May 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH v7 0/2] Add generic serial MIDI driver using serial bus API
by Daniel Kaehn 12 May '22

12 May '22

Generic serial MIDI driver adding support for using serial devices compatible with the serial bus as raw MIDI devices, allowing using additional serial devices not compatible with the existing serial-u16550 driver. Supports only setting standard serial baudrates on the underlying serial device; however, the underlying serial device can be configured so that a requested 38.4 kBaud is actually the standard MIDI 31.25 kBaud. Supports DeviceTree configuration. Changes in v7: - Separate examples in dt-binding to remove need to specify unit name (fixing dt_binding_check error) Changes in v6: - Change compatible "serialmidi" -> serial-midi" - Default current-speed to 38400 (closest baud to MIDI standard speed) - Appropriately stop reading or writing MIDI if input or output _trigger() is called with a parameter of zero, respectively - Zero out corresponding triggered state on close to ensure input and output closing results in the serial port being closed - Fix order of operations in _probe() - Remove "DEBUG" literal from debug messages - Remove unused dt-parsing patch checking for existence of node - Whitespace / tabbing fixes / improvements Changes in v5: - Reword description in dt-binding for clarity - Change 'speed' dt property to standard 'current-speed' - Move MIDI output loop onto workqueue (since this could loop quite a while, if ALSA provides a continuous stream of bytes) - Add tx_state bit flags to snd_serial_generic struct - Safegard critical section in tx_work with atomic bit ops on tx_state - Switch operations on filemode to use atomic bit ops Changes in v4: - Fix regressed typo - Correct 3.84 kBaud -> 38.4 kBaud in DT & Kconfig (sorry about spam - noticed after sending v3 and didn't want to let the error sit around for too long) Changes in v3: - Replace use of snd_printk() with dev_* alternatives - Removed unnecessary initialization of err variables - Replaced instances of `== SERIAL_MODE_NOT_OPENED` with zero check - Loop on output_write to completely fill output buffer if data available - Depend on CONFIG_OF in Kconfig - Replace use of devm_kzalloc() with extra_size allocation in snd_devm_card_new() - Use module_serdev_device_driver() instead of module_init() and module_exit(0) Changes in v2: - Fix 'snd_serial_generic_write_wakeup' missing static keyword - Correct 3.125 kBaud > 31.25 kBaud in documentation for MIDI The need for this driver arose from a project using a Raspberry Pi4 which needed to receive and send raw MIDI with low latency. The pl011 UART used is not compatible with the existing serial MIDI driver made for u16550-style devices. Using a userspace program such as ttymidi to feed input from the TTY device to a virtual ALSA MIDI device was functional, but not ideal. I am not sure if a MIDI driver needing the mentioned 'hack' to clock 38.4 kBaud down to the standard MIDI baud is permissible in the mainline kernel, but am submitting nevertheless in case it is useful. To my knowledge, it doesn't seem that there would be any way for this driver to manually configure a serial port to 31.25 kBaud using the serial bus API (please correct me f I'm wrong). In my use case, I am actually configuring one port to run at 115.2 kBaud for faster communication with a custom onboard MIDI controller. Daniel Kaehn (2): dt-bindings: sound: Add generic serial MIDI device Add generic serial MIDI driver using serial bus API .../bindings/sound/serial-midi.yaml | 51 +++ sound/drivers/Kconfig | 18 + sound/drivers/Makefile | 2 + sound/drivers/serial-generic.c | 374 ++++++++++++++++++ 4 files changed, 445 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/serial-midi.yaml create mode 100644 sound/drivers/serial-generic.c base-commit: c5eb0a61238dd6faf37f58c9ce61c9980aaffd7a -- 2.33.0

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[PATCH v2 0/3] ALSA: hda/cs8409: Add support for Odin Laptop Variants
by Stefan Binding 12 May '22

