From 72cafe63b35d06b5cfbaf807e90ae657907858da Mon Sep 17 00:00:00 2001 From: Andrey Shumilin Date: Fri, 18 Oct 2024 09:00:18 +0300 Subject: [PATCH 01/97] ALSA: firewire-lib: Avoid division by zero in apply_constraint_to_size() The step variable is initialized to zero. It is changed in the loop, but if it's not changed it will remain zero. Add a variable check before the division. The observed behavior was introduced by commit 826b5de90c0b ("ALSA: firewire-lib: fix insufficient PCM rule for period/buffer size"), and it is difficult to show that any of the interval parameters will satisfy the snd_interval_test() condition with data from the amdtp_rate_table[] table. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 826b5de90c0b ("ALSA: firewire-lib: fix insufficient PCM rule for period/buffer size") Signed-off-by: Andrey Shumilin Reviewed-by: Takashi Sakamoto Link: https://patch.msgid.link/20241018060018.1189537-1-shum.sdl@nppct.ru Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index c72b2a75477598..7fc51f829eccac 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -172,6 +172,9 @@ static int apply_constraint_to_size(struct snd_pcm_hw_params *params, step = max(step, amdtp_syt_intervals[i]); } + if (step == 0) + return -EINVAL; + t.min = roundup(s->min, step); t.max = rounddown(s->max, step); t.integer = 1; From 35fdc6e1c16099078bcbd73a6c8f1733ae7f1909 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Jos=C3=A9=20Relvas?= Date: Sun, 20 Oct 2024 11:27:56 +0100 Subject: [PATCH 02/97] ALSA: hda/realtek: Add subwoofer quirk for Acer Predator G9-593 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The Acer Predator G9-593 has a 2+1 speaker system which isn't probed correctly. This patch adds a quirk with the proper pin connections. Note that I do not own this laptop, so I cannot guarantee that this fixes the issue. Testing was done by other users here: https://discussion.fedoraproject.org/t/-/118482 This model appears to have two different dev IDs... - 0x1177 (as seen on the forum link above) - 0x1178 (as seen on https://linux-hardware.org/?probe=127df9999f) I don't think the audio system was changed between model revisions, so the patch applies for both IDs. Signed-off-by: José Relvas Link: https://patch.msgid.link/20241020102756.225258-1-josemonsantorelvas@gmail.com Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3bbf5fab288153..edf688f989c8a1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7649,6 +7649,7 @@ enum { ALC286_FIXUP_ACER_AIO_HEADSET_MIC, ALC256_FIXUP_ASUS_HEADSET_MIC, ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, + ALC255_FIXUP_PREDATOR_SUBWOOFER, ALC299_FIXUP_PREDATOR_SPK, ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, ALC289_FIXUP_DELL_SPK1, @@ -9063,6 +9064,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE }, + [ALC255_FIXUP_PREDATOR_SUBWOOFER] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x17, 0x90170151 }, /* use as internal speaker (LFE) */ + { 0x1b, 0x90170152 } /* use as internal speaker (back) */ + } + }, [ALC299_FIXUP_PREDATOR_SPK] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10150,6 +10158,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1166, "Acer Veriton N4640G", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x1025, 0x1167, "Acer Veriton N6640G", ALC269_FIXUP_LIFEBOOK), + SND_PCI_QUIRK(0x1025, 0x1177, "Acer Predator G9-593", ALC255_FIXUP_PREDATOR_SUBWOOFER), + SND_PCI_QUIRK(0x1025, 0x1178, "Acer Predator G9-593", ALC255_FIXUP_PREDATOR_SUBWOOFER), SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK), SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), SND_PCI_QUIRK(0x1025, 0x1248, "Acer Veriton N4660G", ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE), From 86c96e7289c5758284b562ac7b5c94429f48d2d9 Mon Sep 17 00:00:00 2001 From: Eric Biggers Date: Sun, 20 Oct 2024 10:56:24 -0700 Subject: [PATCH 03/97] ALSA: hda/tas2781: select CRC32 instead of CRC32_SARWATE Fix the kconfig option for the tas2781 HDA driver to select CRC32 rather than CRC32_SARWATE. CRC32_SARWATE is an option from the kconfig 'choice' that selects the specific CRC32 implementation. Selecting a 'choice' option seems to have no effect, but even if it did work, it would be incorrect for a random driver to override the user's choice. CRC32 is the correct option to select for crc32() to be available. