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

[PATCH] ASoC: nau8824: Fix open coded prefix handling
by Mark Brown 30 Jul '21

30 Jul '21

As with the component layer code the nau8824 driver had been doing some open coded pin manipulation which will have been broken now the core is fixed to handle this properly, remove the open coding to avoid the issue. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/nau8824.c | 42 ++++++-------------------------------- 1 file changed, 6 insertions(+), 36 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 15bd8335f667..906cdd2fb54a 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -828,36 +828,6 @@ static void nau8824_int_status_clear_all(struct regmap *regmap) } } -static void nau8824_dapm_disable_pin(struct nau8824 *nau8824, const char *pin) -{ - struct snd_soc_dapm_context *dapm = nau8824->dapm; - const char *prefix = dapm->component->name_prefix; - char prefixed_pin[80]; - - if (prefix) { - snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", - prefix, pin); - snd_soc_dapm_disable_pin(dapm, prefixed_pin); - } else { - snd_soc_dapm_disable_pin(dapm, pin); - } -} - -static void nau8824_dapm_enable_pin(struct nau8824 *nau8824, const char *pin) -{ - struct snd_soc_dapm_context *dapm = nau8824->dapm; - const char *prefix = dapm->component->name_prefix; - char prefixed_pin[80]; - - if (prefix) { - snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", - prefix, pin); - snd_soc_dapm_force_enable_pin(dapm, prefixed_pin); - } else { - snd_soc_dapm_force_enable_pin(dapm, pin); - } -} - static void nau8824_eject_jack(struct nau8824 *nau8824) { struct snd_soc_dapm_context *dapm = nau8824->dapm; @@ -866,8 +836,8 @@ static void nau8824_eject_jack(struct nau8824 *nau8824) /* Clear all interruption status */ nau8824_int_status_clear_all(regmap); - nau8824_dapm_disable_pin(nau8824, "SAR"); - nau8824_dapm_disable_pin(nau8824, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); snd_soc_dapm_sync(dapm); /* Enable the insertion interruption, disable the ejection @@ -897,8 +867,8 @@ static void nau8824_jdet_work(struct work_struct *work) struct regmap *regmap = nau8824->regmap; int adc_value, event = 0, event_mask = 0; - nau8824_dapm_enable_pin(nau8824, "MICBIAS"); - nau8824_dapm_enable_pin(nau8824, "SAR"); + snd_soc_dapm_enable_pin(dapm. "MICBIAS"); + snd_soc_dapm_enable_pin(dapm, "SAR"); snd_soc_dapm_sync(dapm); msleep(100); @@ -909,8 +879,8 @@ static void nau8824_jdet_work(struct work_struct *work) if (adc_value < HEADSET_SARADC_THD) { event |= SND_JACK_HEADPHONE; - nau8824_dapm_disable_pin(nau8824, "SAR"); - nau8824_dapm_disable_pin(nau8824, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); snd_soc_dapm_sync(dapm); } else { event |= SND_JACK_HEADSET; -- 2.20.1

1 1

[PATCH] ALSA: pcm - fix mmap capability check for the snd-dummy driver
by Jaroslav Kysela 30 Jul '21

30 Jul '21

The snd-dummy driver (fake_buffer configuration) uses the ops->page callback for the mmap operations. Allow mmap for this case, too. Cc: <stable(a)vger.kernel.org> Fixes: c4824ae7db41 ("ALSA: pcm: Fix mmap capability check") Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 6a2971a7e6a1..09c0e2a6489c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -246,7 +246,7 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) return false; - if (substream->ops->mmap) + if (substream->ops->mmap || substream->ops->page) return true; switch (substream->dma_buffer.dev.type) { -- 2.31.1

