March 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/2] ASoC: codecs: ES8326: Reducin powerconsumption and
by Zhang Yi 20 Mar '24

20 Mar '24

We changed the configuration related to hibernation. and delete the REG_SUPPLY to cover mute issue. Zhang Yi (2): ASoC: codecs: ES8326: Reducing power consumption ASoC: codecs: ES8326: Delete unused REG_SUPPLY sound/soc/codecs/es8326.c | 42 ++++++++++++++++++++++++++++++--------- 1 file changed, 33 insertions(+), 9 deletions(-) -- 2.17.1

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[PATCH] ALSA: hda/realtek: Add quirks for some Clevo laptops
by Tim Crawford 20 Mar '24

20 Mar '24

Add audio quirks to fix speaker output and headset detection on some new Clevo models: - L240TU (ALC245) - PE60SNE-G (ALC1220) - V350SNEQ (ALC245) Co-authored-by: Jeremy Soller <jeremy(a)system76.com> Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d463d416fc23..25183996756c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2645,6 +2645,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x65f1, "Clevo PC50HS", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65f5, "Clevo PD50PN[NRT]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x66a2, "Clevo PE60RNE", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x66a6, "Clevo PE60SN[CDE]-[GS]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), @@ -10139,12 +10140,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x152d, 0x1082, "Quanta NL3", ALC269_FIXUP_LIFEBOOK), + SND_PCI_QUIRK(0x1558, 0x0353, "Clevo V35[05]SN[CDE]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1323, "Clevo N130ZU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1325, "Clevo N15[01][CW]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1401, "Clevo L140[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1403, "Clevo N140CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1404, "Clevo N150CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x2624, "Clevo L240TU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.44.0

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Re: [PATCH] Add Asus VivoBook S 16X to quirks table for acp6x
by Geraldo Nascimento 20 Mar '24

20 Mar '24

On Tue, Mar 19, 2024 at 05:16:10PM +0800, marcel wrote: > Hi again, > > it's been a while since i sent my patch and I still can't see it in > kernel on github, so there's my question because I'm not sure if I > understand correctly. Should I just wait a little bit more and it just > takes time or send patch again and change things you pointed? Also which > version should I type after [PATCH v...], "1" ? Because it's my first > attempt? Sorry for taking your time and if the questions are obvious, > but it's my first time doing something like this. Hi Marcel, please address my past concerns regarding your patch and resubmit as v2. Remember to Cc: Takashi Iwai and Jaroslav Kysela besides linux-sound mailing list or your patch will fall into limbo again, overlooked by the maintainers. There's also a get_maintainers.pl script in the linux tree. You might wanna use that as it adds some relevant people who might take a interest in your patch. Thanks, Geraldo Nascimento > > Greetings

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[PATCH v14 00/16] Add audio support in v4l2 framework
by Shengjiu Wang 19 Mar '24