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Cc: stable@vger.kernel.org Signed-off-by: Eric Biggers Link: https://patch.msgid.link/20241020175624.7095-1-ebiggers@kernel.org Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bb15a0248250cc..68f1eee9e5c938 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -198,7 +198,7 @@ config SND_HDA_SCODEC_TAS2781_I2C depends on SND_SOC select SND_SOC_TAS2781_COMLIB select SND_SOC_TAS2781_FMWLIB - select CRC32_SARWATE + select CRC32 help Say Y or M here to include TAS2781 I2C HD-audio side codec support in snd-hda-intel driver, such as ALC287. From e3ea2757c312e51bbf62ebc434a6f7df1e3a201f Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 23 Oct 2024 16:13:10 +0800 Subject: [PATCH 04/97] ALSA: hda/realtek: Update default depop procedure Old procedure has a chance to meet Headphone no output. Fixes: c2d6af53a43f ("ALSA: hda/realtek - Add default procedure for suspend and resume state") Signed-off-by: Kailang Yang Link: https://lore.kernel.org/17b717a0a0b04a77aea4a8ec820cba13@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 38 ++++++++++++++++------------------- 1 file changed, 17 insertions(+), 21 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index edf688f989c8a1..3567b14b52b7c2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3868,20 +3868,18 @@ static void alc_default_init(struct hda_codec *codec) hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); - if (hp_pin_sense) + if (hp_pin_sense) { msleep(2); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - - if (hp_pin_sense) - msleep(85); + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + msleep(75); - if (hp_pin_sense) - msleep(100); + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + msleep(75); + } } static void alc_default_shutup(struct hda_codec *codec) @@ -3897,22 +3895,20 @@ static void alc_default_shutup(struct hda_codec *codec) hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); - if (hp_pin_sense) + if (hp_pin_sense) { msleep(2); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - - if (hp_pin_sense) - msleep(85); - - if (!spec->no_shutup_pins) snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - if (hp_pin_sense) - msleep(100); + msleep(75); + if (!spec->no_shutup_pins) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + + msleep(75); + } alc_auto_setup_eapd(codec, false); alc_shutup_pins(codec); } From 78e7be018784934081afec77f96d49a2483f9188 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 18 Oct 2024 13:53:24 +0800 Subject: [PATCH 05/97] ALSA: hda/realtek: Limit internal Mic boost on Dell platform Dell want to limit internal Mic boost on all Dell platform. Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/561fc5f5eff04b6cbd79ed173cd1c1db@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3567b14b52b7c2..784ac058418fc9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7521,6 +7521,7 @@ enum { ALC286_FIXUP_SONY_MIC_NO_PRESENCE, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC269_FIXUP_DELL1_LIMIT_INT_MIC_BOOST, ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, @@ -7555,6 +7556,7 @@ enum { ALC255_FIXUP_ACER_MIC_NO_PRESENCE, ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC255_FIXUP_DELL1_LIMIT_INT_MIC_BOOST, ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, ALC255_FIXUP_HEADSET_MODE, ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, @@ -8114,6 +8116,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC269_FIXUP_DELL1_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + }, [ALC269_FIXUP_DELL2_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -8394,6 +8402,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_HEADSET_MODE }, + [ALC255_FIXUP_DELL1_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE + }, [ALC255_FIXUP_DELL2_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -11076,6 +11090,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"}, {.id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, .name = "dell-headset3"}, {.id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, .name = "dell-headset4"}, + {.id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE_QUIET, .name = "dell-headset4-quiet"}, {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"}, {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, @@ -11630,16 +11645,16 @@ static const struct snd_hda_pin_quirk alc269_fallback_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, {0x19, 0x40000000}, {0x1b, 0x40000000}), - SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE_QUIET, {0x19, 0x40000000}, {0x1b, 0x40000000}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x19, 0x40000000}, {0x1a, 0x40000000}), - SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_LIMIT_INT_MIC_BOOST, {0x19, 0x40000000}, {0x1a, 0x40000000}), - SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, + SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC269_FIXUP_DELL1_LIMIT_INT_MIC_BOOST, {0x19, 0x40000000}, {0x1a, 0x40000000}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1043, "ASUS", ALC2XX_FIXUP_HEADSET_MIC, From ef5fbdf732a158ec27eeba69d8be851351f29f73 Mon Sep 17 00:00:00 2001 From: Piyush Raj Chouhan Date: Mon, 28 Oct 2024 15:55:16 +0000 Subject: [PATCH 06/97] ALSA: hda/realtek: Add subwoofer quirk for Infinix ZERO BOOK 13 Infinix ZERO BOOK 13 has a 2+2 speaker system which isn't probed correctly. This patch adds a quirk with the proper pin connections. Also The mic in this laptop suffers too high gain resulting in mostly fan noise being recorded, This patch Also limit mic boost. HW Probe for device; https://linux-hardware.org/?probe=a2e892c47b Test: All 4 speaker works, Mic has low noise. Signed-off-by: Piyush Raj Chouhan Link: https://patch.msgid.link/20241028155516.15552-1-piyuschouhan1598@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 784ac058418fc9..7f4926194e50f2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7552,6 +7552,7 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC269_FIXUP_DMIC_THINKPAD_ACPI, + ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13, ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO, ALC255_FIXUP_ACER_MIC_NO_PRESENCE, ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -7998,6 +7999,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_U7x7_headset_mic, }, + [ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170151 }, /* use as internal speaker (LFE) */ + { 0x1b, 0x90170152 }, /* use as internal speaker (back) */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, [ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -11003,6 +11014,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x2782, 0x0214, "VAIO VJFE-CL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x2782, 0x0228, "Infinix ZERO BOOK 13", ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13), SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO), SND_PCI_QUIRK(0x2782, 0x1707, "Vaio VJFE-ADL", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), From 4413665dd6c528b31284119e3571c25f371e1c36 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Jan=20Sch=C3=A4r?