2 1

[PATCH v4 0/2] Cirrus Logic CS35L41 Amplifier
by David Rhodes 30 Jul '21

30 Jul '21

ASoC driver and devicetree documentation for a new Cirrus Logic amplifier CS35L41 v4 changes: Fixed spacing for bracketed declarations Fixed comment style in headers Moved private struct from include/sound to local cs35l41.h Reordered probe operations: 1) hardware errata 2) INT pin config 3) INT mask config 4) IRQ registration Moved set_pdata from component_probe to probe Set initial state of AMP_EN bit and corresponding DAPM widget to off Rename invert-pcm DT prop to invert-class-d Change return code handling in IRQ Use constants for regulator num supplies Error message for no SPI dev in spi_otp_setup Restore 0 initializers in IRQ per Charles's comments David Rhodes (2): ASoC: cs35l41: CS35L41 Boosted Smart Amplifier ASoC: cs35l41: Add bindings for CS35L41 .../devicetree/bindings/sound/cs35l41.yaml | 398 ++++ include/sound/cs35l41.h | 60 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs35l41-i2c.c | 116 ++ sound/soc/codecs/cs35l41-spi.c | 145 ++ sound/soc/codecs/cs35l41-tables.c | 612 ++++++ sound/soc/codecs/cs35l41.c | 1775 +++++++++++++++++ sound/soc/codecs/cs35l41.h | 777 ++++++++ 9 files changed, 3899 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l41.yaml create mode 100644 include/sound/cs35l41.h create mode 100644 sound/soc/codecs/cs35l41-i2c.c create mode 100644 sound/soc/codecs/cs35l41-spi.c create mode 100644 sound/soc/codecs/cs35l41-tables.c create mode 100644 sound/soc/codecs/cs35l41.c create mode 100644 sound/soc/codecs/cs35l41.h -- 2.25.1

3 8

[PATCH v1] ASoC: Intel: kbl_da7219_max98357a: fix drv_name
by Lukasz Majczak 30 Jul '21

30 Jul '21

platform_id for kbl_da7219_max98357a was shrunk for kbl_da7219_mx98357a, but the drv_name was changed for kbl_da7219_max98373. Tested on a Pixelbook (Atlas). Fixes: 94efd726b947 ("ASoC: Intel: kbl_da7219_max98357a: shrink platform_id below 20 characters") Cc: <stable(a)vger.kernel.org> # 5.4+ Reported-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> Signed-off-by: Lukasz Majczak <lma(a)semihalf.com> --- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index ba5ff468c265..8cab91a00b1a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -87,7 +87,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { }, { .id = "DLGS7219", - .drv_name = "kbl_da7219_max98357a", + .drv_name = "kbl_da7219_mx98357a", .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_7219_98357_codecs, @@ -113,7 +113,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { }, { .id = "DLGS7219", - .drv_name = "kbl_da7219_mx98373", + .drv_name = "kbl_da7219_max98373", .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_7219_98373_codecs, -- 2.32.0.554.ge1b32706d8-goog

3 4

5.14 regression, Intel SST (BYT) audio no longer works
by Hans de Goede 30 Jul '21

30 Jul '21

Hi All, I just noticed and I won't have time to dig any deeper for approx. the coming 10 days. Still I wanted to report this in case anyone has any ideas. Or maybe you can reproduce and look for a fix ? After building + installing 5.14-rc1 on a HP Elitepad 1000 G2 (Bay Trail) with a RT5642 codec, I noticed that sound over the speakers and over the docks line-out jack no longer works. Downgrading to 5.13 (without any other changes) fixes this, so this seems to be a regression with 5.14. I've no been able to make time to test this on other BYT/CHT hardware, but I suspect that other models will be affected too. Regards, Hans