19 Mar '24

Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. It is a specific use case, there is no reference in current kernel. v4l2 memory to memory is the closed implementation, v4l2 current support video, image, radio, tuner, touch devices, so it is not complicated to add support for this specific audio case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the first 3 patches is for refining the code to make it can be shared by the "memory->asrc->memory" driver. The main change is in the v4l2 side, A /dev/vl4-audioX will be created, user applications only use the ioctl of v4l2 framework. Other change is to add memory to memory support for two kinds of i.MX ASRC module. changes in v14: - document the reservation of 'AUXX' fourcc format. - add v4l2_audfmt_to_fourcc() definition. changes in v13 - change 'pixelformat' to 'audioformat' in dev-audio-mem2mem.rst - add more description for clock drift in ext-ctrls-audio-m2m.rst - Add "media: v4l2-ctrls: add support for fraction_bits" from Hans to avoid build issue for kernel test robot changes in v12 - minor changes according to comments - drop min_buffers_needed = 1 and V4L2_CTRL_FLAG_UPDATE flag - drop bus_info changes in v11 - add add-fixed-point-test-controls in vivid. - add v4l2_ctrl_fp_compose() helper function for min and max changes in v10 - remove FIXED_POINT type - change code base on media: v4l2-ctrls: add support for fraction_bits - fix issue reported by kernel test robot - remove module_alias changes in v9: - add MEDIA_ENT_F_PROC_AUDIO_RESAMPLER. - add MEDIA_INTF_T_V4L_AUDIO - add media controller support - refine the vim2m-audio to support 8k<->16k conversion. changes in v8: - refine V4L2_CAP_AUDIO_M2M to be 0x00000008 - update doc for FIXED_POINT - address comments for imx-asrc changes in v7: - add acked-by from Mark - separate commit for fixed point, m2m audio class, audio rate controls - use INTEGER_MENU for rate, FIXED_POINT for rate offset - remove used fmts - address other comments for Hans changes in v6: - use m2m_prepare/m2m_unprepare/m2m_start/m2m_stop to replace m2m_start_part_one/m2m_stop_part_one, m2m_start_part_two/m2m_stop_part_two. - change V4L2_CTRL_TYPE_ASRC_RATE to V4L2_CTRL_TYPE_FIXED_POINT - fix warning by kernel test rebot - remove some unused format V4L2_AUDIO_FMT_XX - Get SNDRV_PCM_FORMAT from V4L2_AUDIO_FMT in driver. - rename audm2m to viaudm2m. changes in v5: - remove V4L2_AUDIO_FMT_LPCM - define audio pixel format like V4L2_AUDIO_FMT_S8... - remove rate and format in struct v4l2_audio_format. - Add V4L2_CID_ASRC_SOURCE_RATE and V4L2_CID_ASRC_DEST_RATE controls - updata document accordingly. changes in v4: - update document style - separate V4L2_AUDIO_FMT_LPCM and V4L2_CAP_AUDIO_M2M in separate commit changes in v3: - Modify documents for adding audio m2m support - Add audio virtual m2m driver - Defined V4L2_AUDIO_FMT_LPCM format type for audio. - Defined V4L2_CAP_AUDIO_M2M capability type for audio m2m case. - with modification in v4l-utils, pass v4l2-compliance test. changes in v2: - decouple the implementation in v4l2 and ALSA - implement the memory to memory driver as a platfrom driver and move it to driver/media - move fsl_asrc_common.h to include/sound folder Hans Verkuil (1): media: v4l2-ctrls: add support for fraction_bits Shengjiu Wang (15): ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc: move fsl_asrc_common.h to include/sound ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device media: uapi: Add V4L2_CAP_AUDIO_M2M capability flag media: v4l2: Add audio capture and output support media: uapi: Define audio sample format fourcc type media: uapi: Add V4L2_CTRL_CLASS_M2M_AUDIO media: uapi: Add audio rate controls support media: uapi: Declare interface types for Audio media: uapi: Add an entity type for audio resampler media: vivid: add fixed point test controls media: imx-asrc: Add memory to memory driver media: vim2m-audio: add virtual driver for audio memory to memory .../media/mediactl/media-types.rst | 11 + .../userspace-api/media/v4l/buffer.rst | 6 + .../userspace-api/media/v4l/common.rst | 1 + .../media/v4l/dev-audio-mem2mem.rst | 71 + .../userspace-api/media/v4l/devices.rst | 1 + .../media/v4l/ext-ctrls-audio-m2m.rst | 59 + .../userspace-api/media/v4l/pixfmt-audio.rst | 100 ++ .../userspace-api/media/v4l/pixfmt.rst | 1 + .../media/v4l/vidioc-enum-fmt.rst | 2 + .../media/v4l/vidioc-g-ext-ctrls.rst | 4 + .../userspace-api/media/v4l/vidioc-g-fmt.rst | 4 + .../media/v4l/vidioc-querycap.rst | 3 + .../media/v4l/vidioc-queryctrl.rst | 11 +- .../media/videodev2.h.rst.exceptions | 3 + .../media/common/videobuf2/videobuf2-v4l2.c | 4 + drivers/media/platform/nxp/Kconfig | 13 + drivers/media/platform/nxp/Makefile | 1 + drivers/media/platform/nxp/imx-asrc.c | 1256 +++++++++++++++++ drivers/media/test-drivers/Kconfig | 10 + drivers/media/test-drivers/Makefile | 1 + drivers/media/test-drivers/vim2m-audio.c | 793 +++++++++++ drivers/media/test-drivers/vivid/vivid-core.h | 2 + .../media/test-drivers/vivid/vivid-ctrls.c | 26 + drivers/media/v4l2-core/v4l2-compat-ioctl32.c | 9 + drivers/media/v4l2-core/v4l2-ctrls-api.c | 1 + drivers/media/v4l2-core/v4l2-ctrls-core.c | 93 +- drivers/media/v4l2-core/v4l2-ctrls-defs.c | 10 + drivers/media/v4l2-core/v4l2-dev.c | 21 + drivers/media/v4l2-core/v4l2-ioctl.c | 66 + drivers/media/v4l2-core/v4l2-mem2mem.c | 13 +- include/media/v4l2-ctrls.h | 13 +- include/media/v4l2-dev.h | 2 + include/media/v4l2-ioctl.h | 34 + .../fsl => include/sound}/fsl_asrc_common.h | 60 + include/uapi/linux/media.h | 2 + include/uapi/linux/v4l2-controls.h | 9 + include/uapi/linux/videodev2.h | 50 +- sound/soc/fsl/fsl_asrc.c | 144 ++ sound/soc/fsl/fsl_asrc.h | 4 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 233 +++ sound/soc/fsl/fsl_easrc.h | 6 +- 42 files changed, 3128 insertions(+), 27 deletions(-) create mode 100644 Documentation/userspace-api/media/v4l/dev-audio-mem2mem.rst create mode 100644 Documentation/userspace-api/media/v4l/ext-ctrls-audio-m2m.rst create mode 100644 Documentation/userspace-api/media/v4l/pixfmt-audio.rst create mode 100644 drivers/media/platform/nxp/imx-asrc.c create mode 100644 drivers/media/test-drivers/vim2m-audio.c rename {sound/soc/fsl => include/sound}/fsl_asrc_common.h (60%) -- 2.34.1