12 May '22

Add support for new Odin Laptop Variants into CS8409 HDA Driver. Since these laptops require Speaker Playback Switch, support must be added into CS8409 HDA driver. Since CS8409 does not have amplifier parameters for the NID associated with the speaker, the HDA driver does not add the switch automatically, so the driver needs to add this support manually. changes since v1: - add missing break in switch statement Stefan Binding (3): ALSA: hda/cs8409: Support new Odin Variants ALSA: hda/cs8409: Add Speaker Playback Switch for Cyborg ALSA: hda/cs8409: Add Speaker Playback Switch for Warlock sound/pci/hda/patch_cs8409-tables.c | 19 +++++++ sound/pci/hda/patch_cs8409.c | 86 ++++++++++++++++++++++++----- sound/pci/hda/patch_cs8409.h | 5 ++ 3 files changed, 96 insertions(+), 14 deletions(-) -- 2.25.1

2 4

[PATCH -next] ALSA: ppc: tumbler: use for_each_child_of_node() macro
by Yang Yingliang 12 May '22

12 May '22

Use for_each_child_of_node() macro instead of open coding it. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/ppc/tumbler.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index c65e74d7cd0a..c7e150f82bcc 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -1060,8 +1060,7 @@ static struct device_node *find_audio_device(const char *name) if (! gpiop) return NULL; - for (np = of_get_next_child(gpiop, NULL); np; - np = of_get_next_child(gpiop, np)) { + for_each_child_of_node(gpiop, np) { const char *property = of_get_property(np, "audio-gpio", NULL); if (property && strcmp(property, name) == 0) break; -- 2.25.1

1 0

[PATCH] ASoC: ADAU7118: add bindings for managing pins drive strength
by Dylan Laduranty 12 May '22

12 May '22

This allows users to change SDATA and both PDM clocks pins drive strength during device probing according to their need. Update yaml documentation accordingly. Signed-off-by: Dylan Laduranty <dylan.laduranty(a)mesotic.com> --- .../bindings/sound/adi,adau7118.yaml | 24 +++++++ sound/soc/codecs/adau7118.c | 62 +++++++++++++++++-- 2 files changed, 82 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml index fb78967ee17b..71e8a9ff2edf 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml +++ b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml @@ -51,6 +51,27 @@ properties: maximum: 1 default: [0, 0, 1, 1] + adi,pdm-clk0-ds: + description: | + This property set the drive strength of PDM CLK0 output pad. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [3, 2, 1, 0] + default: 2 + + adi,pdm-clk1-ds: + description: | + This property set the drive strength of PDM CLK1 output pad. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [3, 2, 1, 0] + default: 2 + + adi,sdata-ds: + description: | + This property set the drive strength of SDATA output pad. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [3, 2, 1, 0] + default: 2 + required: - "#sound-dai-cells" - compatible @@ -73,6 +94,9 @@ examples: dvdd-supply = <&supply>; adi,pdm-clk-map = <1 1 0 0>; adi,decimation-ratio = <16>; + adi,pdm-clk0-ds = <3>; + adi,pdm-clk1-ds = <3>; + adi,sdata-ds = <3>; }; }; diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c index 841229dcbca1..18c1f246f911 100644 --- a/sound/soc/codecs/adau7118.c +++ b/sound/soc/codecs/adau7118.c @@ -29,6 +29,12 @@ FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) #define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) #define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_DS_PDM_CLK0_MASK GENMASK(1, 0) +#define ADAU7118_DS_PDM_CLK0(x) FIELD_PREP(ADAU7118_DS_PDM_CLK0_MASK, x) +#define ADAU7118_DS_PDM_CLK1_MASK GENMASK(3, 2) +#define ADAU7118_DS_PDM_CLK1(x) FIELD_PREP(ADAU7118_DS_PDM_CLK1_MASK, x) +#define ADAU7118_DS_SDATA_MASK GENMASK(5, 4) +#define ADAU7118_DS_SDATA(x) FIELD_PREP(ADAU7118_DS_SDATA_MASK, x) #define ADAU7118_FULL_SOFT_R_MASK BIT(1) #define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) @@ -489,7 +495,7 @@ static int adau7118_regulator_setup(struct adau7118_data *st) static int adau7118_parset_dt(const struct adau7118_data *st) { int ret; - u32 dec_ratio = 0; + u32 val32 = 0; /* 4 inputs */ u32 clk_map[4], regval; @@ -497,9 +503,9 @@ static int adau7118_parset_dt(const struct adau7118_data *st) return 0; ret = device_property_read_u32(st->dev, "adi,decimation-ratio", - &dec_ratio); + &val32); if (!ret) { - switch (dec_ratio) { + switch (val32) { case 64: regval = ADAU7118_DEC_RATIO(0); break; @@ -510,7 +516,7 @@ static int adau7118_parset_dt(const struct adau7118_data *st) regval = ADAU7118_DEC_RATIO(2); break; default: - dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + dev_err(st->dev, "Invalid dec ratio: %u", val32); return -EINVAL; } @@ -537,6 +543,54 @@ static int adau7118_parset_dt(const struct adau7118_data *st) return ret; } + ret = device_property_read_u32(st->dev, "adi,pdm-clk0-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid pdm-clk0-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_PDM_CLK0_MASK, + ADAU7118_DS_PDM_CLK0(val32)); + if (ret) + return ret; + } + + ret = device_property_read_u32(st->dev, "adi,pdm-clk1-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid pdm-clk1-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_PDM_CLK1_MASK, + ADAU7118_DS_PDM_CLK1(val32)); + if (ret) + return ret; + } + + ret = device_property_read_u32(st->dev, "adi,sdata-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid sdata-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_SDATA_MASK, + ADAU7118_DS_SDATA(val32)); + if (ret) + return ret; + } + return 0; } -- 2.17.1