2 3

[PATCH v2 00/16] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 30 Jul '21

30 Jul '21

This patchset adds ASoC driver support to configure signal processing framework ("AudioReach") which is integral part of Qualcomm next generation audio SDK and will be deployed on upcoming Qualcomm chipsets. It makes use of ASoC Topology to load graphs on to the DSP which is then managed by APM (Audio Processing Manager) service to prepare/start/stop. Here is simpified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +---------+ | | | q6apm | | q6apm | | q6afe | | | | dais | <------> | | <-----> | bedais | | | +---------+ +---------+ +---------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ | |____________________________|______________________________| | | RPMSG (IPC over GLINK) ____________________________|______________________________ | | | | +-----------------------+ | | | | | | v v q6 (Audio DSP) | |+-----+ +----------------------------------+ | || PRM | | APM (Audio Processing Manager) | | |+-----+ | . Graph Management | | | | . Command Handing | | | | . Event Management | | | | ... | | | +----------------------------------+ | | ^ | |____________________________|______________________________| | | LPASS AIF ____________________________|______________________________ | | Audio I/O | | v | | +--------------------------------------------------+ | | | Audio devices | | | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | | | | | +--------------------------------------------------+ | |___________________________________________________________| AudioReach has constructs of sub-graph, container and modules. Each sub-graph can have N containers and each Container can have N Modules and connections between them can be linear or non-linear. An audio function can be realized with one or many connected sub-graphs. There are also control/event paths between modules that can be wired up while building graph to achieve various control mechanism between modules. These concepts of Sub-Graph, Containers and Modules are represented in ASoC topology. Here is simple I2S graph with a Write Shared Memory and a Volume control module within a single Subgraph (1) with one Container (1) and 5 modules. ____________________________________________________________ | Sub-Graph [1] | | _______________________________________________________ | | | Container [1] | | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]-> [I2S-EP]| | | |_______________________________________________________| | |____________________________________________________________| For now this graph is split into two subgraphs to achieve dpcm like below: ________________________________________________ _________________ | Sub-Graph [1] | | Sub-Graph [2] | | ____________________________________________ | | _____________ | | | Container [1] | | | |Container [2]| | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]| | | | [I2S-EP] | | | |____________________________________________| | | |_____________| | |________________________________________________| |_________________| _________________ | Sub-Graph [3] | | _____________ | | |Container [3]| | | | [DMA-EP] | | | |_____________| | |_________________| This patchset adds very minimal support for AudioReach which includes supporting sub-graphs containing CODEC DMA ports and simple PCM Decoder/Encoder and Logger Modules. Additional capabilities will be built over time to expose features offered by AudioReach. This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5 and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via VA Macro and WCD938x Codec connected via TX and RX Macro. Sample WIP ASoC graphs are available at https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/ Thanks, srini Changes since RFC: - moved GPR support into APR driver to avoid code duplication. - Fixed various dt-bindings-check warnings and errors. - Moved include/dt-bindings headers to dt-bindings patch. - added PGA widget support - Fixed few ordering issues while testing with ASoC Topology 2.0 - Removed duplicate defines in various headers Srinivas Kandagatla (16): soc: dt-bindings: qcom: add gpr bindings soc: qcom: apr: make code more reuseable soc: qcom: apr: Add GPR support ASoC: qcom: dt-bindings: add bindings Audio Processing manager ASoC: qcom: audioreach: add basic pkt alloc support ASoC: qcom: audioreach: add q6apm support ASoC: qcom: audioreach: add module configuration command helpers ASoC: qcom: audioreach: add topology support ASoC: qcom: audioreach: add q6apm-dai support ASoC: qcom: audioreach: add bedai support ASoC: qcom: dt-bindings: add bindings for Proxy Resource Manager ASoC: qcom: audioreach: add q6prm support ASoC: qcom: dt-bindings: add audioreach soundcard compatibles ASoC: qcom: audioreach: add volume ctrl module support ASoC: qcom: audioreach: topology add dapm pga support ASoC: qcom: sm8250: Add audioreach support .../bindings/soc/qcom/qcom,gpr.yaml | 83 ++ .../devicetree/bindings/sound/qcom,q6apm.yaml | 87 ++ .../devicetree/bindings/sound/qcom,q6prm.yaml | 43 + .../bindings/sound/qcom,sm8250.yaml | 43 + drivers/soc/qcom/Kconfig | 8 + drivers/soc/qcom/apr.c | 270 +++- include/dt-bindings/soc/qcom,gpr.h | 18 + include/dt-bindings/sound/qcom,q6apm.h | 8 + include/linux/soc/qcom/apr.h | 69 +- include/uapi/sound/snd_ar_tokens.h | 198 +++ sound/soc/qcom/Kconfig | 19 + sound/soc/qcom/Makefile | 1 + sound/soc/qcom/audioreach/Makefile | 12 + sound/soc/qcom/audioreach/audioreach.c | 1162 +++++++++++++++++ sound/soc/qcom/audioreach/audioreach.h | 676 ++++++++++ sound/soc/qcom/audioreach/q6apm-bedai.c | 323 +++++ sound/soc/qcom/audioreach/q6apm-dai.c | 494 +++++++ sound/soc/qcom/audioreach/q6apm.c | 963 ++++++++++++++ sound/soc/qcom/audioreach/q6apm.h | 174 +++ sound/soc/qcom/audioreach/q6prm.c | 414 ++++++ sound/soc/qcom/audioreach/topology.c | 1075 +++++++++++++++ sound/soc/qcom/sm8250.c | 144 +- 22 files changed, 6236 insertions(+), 48 deletions(-) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,gpr.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6prm.yaml create mode 100644 include/dt-bindings/soc/qcom,gpr.h create mode 100644 include/dt-bindings/sound/qcom,q6apm.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/qcom/audioreach/Makefile create mode 100644 sound/soc/qcom/audioreach/audioreach.c create mode 100644 sound/soc/qcom/audioreach/audioreach.h create mode 100644 sound/soc/qcom/audioreach/q6apm-bedai.c create mode 100644 sound/soc/qcom/audioreach/q6apm-dai.c create mode 100644 sound/soc/qcom/audioreach/q6apm.c create mode 100644 sound/soc/qcom/audioreach/q6apm.h create mode 100644 sound/soc/qcom/audioreach/q6prm.c create mode 100644 sound/soc/qcom/audioreach/topology.c -- 2.21.0