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[PATCH v1 1/1] ASoC: fsl: imx-es8328: Switch to using gpiod API
by Andy Shevchenko 18 Mar '24

18 Mar '24

This updates the driver to gpiod API, and removes yet another use of of_get_named_gpio(). Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/fsl/imx-es8328.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index 6f0d031c1d5f..ad915c55cc35 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -3,7 +3,7 @@ // Copyright 2012 Freescale Semiconductor, Inc. // Copyright 2012 Linaro Ltd. -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> #include <linux/of_platform.h> @@ -23,7 +23,7 @@ struct imx_es8328_data { struct snd_soc_card card; char codec_dai_name[DAI_NAME_SIZE]; char platform_name[DAI_NAME_SIZE]; - int jack_gpio; + struct gpio_desc *jack_gpiod; }; static struct snd_soc_jack_gpio headset_jack_gpios[] = { @@ -54,8 +54,8 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) struct imx_es8328_data, card); int ret = 0; - /* Headphone jack detection */ - if (gpio_is_valid(data->jack_gpio)) { + if (data->jack_gpiod) { + /* Headphone jack detection */ ret = snd_soc_card_jack_new_pins(rtd->card, "Headphone", SND_JACK_HEADSET | SND_JACK_BTN_0, &headset_jack, @@ -64,7 +64,7 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - headset_jack_gpios[0].gpio = data->jack_gpio; + headset_jack_gpios[0].desc = data->jack_gpiod; ret = snd_soc_jack_add_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios), headset_jack_gpios); @@ -174,7 +174,11 @@ static int imx_es8328_probe(struct platform_device *pdev) data->dev = dev; - data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0); + data->jack_gpiod = devm_gpiod_get_optional(dev, "jack"); + if (IS_ERR(data->jack_gpiod)) { + ret = PTR_ERR(data->jack_gpiod); + goto put_device; + } /* * CPU == Platform -- 2.43.0.rc1.1.gbec44491f096

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[PATCH V0 0/1] ASoC: soc-compress: Fix and add missing DPCM locking
by Shalini Manjunatha 18 Mar '24

18 Mar '24

We find mising DPCM locking inside soc_compr_set_params_fe() before calling dpcm_be_dai_hw_params() and dpcm_be_dai_prepare() which cause lockdep assert for DPCM lock not held in __soc_pcm_hw_params() and __soc_pcm_prepare() in above flow when ever we play compress offload audio playback use case. To fix this issue added DPCM lock and unlock in both places of above code flow mentioned. Shalini Manjunatha (1): ASoC: soc-compress: Fix and add DPCM locking sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) -- 2.7.4

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[PATCH] ALSA: hda/realtek: fix the hp playback volume issue for LG machines
by Hui Wang 18 Mar '24

18 Mar '24

Recently we tested the headphone playback on 2 LG machines, if we set the volume to the max value or near to the max value, the sound is too loud, it could even bring harm to listeners. A workaround is to decrease the max volume to a reasonable value for the headphone's amplifier, then the users couldn't set the volume bigger than that value from the userspace. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e904f62e1952..d463d416fc23 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6964,6 +6964,25 @@ static void alc256_fixup_mic_no_presence_and_resume(struct hda_codec *codec, } } +static void alc256_decrease_headphone_amp_val(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + u32 caps; + u8 nsteps, offs; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + caps = query_amp_caps(codec, 0x3, HDA_OUTPUT); + nsteps = ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) - 10; + offs = ((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT) - 10; + caps &= ~AC_AMPCAP_NUM_STEPS & ~AC_AMPCAP_OFFSET; + caps |= (nsteps << AC_AMPCAP_NUM_STEPS_SHIFT) | (offs << AC_AMPCAP_OFFSET_SHIFT); + + if (snd_hda_override_amp_caps(codec, 0x3, HDA_OUTPUT, caps)) + codec_warn(codec, "failed to override amp caps for NID 0x3\n"); +} + static void alc_fixup_dell4_mic_no_presence_quiet(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -7382,6 +7401,7 @@ enum { ALC294_FIXUP_CS35L41_I2C_2, ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED, ALC256_FIXUP_ACER_SFG16_MICMUTE_LED, + ALC256_FIXUP_HEADPHONE_AMP_VOL, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9581,6 +9601,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc256_fixup_acer_sfg16_micmute_led, }, + [ALC256_FIXUP_HEADPHONE_AMP_VOL] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_decrease_headphone_amp_val, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -10319,6 +10343,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x9e56, "Lenovo ZhaoYang CF4620Z", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_NO_AUDIO_JACK), SND_PCI_QUIRK(0x1849, 0xa233, "Positivo Master C6300", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1854, 0x0440, "LG CQ6", ALC256_FIXUP_HEADPHONE_AMP_VOL), + SND_PCI_QUIRK(0x1854, 0x0441, "LG CQ6 AIO", ALC256_FIXUP_HEADPHONE_AMP_VOL), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x19e5, 0x320f, "Huawei WRT-WX9 ", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), -- 2.34.1