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[PATCH -next 0/2] ASoC: codecs: Fix build error
by Hui Tang 12 May '22

12 May '22

Fix two build error, as follows: tanghui (2): ASoC: max98396: Fix build error for implicit function declaration ASoC: tlv320adc3xxx: Fix build error for implicit function declaration sound/soc/codecs/Kconfig | 2 ++ 1 file changed, 2 insertions(+) -- 2.17.1

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Re: kismet: WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A when selected by SND_SOC_INTEL_SOF_CS42L42_MACH
by Pierre-Louis Bossart 11 May '22

11 May '22

On 5/10/22 22:36, kernel test robot wrote: > tree: > https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git master > head: feb9c5e19e913b53cb536a7aa7c9f20107bb51ec > commit: 5a7f27a624d9e33262767b328aa7a4baf7846c14 ASoC: Intel: add > sof-cs42l42 machine driver > date: 1 year ago > config: (attached as .config) > reproduce: > # > https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… > > git remote add linus > https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git > git fetch --no-tags linus master > git checkout 5a7f27a624d9e33262767b328aa7a4baf7846c14 > # 1. reproduce by kismet > # install kmax per https://github.com/paulgazz/kmax/README.md > kismet --linux-ksrc=linux --selectees > CONFIG_SND_SOC_MAX98357A --selectors > CONFIG_SND_SOC_INTEL_SOF_CS42L42_MACH -a=x86_64 > # 2. reproduce by make > # save the config file to linux source tree > cd linux > make ARCH=x86_64 olddefconfig > > If you fix the issue, kindly add following tag as appropriate > Reported-by: kernel test robot <yujie.liu(a)intel.com> > > > kismet warnings: (new ones prefixed by >>) >>> kismet: WARNING: unmet direct dependencies detected for > SND_SOC_MAX98357A when selected by SND_SOC_INTEL_SOF_CS42L42_MACH > > WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A > Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && > GPIOLIB [=n] > Selected by [y]: > - SND_SOC_INTEL_SOF_CS42L42_MACH [=y] && SOUND [=y] && !UML && SND > [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK > [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && > SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && > (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) I can reproduce this, but I question the hard dependency on GPIOLIB in the amplifier Kconfig: max98357a->sdmode = devm_gpiod_get_optional(&pdev->dev, "sdmode", GPIOD_OUT_LOW); If the gpio is optional, max98357a.c should build and compile without GPIOLIB, no? > WARNING: unmet direct dependencies detected for SND_SOC_DMIC > Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && > GPIOLIB [=n] > Selected by [y]: > - SND_SOC_INTEL_SOF_CS42L42_MACH [=y] && SOUND [=y] && !UML && SND > [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK > [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && > SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && > (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) I can't reproduce this one with Mark's for-next branch, this was fixed I believe.