= Date: Tue, 29 Oct 2024 23:12:49 +0100 Subject: [PATCH 07/97] ALSA: usb-audio: Add quirks for Dell WD19 dock MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The WD19 family of docks has the same audio chipset as the WD15. This change enables jack detection on the WD19. We don't need the dell_dock_mixer_init quirk for the WD19. It is only needed because of the dell_alc4020_map quirk for the WD15 in mixer_maps.c, which disables the volume controls. Even for the WD15, this quirk was apparently only needed when the dock firmware was not updated. Signed-off-by: Jan Schär Cc: Link: https://patch.msgid.link/20241029221249.15661-1-jan@jschaer.ch Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 2a9594f34dac6f..6456e87e2f3974 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -4042,6 +4042,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) break; err = dell_dock_mixer_init(mixer); break; + case USB_ID(0x0bda, 0x402e): /* Dell WD19 dock */ + err = dell_dock_mixer_create(mixer); + break; case USB_ID(0x2a39, 0x3fd2): /* RME ADI-2 Pro */ case USB_ID(0x2a39, 0x3fd3): /* RME ADI-2 DAC */ From 0b04fbe886b4274c8e5855011233aaa69fec6e75 Mon Sep 17 00:00:00 2001 From: Christoffer Sandberg Date: Tue, 29 Oct 2024 16:16:52 +0100 Subject: [PATCH 08/97] ALSA: hda/realtek: Fix headset mic on TUXEDO Gemini 17 Gen3 Quirk is needed to enable headset microphone on missing pin 0x19. Signed-off-by: Christoffer Sandberg Signed-off-by: Werner Sembach Cc: Link: https://patch.msgid.link/20241029151653.80726-1-wse@tuxedocomputers.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7f4926194e50f2..e06a6fdc0bab78 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10750,6 +10750,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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, 0x28c1, "Clevo V370VND", ALC2XX_FIXUP_HEADSET_MIC), 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), From e49370d769e71456db3fbd982e95bab8c69f73e8 Mon Sep 17 00:00:00 2001 From: Christoffer Sandberg Date: Tue, 29 Oct 2024 16:16:53 +0100 Subject: [PATCH 09/97] ALSA: hda/realtek: Fix headset mic on TUXEDO Stellaris 16 Gen6 mb1 Quirk is needed to enable headset microphone on missing pin 0x19. Signed-off-by: Christoffer Sandberg Signed-off-by: Werner Sembach Cc: Link: https://patch.msgid.link/20241029151653.80726-2-wse@tuxedocomputers.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e06a6fdc0bab78..571fa8a6c9e120 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11008,6 +11008,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d05, 0x115c, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d05, 0x1387, "TongFang GMxIXxx", ALC2XX_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1d05, 0x1409, "TongFang GMxIXxx", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1d17, 0x3288, "Haier Boyue G42", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), From bd0aff85d5f3f3fc22735ab5869008dfd8ab4867 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Thu, 31 Oct 2024 12:33:02 +0200 Subject: [PATCH 10/97] ASoC: codecs: wcd937x: Remove unused of_gpio.h of_gpio.h is deprecated and subject to remove. The drivers in question don't use it, simply remove the unused header. Signed-off-by: Andy Shevchenko Link: https://patch.msgid.link/20241031103302.2450830-1-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/wcd937x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wcd937x.c b/sound/soc/codecs/wcd937x.c index 45f32d28190813..a11f9be91da4e6 100644 --- a/sound/soc/codecs/wcd937x.c +++ b/sound/soc/codecs/wcd937x.c @@ -7,7 +7,6 @@ #include #include #include -#include #include #include #include From 019610566757a749dde7e0c92777d2c1613afef8 Mon Sep 17 00:00:00 2001 From: anish kumar Date: Wed, 30 Oct 2024 20:58:29 -0700 Subject: [PATCH 11/97] ASoC: doc: update clock api details Added ASoC clock api kernel doc in this document. Signed-off-by: anish kumar Link: https://patch.msgid.link/20241031035829.54852-1-yesanishhere@gmail.com Signed-off-by: Mark Brown --- Documentation/sound/soc/clocking.rst | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/Documentation/sound/soc/clocking.rst b/Documentation/sound/soc/clocking.rst index 32122d6877a3d3..25d016ea8b65f6 100644 --- a/Documentation/sound/soc/clocking.rst +++ b/Documentation/sound/soc/clocking.rst @@ -42,5 +42,17 @@ rate, number of channels and word size) to save on power. It is also desirable to use the codec (if possible) to drive (or master) the audio clocks as it usually gives more accurate sample rates than the CPU. +ASoC provided clock APIs +------------------------ +.. kernel-doc:: sound/soc/soc-dai.c + :identifiers: snd_soc_dai_set_sysclk +.. kernel-doc:: sound/soc/soc-dai.c + :identifiers: snd_soc_dai_set_clkdiv + +.. kernel-doc:: sound/soc/soc-dai.c + :identifiers: snd_soc_dai_set_pll + +.. kernel-doc:: sound/soc/soc-dai.c + :identifiers: snd_soc_dai_set_bclk_ratio From c9363bbb0f68dd1ddb8be7bbfe958cdfcd38d851 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Jaros=C5=82aw=20Janik?