5 39

[PATCH] ALSA: pcm: Fix mmap capability check
by Takashi Iwai 30 Jul '21

30 Jul '21

The hw_support_mmap() doesn't cover all memory allocation types and might use a wrong device pointer for checking the capability. Check the all memory allocation types more completely. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index c88c4316c417..6919d2943b9d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -246,12 +246,18 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) return false; - if (substream->ops->mmap || - (substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV && - substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV_UC)) + if (substream->ops->mmap) return true; - return dma_can_mmap(substream->dma_buffer.dev.dev); + switch (substream->dma_buffer.dev.type) { + case SNDRV_DMA_TYPE_UNKNOWN: + return false; + case SNDRV_DMA_TYPE_CONTINUOUS: + case SNDRV_DMA_TYPE_VMALLOC: + return true; + default: + return dma_can_mmap(substream->dma_buffer.dev.dev); + } } static int constrain_mask_params(struct snd_pcm_substream *substream, -- 2.26.2

2 2

[PATCH 1/3] ASoC: cs42l42: Correct definition of ADC Volume control
by Richard Fitzgerald 29 Jul '21

29 Jul '21

The ADC volume is a signed 8-bit number with range -97 to +12, with -97 being mute. Use a SOC_SINGLE_S8_TLV() to define this and fix the DECLARE_TLV_DB_SCALE() to have the correct start and mute flag. Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index eff013f295be..2fd20511f246 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -405,7 +405,7 @@ static const struct regmap_config cs42l42_regmap = { .use_single_write = true, }; -static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false); +static DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 100, true); static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); static const char * const cs42l42_hpf_freq_text[] = { @@ -444,8 +444,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_ADC_INV_SHIFT, true, false), SOC_SINGLE("ADC Boost Switch", CS42L42_ADC_CTL, CS42L42_ADC_DIG_BOOST_SHIFT, true, false), - SOC_SINGLE_SX_TLV("ADC Volume", CS42L42_ADC_VOLUME, - CS42L42_ADC_VOL_SHIFT, 0xA0, 0x6C, adc_tlv), + SOC_SINGLE_S8_TLV("ADC Volume", CS42L42_ADC_VOLUME, -97, 12, adc_tlv), SOC_SINGLE("ADC WNF Switch", CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_WNF_EN_SHIFT, true, false), SOC_SINGLE("ADC HPF Switch", CS42L42_ADC_WNF_HPF_CTL, -- 2.11.0

2 4

[PATCH] ALSA: usb-audio: Avoid unnecessary or invalid connector selection at resume
by Takashi Iwai 29 Jul '21