1 1

[PATCH 1/2] ASoc: fsl_micfil: explicitly clear software reset bit
by Shengjiu Wang 11 May '22

11 May '22

SRES is self-cleared bit, but REG_MICFIL_CTRL1 is defined as non volatile register, it still remain in regmap cache after set, then every update of REG_MICFIL_CTRL1, software reset happens. to avoid this, clear it explicitly. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_micfil.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index cd85c8714f97..2149fac0dcc6 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -179,6 +179,17 @@ static int fsl_micfil_reset(struct device *dev) if (ret) return ret; + /* + * SRES is self-cleared bit, but REG_MICFIL_CTRL1 is defined + * as non-volatile register, so SRES still remain in regmap + * cache after set, that every update of REG_MICFIL_CTRL1, + * software reset happens. so clear it explicitly. + */ + ret = regmap_clear_bits(micfil->regmap, REG_MICFIL_CTRL1, + MICFIL_CTRL1_SRES); + if (ret) + return ret; + return 0; } -- 2.17.1

2 2

[PATCH 0/3] ASoC: Intel: boards: Chromebook configuration updates
by Pierre-Louis Bossart 11 May '22

11 May '22

two minor changes to enable DMIC and capture for CS35L41, and one new configuration for AlderLake hardware. Brent Lu (2): ASoC: Intel: sof_ssp_amp: fix no DMIC BE Link on Chromebooks ASoC: Intel: sof_ssp_amp: enable capture stream from cs35l41 Vamshi Krishna (1): ASoC: Intel: sof_rt5682: Add support for adl_rt1019_rt5682s sound/soc/intel/boards/sof_realtek_common.c | 39 +++++++++++++++++++ sound/soc/intel/boards/sof_realtek_common.h | 5 +++ sound/soc/intel/boards/sof_rt5682.c | 13 +++++++ sound/soc/intel/boards/sof_ssp_amp.c | 19 ++++++++- .../intel/common/soc-acpi-intel-adl-match.c | 7 ++++ 5 files changed, 81 insertions(+), 2 deletions(-) -- 2.30.2

2 4

[PATCH] [v2] ASoC: Intel: sof_cs42l42: adding support for ADL configuration and BT offload audio
by Terry Chen 11 May '22

11 May '22

To be able to do driver data for adl_mx98360a_cs4242 which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on ADL platform. Also add the capability to machine driver of creating DAI Link for BT offload. Although BT offload always uses SSP2 port but we reserve the flexibility to assign the port number in macro. Signed-off-by: Terry Chen <terry_chen(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/sof_cs42l42.c | 88 ++++++++++++++++++- .../intel/common/soc-acpi-intel-adl-match.c | 8 ++ 2 files changed, 92 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index ce78c18798876..4ac4f51c7c115 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -41,8 +41,13 @@ #define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) #define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(25) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_BT_OFFLOAD_PRESENT BIT(25) +#define SOF_CS42L42_SSP_BT_SHIFT 26 +#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) +#define SOF_CS42L42_SSP_BT(quirk) \ + (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(29) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(30) enum { LINK_NONE = 0, @@ -50,6 +55,7 @@ enum { LINK_SPK = 2, LINK_DMIC = 3, LINK_HDMI = 4, + LINK_BT = 5, }; /* Default: SSP2 */ @@ -278,6 +284,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int create_spk_amp_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -467,9 +480,52 @@ static int create_hdmi_dai_links(struct device *dev, return -ENOMEM; } +static int create_bt_offload_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_bt) +{ + int ret = 0; + + /* bt offload */ + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) + return 0; + + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ssp_bt); + if (!links[*id].name) + goto devm_err; + + links[*id].id = *id; + links[*id].codecs = dummy_component; + links[*id].num_codecs = ARRAY_SIZE(dummy_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_bt); + if (!links[*id].cpus->dai_name) + goto devm_err; + + (*id)++; + + return 0; + +devm_err: + return ret; +} + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, + int ssp_bt, int dmic_be_num, int hdmi_num) { @@ -522,6 +578,14 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; } break; + case LINK_BT: + ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); + if (ret < 0) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + break; case LINK_NONE: /* caught here if it's not used as terminator in macro */ default: @@ -543,7 +607,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; - int ret, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -568,6 +632,9 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + SOF_CS42L42_SSP_BT_SHIFT; + ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; @@ -578,9 +645,11 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_cs42l42.num_links++; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, - dmic_be_num, hdmi_num); + ssp_bt, dmic_be_num, hdmi_num); if (!dai_links) return -ENOMEM; @@ -621,6 +690,17 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1)) | SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), }, + { + .name = "adl_mx98360a_cs4242", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1) | + SOF_CS42L42_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_SSP_BT(2)) | + SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 7c8cd00457f81..3f40519250a90 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -384,6 +384,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-cs35l41.tplg", }, + { + .id = "10134242", + .drv_name = "adl_mx98360a_cs4242", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.31.0