= Date: Wed, 30 Oct 2024 18:18:12 +0100 Subject: [PATCH 12/97] Revert "ALSA: hda/conexant: Mute speakers at suspend / shutdown" MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Commit 4f61c8fe3520 ("ALSA: hda/conexant: Mute speakers at suspend / shutdown") mutes speakers on system shutdown or whenever HDA controller is suspended by PM; this however interacts badly with Thinkpad's ACPI firmware behavior which uses beeps to signal various events (enter/leave suspend or hibernation, AC power connect/disconnect, low battery, etc.); now those beeps are either muted altogether (for suspend/hibernate/ shutdown related events) or work more or less randomly (eg. AC plug/unplug is only audible when you are playing music at the moment, because HDA device is likely in suspend mode otherwise). Since the original bug report mentioned in 4f61c8fe3520 complained about Lenovo's Thinkpad laptop - revert this commit altogether. Fixes: 4f61c8fe3520 ("ALSA: hda/conexant: Mute speakers at suspend / shutdown") Signed-off-by: Jarosław Janik Link: https://patch.msgid.link/20241030171813.18941-2-jaroslaw.janik@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c74f6742c35955..b2bcdf76da3058 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -205,8 +205,6 @@ static void cx_auto_shutdown(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - snd_hda_gen_shutup_speakers(codec); - /* Turn the problematic codec into D3 to avoid spurious noises from the internal speaker during (and after) reboot */ cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false); From 1ed9b927e7dd8b8cff13052efe212a8ff72ec51d Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Thu, 31 Oct 2024 18:37:04 +0200 Subject: [PATCH 13/97] regmap: maple: Provide lockdep (sub)class for maple tree's internal lock In some cases when using the maple tree register cache, the lockdep validator might complain about invalid deadlocks: [7.131886] Possible interrupt unsafe locking scenario: [7.131890] CPU0 CPU1 [7.131893] ---- ---- [7.131896] lock(&mt->ma_lock); [7.131904] local_irq_disable(); [7.131907] lock(rockchip_drm_vop2:3114:(&vop2_regmap_config)->lock); [7.131916] lock(&mt->ma_lock); [7.131925] [7.131928] lock(rockchip_drm_vop2:3114:(&vop2_regmap_config)->lock); [7.131936] *** DEADLOCK *** [7.131939] no locks held by swapper/0/0. [7.131944] the shortest dependencies between 2nd lock and 1st lock: [7.131950] -> (&mt->ma_lock){+.+.}-{2:2} { [7.131966] HARDIRQ-ON-W at: [7.131973] lock_acquire+0x200/0x330 [7.131986] _raw_spin_lock+0x50/0x70 [7.131998] regcache_maple_write+0x68/0xe0 [7.132010] regcache_write+0x6c/0x90 [7.132019] _regmap_read+0x19c/0x1d0 [7.132029] _regmap_update_bits+0xc0/0x148 [7.132038] regmap_update_bits_base+0x6c/0xa8 [7.132048] rk8xx_probe+0x22c/0x3d8 [7.132057] rk8xx_spi_probe+0x74/0x88 [7.132065] spi_probe+0xa8/0xe0 [...] [7.132675] } [7.132678] ... key at: [] __key.0+0x0/0x10 [7.132691] ... acquired at: [7.132695] _raw_spin_lock+0x50/0x70 [7.132704] regcache_maple_write+0x68/0xe0 [7.132714] regcache_write+0x6c/0x90 [7.132724] _regmap_read+0x19c/0x1d0 [7.132732] _regmap_update_bits+0xc0/0x148 [7.132741] regmap_field_update_bits_base+0x74/0xb8 [7.132751] vop2_plane_atomic_update+0x480/0x14d8 [rockchipdrm] [7.132820] drm_atomic_helper_commit_planes+0x1a0/0x320 [drm_kms_helper] [...] [7.135112] -> (rockchip_drm_vop2:3114:(&vop2_regmap_config)->lock){-...