29 Jul '21

The recent fix for the resume on Lenovo machines seems causing a regression on others. It's because the change always triggers the connector selection no matter which widget node type is. This patch addresses the regression by setting the resume callback selectively only for the connector widget. Fixes: 44609fc01f28 ("ALSA: usb-audio: Check connector value on resume") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213897 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 35 ++++++++++++++++++++--------------- 1 file changed, 20 insertions(+), 15 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index f4cdaf1ba44a..9b713b4a5ec4 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1816,6 +1816,15 @@ static void get_connector_control_name(struct usb_mixer_interface *mixer, strlcat(name, " - Output Jack", name_size); } +/* get connector value to "wake up" the USB audio */ +static int connector_mixer_resume(struct usb_mixer_elem_list *list) +{ + struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); + + get_connector_value(cval, NULL, NULL); + return 0; +} + /* Build a mixer control for a UAC connector control (jack-detect) */ static void build_connector_control(struct usb_mixer_interface *mixer, const struct usbmix_name_map *imap, @@ -1833,6 +1842,10 @@ static void build_connector_control(struct usb_mixer_interface *mixer, if (!cval) return; snd_usb_mixer_elem_init_std(&cval->head, mixer, term->id); + + /* set up a specific resume callback */ + cval->head.resume = connector_mixer_resume; + /* * UAC2: The first byte from reading the UAC2_TE_CONNECTOR control returns the * number of channels connected. @@ -3642,23 +3655,15 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) return 0; } -static int default_mixer_resume(struct usb_mixer_elem_list *list) -{ - struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); - - /* get connector value to "wake up" the USB audio */ - if (cval->val_type == USB_MIXER_BOOLEAN && cval->channels == 1) - get_connector_value(cval, NULL, NULL); - - return 0; -} - static int default_mixer_reset_resume(struct usb_mixer_elem_list *list) { - int err = default_mixer_resume(list); + int err; - if (err < 0) - return err; + if (list->resume) { + err = list->resume(list); + if (err < 0) + return err; + } return restore_mixer_value(list); } @@ -3697,7 +3702,7 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, list->id = unitid; list->dump = snd_usb_mixer_dump_cval; #ifdef CONFIG_PM - list->resume = default_mixer_resume; + list->resume = NULL; list->reset_resume = default_mixer_reset_resume; #endif } -- 2.26.2

1 0

[PATCH] ASoC: samsung: Constify static snd_soc_ops
by Rikard Falkeborn 29 Jul '21