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[v5] ASoC: Intel: sof_cs42l42: adding support for ADL configuration and BT offload audio
by Terry Chen 11 May '22

11 May '22

To be able to do driver data for adl_mx98360a_cs4242 which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on ADL platform. Also add the capability to machine driver of creating DAI Link for BT offload. Although BT offload always uses SSP2 port but we reserve the flexibility to assign the port number in macro. Signed-off-by: Terry Chen <terry_chen(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/sof_cs42l42.c | 92 ++++++++++++++++++- .../intel/common/soc-acpi-intel-adl-match.c | 7 ++ 2 files changed, 95 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index ce78c18798876..442f13de6cd55 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -41,8 +41,13 @@ #define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) #define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(25) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_BT_OFFLOAD_PRESENT BIT(25) +#define SOF_CS42L42_SSP_BT_SHIFT 26 +#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) +#define SOF_CS42L42_SSP_BT(quirk) \ + (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(29) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(30) enum { LINK_NONE = 0, @@ -50,6 +55,7 @@ enum { LINK_SPK = 2, LINK_DMIC = 3, LINK_HDMI = 4, + LINK_BT = 5, }; /* Default: SSP2 */ @@ -278,6 +284,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int create_spk_amp_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -467,9 +480,56 @@ static int create_hdmi_dai_links(struct device *dev, return -ENOMEM; } +static int create_bt_offload_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_bt) +{ + int ret = 0; + + /* bt offload */ + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) + return 0; + + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ssp_bt); + if (!links[*id].name) { + ret = -ENOMEM; + goto devm_err; + } + + links[*id].id = *id; + links[*id].codecs = dummy_component; + links[*id].num_codecs = ARRAY_SIZE(dummy_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_bt); + if (!links[*id].cpus->dai_name) { + ret = -ENOMEM; + goto devm_err; + } + + (*id)++; + + return 0; + +devm_err: + return -ENOMEM; +} + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, + int ssp_bt, int dmic_be_num, int hdmi_num) { @@ -522,6 +582,14 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; } break; + case LINK_BT: + ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); + if (ret < 0) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + break; case LINK_NONE: /* caught here if it's not used as terminator in macro */ default: @@ -543,7 +611,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; - int ret, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -568,6 +636,9 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + SOF_CS42L42_SSP_BT_SHIFT; + ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; @@ -578,9 +649,11 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_cs42l42.num_links++; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, - dmic_be_num, hdmi_num); + ssp_bt, dmic_be_num, hdmi_num); if (!dai_links) return -ENOMEM; @@ -621,6 +694,17 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1)) | SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), }, + { + .name = "adl_mx98360a_cs4242", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1) | + SOF_CS42L42_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_SSP_BT(2)) | + SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 7c8cd00457f81..311adeb3afbc4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -384,6 +384,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-cs35l41.tplg", }, + { + .id = "10134242", + .drv_name = "adl_mx98360a_cs4242", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.25.1

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