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[PATCH] ALSA: hda: cs35l41: Support HP Spectre x360 14 eu0000
by Anthony I Gilea 18 Mar '24

18 Mar '24

The new HP Spectre x360 has _DSD for CS35L41 amps in ACPI but reset-gpios and spk-id-gpios are merged into single Package of size 4 so _DSD parser fails to parse it. Overwrite broken _DSD with the correct configuration. Signed-off-by: Anthony I Gilea <i(a)cpp.in> --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ 1 file changed, 2 insertions(+) diff --ruNp a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c --- a/sound/pci/hda/cs35l41_hda_property.c 2024-03-10 23:38:09.000000000 +0300 +++ b/sound/pci/hda/cs35l41_hda_property.c 2024-03-14 18:22:01.887566917 +0300 @@ -64,6 +64,7 @@ static const struct cs35l41_config cs35l { "103C8BE5", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8BE6", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8B3A", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8C15", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "104312AF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431433", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431463", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, @@ -405,6 +406,7 @@ static const struct cs35l41_prop_model c { "CSC3551", "103C8BE5", generic_dsd_config }, { "CSC3551", "103C8BE6", generic_dsd_config }, { "CSC3551", "103C8B3A", generic_dsd_config }, + { "CSC3551", "103C8C15", generic_dsd_config }, { "CSC3551", "104312AF", generic_dsd_config }, { "CSC3551", "10431433", generic_dsd_config }, { "CSC3551", "10431463", generic_dsd_config },

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[PATCH v11] ASoc: tas2783: Add tas2783 codec driver
by Shenghao Ding 18 Mar '24

