the clock. Without the rate locking, the last consumer to set the rate will be able to mess with the rate. Since audio IPs are most likely to request the same rate, we enforce that the only clks able to modify gck rate are those of audio IPs. To remain consistent, we deny other clocks to be children of audio_pll. I suggest this whole series going through the clk subsystem. v5: - replaced clk_audio_pll_frac_round_rate by clk_audio_pll_frac_determine_rate to also return min and max rate for the clk, suggested by Stephen Boyd during v3, v4: - initialized the clk_init_data structure, - switched to of_clk_parent_fill instead of manually setting parent_names, - removed include of clkdev.h in favor of slab.h, - removed unnecessary second cast in operation, - used clamp instead of two conditions, - removed dependency on clock-output-names, the name is gotten from DT node, - merged all three drivers for audio PLLs (FRAC, PMC, PAD) into one, v3: - added a flag per generated clock to know which one is allowed to modify audio_pll rate, - set the flag in setup function depending on the compatible and the clock ID, v2: - split big patch containing dt-binding, pll drivers and classd modifications, - removed unused AUDIO_PLL_*FOUT* defines from clk-audio-pll-pmc, - added conditions for audio pll rate setting restriction for SSC and I2S, - renamed all mentions of ACLK to GCLK in classD driver, Thanks, Quentin [1] https://patchwork.kernel.org/patch/9462351/ [2] https://patchwork.kernel.org/patch/9462347/ [3] https://patchwork.kernel.org/patch/9462349/ [4] https://www.spinics.net/lists/arm-kernel/msg436120.html [5] http://www.spinics.net/lists/linux-clk/msg17927.html Quentin Schulz (7): clk: at91: clk-generated: remove useless divisor loop dt-bindings: clk: at91: add audio plls to the compatible list clk: at91: add audio pll clock drivers clk: at91: clk-generated: create function to find best_diff clk: at91: clk-generated: make gclk determine audio_pll rate ASoC: atmel-classd: remove aclk clock from DT binding ASoC: atmel-classd: remove aclk clock Documentation/devicetree/bindings/clock/at91-clock.txt | 10 +- Documentation/devicetree/bindings/sound/atmel-classd.txt | 9 +- arch/arm/mach-at91/Kconfig | 4 +- drivers/clk/at91/Makefile | 1 +- drivers/clk/at91/clk-audio-pll.c | 536 ++++++++- drivers/clk/at91/clk-generated.c | 101 +- include/linux/clk/at91_pmc.h | 25 +- sound/soc/atmel/atmel-classd.c | 47 +- 8 files changed, 675 insertions(+), 58 deletions(-) create mode 100644 drivers/clk/at91/clk-audio-pll.c base-commit: 22bbe310b60b37b029659460843265f190364e48 -- git-series 0.9.1

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[alsa-devel] [PATCH 2/2] ASoC: cs43130: Add devicetree bindings for CS43130
by Li Xu 10 Aug '17

10 Aug '17

Add devicetree bindings documentation file for Cirrus Logic CS43130 codec. Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- .../devicetree/bindings/sound/cs43130.txt | 72 ++++++++++++++++++++++ 1 file changed, 72 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs43130.txt diff --git a/Documentation/devicetree/bindings/sound/cs43130.txt b/Documentation/devicetree/bindings/sound/cs43130.txt new file mode 100644 index 0000000..51ea38a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs43130.txt @@ -0,0 +1,72 @@ +CS43130 DAC + +Required properties: + + - compatible : "cirrus,cs43130" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply, VL-supply, VCP-supply, VD-supply: + power supplies for the device, as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + +Optional properties: + + - reset-gpios : Active low GPIO used to reset the device + + - cirrus,xtal-ibias: + When external MCLK is generated by external crystal + oscillator, CS43130 can be used to provide bias current + for external crystal. Amount of bias current sent is + set as: + 1 = 7.5uA + 2 = 12.5uA + 3 = 15uA + + - cirrus,dc-measure: + Boolean, define to enable headphone DC impedance measurement. + + - cirrus,ac-measure: + Boolean, define to enable headphone AC impedance measurement. + DC impedance must also be enabled for AC impedance measurement. + + - cirrus,dc-threshold: + Define 2 DC impedance thresholds in ohms for HP output control. + Default values are 50 and 120 Ohms. + + - cirrus,ac-freq: + Define the frequencies at which to measure HP AC impedance. + Only used if "cirrus,dc-measure" is defined. + Exactly 10 frequencies must be defined. + If this properties is undefined, by default, + following frequencies are used: + <24 43 93 200 431 928 2000 4309 9283 20000> + The above frequencies are logarithmically equally spaced. + Log base is 10. + + - cirrus,idle-bias-off: + Boolean, define this flag such that the device is only powered on when + playing audio. Otherwise, the device is turned off. + If this flag is defined, HP detection and impedance measurement is disabled. + +Example: + +cs43130: audio-codec@30 { + compatible = "cirrus,cs43130"; + reg = <0x30>; + reset-gpios = <&axi_gpio 54 0>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + VL-supply = <&dummy_vreg>; + VCP-supply = <&dummy_vreg>; + VD-supply = <&dummy_vreg>; + cirrus,xtal-ibias = <2>; + interrupt-parent = <&gpio0>; + interrupts = <55 8>; + cirrus,idle-bias-off; + cirrus,dc-measure; + cirrus,ac-measure; + cirrus,dc-threshold = /bits/ 16 <20 100>; + cirrus,ac-freq = /bits/ 16 <24 43 93 200 431 928 2000 4309 9283 20000>; +}; -- 1.9.1

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