29 Jul '21

These are only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make them const to allow the compiler to put them in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/samsung/aries_wm8994.c | 2 +- sound/soc/samsung/arndale.c | 4 ++-- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/littlemill.c | 2 +- sound/soc/samsung/midas_wm1811.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 4 ++-- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/smartq_wm8987.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 2 +- sound/soc/samsung/tm2_wm5110.c | 6 +++--- sound/soc/samsung/tobermory.c | 2 +- 13 files changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c index 0ac5956ba270..313ab650f8d9 100644 --- a/sound/soc/samsung/aries_wm8994.c +++ b/sound/soc/samsung/aries_wm8994.c @@ -310,7 +310,7 @@ static int aries_hw_free(struct snd_pcm_substream *substream) /* * Main DAI operations */ -static struct snd_soc_ops aries_ops = { +static const struct snd_soc_ops aries_ops = { .hw_params = aries_hw_params, .hw_free = aries_hw_free, }; diff --git a/sound/soc/samsung/arndale.c b/sound/soc/samsung/arndale.c index 28587375813a..606ac5e33a8e 100644 --- a/sound/soc/samsung/arndale.c +++ b/sound/soc/samsung/arndale.c @@ -48,7 +48,7 @@ static int arndale_rt5631_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops arndale_rt5631_ops = { +static const struct snd_soc_ops arndale_rt5631_ops = { .hw_params = arndale_rt5631_hw_params, }; @@ -80,7 +80,7 @@ static int arndale_wm1811_hw_params(struct snd_pcm_substream *substream, rclk + 1, SND_SOC_CLOCK_IN); } -static struct snd_soc_ops arndale_wm1811_ops = { +static const struct snd_soc_ops arndale_wm1811_ops = { .hw_params = arndale_wm1811_hw_params, }; diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 8aa78ff640f5..c994e67d1eaf 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -112,7 +112,7 @@ static int h1940_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops h1940_ops = { +static const struct snd_soc_ops h1940_ops = { .startup = h1940_startup, .hw_params = h1940_hw_params, }; diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index a1ff1400857e..390f2dd735ad 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -130,7 +130,7 @@ static int littlemill_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops littlemill_ops = { +static const struct snd_soc_ops littlemill_ops = { .hw_params = littlemill_hw_params, }; diff --git a/sound/soc/samsung/midas_wm1811.c b/sound/soc/samsung/midas_wm1811.c index 1f9a553edf19..a2019535a0b1 100644 --- a/sound/soc/samsung/midas_wm1811.c +++ b/sound/soc/samsung/midas_wm1811.c @@ -129,7 +129,7 @@ static int midas_aif1_hw_params(struct snd_pcm_substream *substream, return midas_start_fll1(rtd, pll_out); } -static struct snd_soc_ops midas_aif1_ops = { +static const struct snd_soc_ops midas_aif1_ops = { .hw_params = midas_aif1_hw_params, }; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 9266070e0181..c98b68567a89 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -106,7 +106,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) /* * Neo1973 WM8753 HiFi DAI opserations. */ -static struct snd_soc_ops neo1973_hifi_ops = { +static const struct snd_soc_ops neo1973_hifi_ops = { .hw_params = neo1973_hifi_hw_params, .hw_free = neo1973_hifi_hw_free, }; @@ -158,7 +158,7 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); } -static struct snd_soc_ops neo1973_voice_ops = { +static const struct snd_soc_ops neo1973_voice_ops = { .hw_params = neo1973_voice_hw_params, .hw_free = neo1973_voice_hw_free, }; diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 400a7f77c711..6ea1c8cc9167 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -62,7 +62,7 @@ static struct snd_soc_jack_gpio hp_jack_gpios[] = { }, }; -static struct snd_soc_ops rx1950_ops = { +static const struct snd_soc_ops rx1950_ops = { .startup = rx1950_startup, .hw_params = rx1950_hw_params, }; diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index c95629becbc3..cee39ad16667 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -70,7 +70,7 @@ static int smartq_hifi_hw_params(struct snd_pcm_substream *substream, /* * SmartQ WM8987 HiFi DAI operations. */ -static struct snd_soc_ops smartq_hifi_ops = { +static const struct snd_soc_ops smartq_hifi_ops = { .hw_params = smartq_hifi_hw_params, }; diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index ed753a2f202e..78703d095a6f 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -86,7 +86,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, /* * SMDK WM8580 DAI operations. */ -static struct snd_soc_ops smdk_ops = { +static const struct snd_soc_ops smdk_ops = { .hw_params = smdk_hw_params, }; diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 39a7a449f554..7661b637946d 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -73,7 +73,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, /* * SMDK WM8994 DAI operations. */ -static struct snd_soc_ops smdk_ops = { +static const struct snd_soc_ops smdk_ops = { .hw_params = smdk_hw_params, }; diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index a01640576f71..029448f5bedb 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -85,7 +85,7 @@ static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops smdk_wm8994_pcm_ops = { +static const struct snd_soc_ops smdk_wm8994_pcm_ops = { .hw_params = smdk_wm8994_pcm_hw_params, }; diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 84c2c63d5a87..a2c77e6defec 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -126,7 +126,7 @@ static int tm2_aif1_hw_params(struct snd_pcm_substream *substream, return tm2_start_sysclk(rtd->card); } -static struct snd_soc_ops tm2_aif1_ops = { +static const struct snd_soc_ops tm2_aif1_ops = { .hw_params = tm2_aif1_hw_params, }; @@ -200,7 +200,7 @@ static int tm2_aif2_hw_free(struct snd_pcm_substream *substream) return ret; } -static struct snd_soc_ops tm2_aif2_ops = { +static const struct snd_soc_ops tm2_aif2_ops = { .hw_params = tm2_aif2_hw_params, .hw_free = tm2_aif2_hw_free, }; @@ -254,7 +254,7 @@ static int tm2_hdmi_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tm2_hdmi_ops = { +static const struct snd_soc_ops tm2_hdmi_ops = { .hw_params = tm2_hdmi_hw_params, }; diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index c962d2c2a7f7..15223d860cb7 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -105,7 +105,7 @@ static int tobermory_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tobermory_ops = { +static const struct snd_soc_ops tobermory_ops = { .hw_params = tobermory_hw_params, }; -- 2.32.0

3 2