18 Mar '24

The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp speaker protection algorithm. The integrated speaker voltage and current sense provides for real-time monitoring of loudspeakers. The ASoC component provides the majority of the functionality of the device, all the audio functions. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v11: - simplify tasdevice_set_sdw_stream - fixed some Linux coding style - fixed the spelling mistakes - Select left/right channel based on unique id - a longer description has been added - remove unused crc8 in KCONFIG - Invert of RE here means reciprocal of RE - correct indentation - correct the wrong pow calibrated register - correct comment on RESET register - fixed usually lines don't end with an open parenthesis - fixed tasdevice_sdw_shutdown - use defines for function, entity, etc. for readable - remove unnecessary codec_lock in component_probe() - remove struct sdw_stream_data - remove unused macro definitions - use BIT(2) and BIT(1) instead of 0x2 and 0x1 - implement pm_runtime - reg_start is u8, reg_start + i only adds i byte, it should be reg_start + i * 4, because the calibration values - fixed total_sz in tas2783_calibration - add a define for TASDEVICE_REG(0, 0, 0x0a) - rework macro style in header file - Add select CRC32 - rewrite the mute and the return value handling - Add comments on bin file - Add link_id into the fw name to distinguish the same sdw id on different sdw id. - remove unnecessary mutex_lock - use set_active() in tasdevice_io_init and move all other parts in the probe. - move pm_runtime_get_noresume to the probe - fix the return value of putvol function - add put_noidle in tasdevice_sdw_remove - remove unused header file(of.h) - correct the style of the multi-line comment - Set the readable registers in tas2783_readable_register for debugfs - Use new PM macros - Use default put and get inline SOC_SINGLE_RANGE_TLV and SOC_SINGLE_SX_TLV - Use respective %pt to log the calibrated time - Remove unnecessary struct tm - Add trailing comma tasdevice_dapm_widgets and tasdevice_audio_map - Rewrite the return value handling in tasdevice_sdw_probe - use dev_num instead of id.unique_id - rework the log - Add code to check the size of calibrated data in UEFI - Add description of Layout of calibrated Data in UEFI - switch back link_id and unique_id to identify the devcie - unique_id offset as the device index of the calibrated data - update calibrated data layout stored in UEFI, add link_id and unique_id into the calibrated data in case of tas2783 sitting in different SDQW buses. - correct the wrong unique_id - set the unique_id as hex style in dsp bin file name --- sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 932 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 106 ++++ 4 files changed, 1056 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 59f9742e9ff4..6ba3fd0c0811 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -247,6 +247,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1847,6 +1848,21 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + depends on EFI + select REGMAP + select REGMAP_SOUNDWIRE + select CRC32 + select CRC32_SARWATE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + TAS2783 supports the MIPI SoundWire v1.2 compatible interface + for audio and control data. At the end of calibrated data appends + the crc data with crc32_sarwate for error detection. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f53baa2b9565..6afceb126ae3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -284,6 +284,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -664,6 +665,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..71c218792bc9 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,932 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Kevin Lu <kevin-lu(a)ti.com> +// Shenghao Ding <shenghao-ding(a)ti.com> +// + +#include <asm/unaligned.h> +#include <linux/acpi.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/pcm_params.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include <sound/tas2781-tlv.h> +#include "tas2783.h" + +static const unsigned int tas2783_cali_reg[] = { + TAS2783_CALIBRATION_RE, /* Resistance */ + TAS2783_CALIBRATION_RE_LOW, /* Low limitation of RE */ + TAS2783_CALIBRATION_INV_RE, /* Reciprocal of RE */ + TAS2783_CALIBRATION_POW, /* RMS Power */ + TAS2783_CALIBRATION_TLIMIT /* Temperature limitation */ +}; + +static const struct reg_default tas2783_reg_defaults[] = { + /* Default values for ROM mode. Activated. */ + { 0x8002, 0x1a }, /* AMP inactive. */ + { 0x8097, 0xc8 }, + { 0x80b5, 0x74 }, + { 0x8099, 0x20 }, + { 0xfe8d, 0x0d }, + { 0xfebe, 0x4a }, + { 0x8230, 0x00 }, + { 0x8231, 0x00 }, + { 0x8232, 0x00 }, + { 0x8233, 0x01 }, + { 0x8418, 0x00 }, /* Set volume to 0 dB. */ + { 0x8419, 0x00 }, + { 0x841a, 0x00 }, + { 0x841b, 0x00 }, + { 0x8428, 0x40 }, /* Unmute channel */ + { 0x8429, 0x00 }, + { 0x842a, 0x00 }, + { 0x842b, 0x00 }, + { 0x8548, 0x00 }, /* Set volume to 0 dB. */ + { 0x8549, 0x00 }, + { 0x854a, 0x00 }, + { 0x854b, 0x00 }, + { 0x8558, 0x40 }, /* Unmute channel */ + { 0x8559, 0x00 }, + { 0x855a, 0x00 }, + { 0x855b, 0x00 }, + { 0x800a, 0x3a }, /* Enable both channel */ + { 0x800e, 0x44 }, + { 0x800f, 0x40 }, + { 0x805c, 0x99 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_VOLUME, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0x807F: /* Page 0 */ + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + /* + * Only software reset register was volatiled. + * Bit is self clearing. + * 0b = Don't reset + * 1b = reset + */ + case 0x8001: + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x44ffffff, + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, dvc_tlv), + SOC_SINGLE_SX_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, amp_vol_tlv), +}; + +static void tas2783_apply_calibv1(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + struct regmap *map = tas_dev->regmap; + u8 *cali_start; + u16 offset; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + offset = tas_dev->sdw_peripheral->id.unique_id - + TAS2783_UNIQUE_ID_BASE; + if (offset >= TAS2783_MAX_DEV_NUM) { + dev_err(tas_dev->dev, "%s: offset(%u) is out of range.\n", + __func__, offset); + return; + } + + /* + * The area saving tas2783 calibrated data is specified by its + * unique_id offset. cali_start is the first address of current + * tas2783's calibrated data. + */ + cali_start = (u8 *)(cali_data + offset * sizeof(tas2783_cali_reg)); + for (int i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + ret = regmap_bulk_write(map, tas2783_cali_reg[i], + &cali_start[4 * i], 4); + if (ret) { + dev_err(tas_dev->dev, "Cali failed %x:%d\n", + tas2783_cali_reg[i], ret); + break; + } + } +} + +static void tas2783_apply_calibv2(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + const unsigned int arr_size = ARRAY_SIZE(tas2783_cali_reg); + struct regmap *map = tas_dev->regmap; + unsigned int dev_sum = cali_data[1], i, j, k; + u8 *cali_start; + u16 dev_info; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + dev_info = tas_dev->sdw_peripheral->bus->link_id | + tas_dev->sdw_peripheral->id.unique_id << 16; + + /* + * The area saving tas2783 calibrated data is specified by its + * unique_id offset. cali_start is the first address of current + * tas2783's calibrated data. + */ + cali_start = (u8 *)&cali_data[3]; + for (i = 0; i < dev_sum; i++) { + k = i * (arr_size + 1) + 3; + if (dev_info != cali_data[k]) { + for (j = 0; j < arr_size; j++) { + k = 4 * (k + 1 + j); + ret = regmap_bulk_write(map, + tas2783_cali_reg[j], + &cali_start[k], 4); + if (ret) { + dev_err(tas_dev->dev, + "Cali failed %x:%d\n", + tas2783_cali_reg[j], ret); + break; + } + } + break; + } + } +} + +/* + * Load the calibration data, including speaker impedance, f0, etc. + * Calibration is done by the manufacturer in the factory. The calibration + * data are used by the algorithm for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * The DSP will work with default data values if calibrated data are + * missing or are invalid. + * Layout of calibrated Data in UEFI(total 256 bytes): + * V2: + * ChipID (2783, 4 bytes) + * Device-Sum (4 bytes) + * TimeStamp of Calibration (4 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i info () { + * SDW link id (2bytes) + * SDW unique_id (2bytes) + * } + * Calibrated Data of Device #i (20 bytes) + * } + * CRC (4 bytes) + * Reserved (the rest) + * Or + * V1: + * Calibrated Data of Dev 0(unique_id offset 0) (20 bytes) + * Calibrated Data of Dev 1(unique_id offset 1) (20 bytes) + * Calibrated Data of Dev 2(unique_id offset 2) (20 bytes) + * Calibrated Data of Dev 3(unique_id offset 3) (20 bytes) + * Calibrated Data of Dev 4(unique_id offset 4) (20 bytes) + * Calibrated Data of Dev 5(unique_id offset 5) (20 bytes) + * Calibrated Data of Dev 6(unique_id offset 6) (20 bytes) + * Calibrated Data of Dev 7(unique_id offset 7) (20 bytes) + * TimeStamp of Calibration (4 bytes) + * CRC (4 bytes) + * Reserved (88 bytes) + */ +static void tas2783_calibration(struct tasdevice_priv *tas_dev) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct calibration_data cali_data; + unsigned int *tmp_val; + unsigned int crc; + efi_status_t status; + + cali_data.total_sz = 0; + + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, NULL); + if (status == EFI_BUFFER_TOO_SMALL + && cali_data.total_sz < TAS2783_MAX_CALIDATA_SIZE) { + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, + cali_data.data); + dev_dbg(tas_dev->dev, "%s: cali get %lx bytes result:%ld\n", + __func__, cali_data.total_sz, status); + } + if (status != 0) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return; + } + + tmp_val = (unsigned int *)cali_data.data; + + if (tmp_val[0] == 2783) { + /* Calibrated Data V2 */ + unsigned int dev_sum = tmp_val[1]; + + if (dev_sum > TAS2783_MAX_DEV_NUM || + dev_sum == 0) { + dev_dbg(tas_dev->dev, "No dev in calibrated data V2."); + return; + } + crc = crc32(~0, cali_data.data, 12 + dev_sum * 24) ^ ~0; + if (crc == tmp_val[3 + dev_sum * 6]) { + tas2783_apply_calibv2(tas_dev, tmp_val); + dev_dbg(tas_dev->dev, "V2: %ptTs", &tmp_val[40]); + } else { + dev_dbg(tas_dev->dev, + "V2: CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[41]); + } + } else { + /* Calibrated Data V1 */ + /* 8 devs * 20 bytes calibrated data/dev + 4 bytes Timestamp */ + crc = crc32(~0, cali_data.data, 164) ^ ~0; + if (crc == tmp_val[41]) { + /* Date and time of when calibration was done. */ + tas2783_apply_calibv1(tas_dev, tmp_val); + dev_dbg(tas_dev->dev, "V1: %ptTs", &tmp_val[40]); + } else { + dev_dbg(tas_dev->dev, + "V1: CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[41]); + } + } +} + +static void tasdevice_dspfw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct tas2783_firmware_node *p; + struct regmap *map = tas_dev->regmap; + unsigned char *buf = NULL; + int offset = 0, img_sz; + int ret; + + if (!fmw || !fmw->data) { + dev_warn(tas_dev->dev, + "%s: failed to read %s: work in bypass-dsp mode.\n", + __func__, tas_dev->dspfw_binaryname); + return; + } + buf = (unsigned char *)fmw->data; + + img_sz = get_unaligned_le32(&buf[offset]); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_warn(tas_dev->dev, "%s: size not matching, %d %u.", + __func__, (int)fmw->size, img_sz); + return; + } + + while (offset < img_sz) { + p = (struct tas2783_firmware_node *)(buf + offset); + if (p->length > 1) { + ret = regmap_bulk_write(map, p->download_addr, + buf + offset + sizeof(unsigned int) * 5, p->length); + } else { + ret = regmap_write(map, p->download_addr, + *(buf + offset + sizeof(unsigned int) * 5)); + } + + if (ret != 0) { + dev_dbg(tas_dev->dev, + "%s: load FW fail: %d, work in bypass.\n", + __func__, ret); + return; + } + offset += sizeof(unsigned int) * 5 + p->length; + } + + tas2783_calibration(tas_dev); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC"), +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"}, +}; + +static int tasdevice_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + int ret; + + dev_dbg(dai->dev, "%s: dai_name %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(tas_priv->sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) { + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); + return ret; + } + + dev_dbg(dai->dev, "%s: format: %i rate: %i\n", __func__, + params_format(params), params_rate(params)); + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tas_priv->sdw_peripheral, sdw_stream); + + return 0; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct regmap *map = tas_dev->regmap; + int ret; + + dev_dbg(tas_dev->dev, "%s: %d.\n", __func__, mute); + + if (mute) { + if (direction == SNDRV_PCM_STREAM_CAPTURE) { + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_INACTIVE); + if (ret) + dev_err(tas_dev->dev, + "%s: Disable AEF failed.\n", __func__); + } else { + /* FU23 mute (0x40400108) */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 1); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 mute failed.\n", __func__); + goto out; + } + /* + * Both playback and echo data will be shutdown in + * playback stream. + */ + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK | + TAS2783_REG_AEF_MASK, + TAS2783_REG_PWR_MODE_ACTIVE | + TAS2783_REG_PWR_MODE_SW_PWD); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR&AEF shutdown failed.\n", + __func__); + goto out; + } + tas_dev->pstream = false; + } + } else { + /* FU23 Unmute, 0x40400108. */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 Unmute failed.\n", __func__); + goto out; + } + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK, + TAS2783_REG_PWR_MODE_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR Unmute failed.\n", __func__); + goto out; + } + tas_dev->pstream = true; + } else { + /* Capture stream is the echo ref data for voice. + * Without playback, it can't be active. + */ + if (tas_dev->pstream == true) { + ret = regmap_update_bits(map, + TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: AEF enable failed.\n", + __func__); + goto out; + } + } else { + dev_err(tas_dev->dev, + "%s: No playback, no AEF!", __func__); + ret = -EINVAL; + } + } + } +out: + if (ret) + dev_err(tas_dev->dev, "Mute or unmute %d failed %d.\n", + mute, ret); + + return ret; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_comp_probe(struct snd_soc_component *comp) +{ + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(comp); + acpi_handle handle = ACPI_HANDLE(tas_dev->dev); + const struct firmware *fw_entry = NULL; + const char *sub = NULL; + int ret; + + if (handle) { + sub = acpi_get_subsystem_id(handle); + if (IS_ERR(sub)) + sub = NULL; + } + + tas_dev->component = comp; + + /* + * Each tas2783 in the system has its own dspfw. + */ + if (comp->name_prefix) { + /* + * name_prefix.bin stores the dsp firmware including speaker + * protection algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied into + * firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, "%s-tas2783.bin", + comp->name_prefix); + } else { + /* Compatible with the previous naming rule */ + if (sub) { + /* + * subsys_id-link_id[0,1,...,N]-unique_id + * [0x8,...,0xf].bin stores the dsp firmware including + * speaker protection algorithm, audio acoustic + * algorithm, speaker characters and algorithm params, + * it must be copied into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "%s-%d-0x%x.bin", sub, + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } else { + /* + * tas2783-link_id[0,1,...,N]-unique_id + * [0x8,...,0xf].bin stores the dsp firmware including + * speaker protection algorithm, audio acoustic + * algorithm, speaker characters and algorithm params, + * it must be copied into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "tas2783-%d-0x%x.bin", + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } + } + + ret = request_firmware(&fw_entry, tas_dev->dspfw_binaryname, + tas_dev->dev); + if (ret) { + dev_err(tas_dev->dev, + "%s: request_firmware %x open status: %d.\n", __func__, + tas_dev->sdw_peripheral->id.unique_id, ret); + goto out; + } + + tasdevice_dspfw_ready(fw_entry, tas_dev); + +out: + if (fw_entry) + release_firmware(fw_entry); + return 0; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_comp_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + return ret; + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + pm_runtime_enable(tas_dev->dev); + + return 0; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + struct sdw_dpn_prop *dpn; + unsigned long addr; + int nval, i, j; + u32 bit; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = BIT(2); /* BITMAP: 00000100 */ + prop->sink_ports = BIT(1); /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static int tasdevice_io_init(struct device *dev, + struct sdw_slave *slave) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(tas_priv->regmap, false); + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + * Update count of parent 'active' children. + */ + pm_runtime_set_active(&slave->dev); + } + + /* sw reset */ + ret = regmap_write(tas_priv->regmap, TAS2873_REG_SWRESET, + TAS2873_REG_SWRESET_RESET); + if (ret) { + dev_err(tas_priv->dev, "Reset failed.\n"); + goto out; + } + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, false); + regcache_mark_dirty(tas_priv->regmap); + } + + tas_priv->first_hw_init = true; + tas_priv->hw_init = true; + +out: + return ret; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tas_priv->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tas_priv->hw_init = false; + /* + * Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_priv->hw_init || tas_priv->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return -ENOMEM; + + tas_dev->dev = dev; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_peripheral = peripheral; + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) + return dev_err_probe(dev, PTR_ERR(tas_dev->regmap), + "Failed devm_regmap_init_sdw."); + + return tasdevice_init(tas_dev); +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(&peripheral->dev); + + if (tas_dev->first_hw_init) + pm_runtime_disable(tas_dev->dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {} +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static int __maybe_unused tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (!tas_priv->hw_init) + return 0; + + regcache_cache_only(tas_priv->regmap, true); + + return 0; +} + +#define TAS2783_PROBE_TIMEOUT_MS 5000 + +static int __maybe_unused tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long time; + + if (!tas_priv->first_hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT_MS)); + if (!time) { + dev_err(&slave->dev, "Init not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(tas_priv->regmap, false); + regcache_sync(tas_priv->regmap); + + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume) + RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume, NULL) +}; + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = &tas2783_sdca_pm, + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..631776dcec4c --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,106 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2023 - 2024 Texas Instruments Incorporated + * https://www.ti.com + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + * Shenghao Ding <shenghao-ding(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +/* Unique id start */ +#define TAS2783_UNIQUE_ID_BASE 0x08 +#define TAS2783_MAX_DEV_NUM 8 + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 + \ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0x00, 0x01) +#define TAS2873_REG_SWRESET_MASK BIT(0) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +#define TAS2873_REG_PWR_CTRL TASDEVICE_REG(0x0, 0x00, 0x02) +#define TAS2783_REG_AEF_MASK GENMASK(4, 3) +#define TAS2783_REG_AEF_ACTIVE 0 +#define TAS2783_REG_AEF_INACTIVE GENMASK(4, 3) +#define TAS2783_REG_PWR_MODE_MASK GENMASK(2, 0) +#define TAS2783_REG_PWR_MODE_ACTIVE 0 +#define TAS2783_REG_PWR_MODE_SW_PWD BIT(1) + +#define TAS2783_REG_TDM_RX_CFG TASDEVICE_REG(0x0, 0x00, 0x0a) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1a) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x13, 0x70) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +/* TAS2783 SDCA Control - function number */ +#define TAS2783_FUNC_TYPE_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_FU_VOLUME 0x02 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 +#define TAS2783_DSPFW_FILENAME_LEN 64 + +#define TAS2783_MAX_CALIDATA_SIZE 256 + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct snd_soc_component *component; + struct sdw_slave *sdw_peripheral; + enum sdw_slave_status status; + struct sdw_bus_params params; + struct regmap *regmap; + struct device *dev; + unsigned char dspfw_binaryname[TAS2783_DSPFW_FILENAME_LEN]; + unsigned char dev_name[32]; + unsigned int chip_id; + bool pstream; + bool hw_init; + bool first_hw_init; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

4 6

[PATCH] ALSA: core: add kunitconfig
by Takashi Sakamoto 17 Mar '24

17 Mar '24

It is helpful to add .kunitconfig if we work with the tools provided by KUnit project. The file describes the series of kernel configurations to satisfy the dependency to build the target test. For example: $ ./tools/testing/kunit/kunit.py run --arch=arm64 --cross_compile=aarch64-linux-gnu- --kunitconfig=sound/core/ [11:35:13] Configuring KUnit Kernel ... Regenerating .config ... Populating config with: $ make ARCH=arm64 O=.kunit olddefconfig CROSS_COMPILE=aarch64-linux-gnu- [11:35:19] Building KUnit Kernel ... Populating config with: $ make ARCH=arm64 O=.kunit olddefconfig CROSS_COMPILE=aarch64-linux-gnu- Building with: $ make ARCH=arm64 O=.kunit --jobs=8 CROSS_COMPILE=aarch64-linux-gnu- [11:37:35] Starting KUnit Kernel (1/1)... [11:37:35] ============================================================ Running tests with: $ qemu-system-aarch64 -nodefaults -m 1024 -kernel .kunit/arch/arm64/boot/Image.gz -append 'kunit.enable=1 console=ttyAMA0 kunit_shutdown=reboot' -no-reboot -nographic -serial stdio -machine virt -cpu max,pauth-impdef=on [11:37:35] ============== sound-core-test (10 subtests) =============== [11:37:35] [PASSED] test_phys_format_size [11:37:35] [PASSED] test_format_width [11:37:35] [PASSED] test_format_endianness [11:37:35] [PASSED] test_format_signed [11:37:35] [PASSED] test_format_fill_silence [11:37:35] [PASSED] test_playback_avail [11:37:35] [PASSED] test_capture_avail [11:37:35] [PASSED] test_card_set_id [11:37:35] [PASSED] test_pcm_format_name [11:37:35] [PASSED] test_card_add_component [11:37:35] ================= [PASSED] sound-core-test ================= [11:37:35] ============================================================ [11:37:35] Testing complete. Ran 10 tests: passed: 10 [11:37:35] Elapsed time: 142.333s total, 5.617s configuring, 136.047s building, 0.630s running Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/.kunitconfig | 5 +++++ 1 file changed, 5 insertions(+) create mode 100644 sound/core/.kunitconfig diff --git a/sound/core/.kunitconfig b/sound/core/.kunitconfig new file mode 100644 index 000000000000..440f974ba0b7 --- /dev/null +++ b/sound/core/.kunitconfig @@ -0,0 +1,5 @@ +CONFIG_KUNIT=y +CONFIG_SOUND=y +CONFIG_SND=y +CONFIG_SND_PCM=y +CONFIG_SND_CORE_TEST=y -- 2.43.0

2 1