}-{2:2} { [7.135130] IN-HARDIRQ-W at: [7.135136] lock_acquire+0x200/0x330 [7.135147] _raw_spin_lock_irqsave+0x6c/0x98 [7.135157] regmap_lock_spinlock+0x20/0x40 [7.135166] regmap_read+0x44/0x90 [7.135175] vop2_isr+0x90/0x290 [rockchipdrm] [7.135225] __handle_irq_event_percpu+0x124/0x2d0 In the example above, the validator seems to get the scope of dependencies wrong, since the regmap instance used in rk8xx-spi driver has nothing to do with the instance from vop2. Improve validation by sharing the regmap's lockdep class with the maple tree's internal lock, while also providing a subclass for the latter. Signed-off-by: Cristian Ciocaltea Link: https://patch.msgid.link/20241031-regmap-maple-lockdep-fix-v2-1-06a3710f3623@collabora.com Signed-off-by: Mark Brown --- drivers/base/regmap/internal.h | 1 + drivers/base/regmap/regcache-maple.c | 3 +++ drivers/base/regmap/regmap.c | 1 + 3 files changed, 5 insertions(+) diff --git a/drivers/base/regmap/internal.h b/drivers/base/regmap/internal.h index 83acccdc100897..bdb450436cbc53 100644 --- a/drivers/base/regmap/internal.h +++ b/drivers/base/regmap/internal.h @@ -59,6 +59,7 @@ struct regmap { unsigned long raw_spinlock_flags; }; }; + struct lock_class_key *lock_key; regmap_lock lock; regmap_unlock unlock; void *lock_arg; /* This is passed to lock/unlock functions */ diff --git a/drivers/base/regmap/regcache-maple.c b/drivers/base/regmap/regcache-maple.c index 8d27d3653ea3e7..23da7b31d71534 100644 --- a/drivers/base/regmap/regcache-maple.c +++ b/drivers/base/regmap/regcache-maple.c @@ -355,6 +355,9 @@ static int regcache_maple_init(struct regmap *map) mt_init(mt); + if (!mt_external_lock(mt) && map->lock_key) + lockdep_set_class_and_subclass(&mt->ma_lock, map->lock_key, 1); + if (!map->num_reg_defaults) return 0; diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 4ded93687c1f0a..53131a7ede0a6a 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -745,6 +745,7 @@ struct regmap *__regmap_init(struct device *dev, lock_key, lock_name); } map->lock_arg = map; + map->lock_key = lock_key; } /* From 485df22866559e2f821a9754d51a9755ce56e7aa Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 1 Nov 2024 07:38:01 +0530 Subject: [PATCH 14/97] ASoC: sdw_utils/intel/amd: refactor dai link init logic Add 'no_pcm' as parameter for asoc_sdw_init_dai_link() so that same function can be used for SOF and legacy(No DSP) stack. Pass 'no_pcm' as 1 for Intel and AMD SOF based machine drivers. Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Liam Girdwood Reviewed-by: Ranjani Sridharan Link: https://patch.msgid.link/20241101020802.1103181-2-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 5 +++-- sound/soc/amd/acp/acp-sdw-sof-mach.c | 8 ++++---- sound/soc/intel/boards/sof_sdw.c | 12 ++++++------ sound/soc/sdw_utils/soc_sdw_utils.c | 9 +++++---- 4 files changed, 18 insertions(+), 16 deletions(-) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index a25f94d6eb678c..0e82598e10af05 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -152,14 +152,15 @@ void asoc_sdw_init_dai_link(struct device *dev, struct snd_soc_dai_link *dai_lin struct snd_soc_dai_link_component *cpus, int cpus_num, struct snd_soc_dai_link_component *platform_component, int num_platforms, struct snd_soc_dai_link_component *codecs, - int codecs_num, int (*init)(struct snd_soc_pcm_runtime *rtd), + int codecs_num, int no_pcm, + int (*init)(struct snd_soc_pcm_runtime *rtd), const struct snd_soc_ops *ops); int asoc_sdw_init_simple_dai_link(struct device *dev, struct snd_soc_dai_link *dai_links, int *be_id, char *name, int playback, int capture, const char *cpu_dai_name, const char *platform_comp_name, int num_platforms, const char *codec_name, - const char *codec_dai_name, + const char *codec_dai_name, int no_pcm, int (*init)(struct snd_soc_pcm_runtime *rtd), const struct snd_soc_ops *ops); diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index 36e6d6db90c172..8fce8cb957c9b8 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -236,7 +236,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, asoc_sdw_init_dai_link(dev, *dai_links, be_id, name, playback, capture, cpus, num_cpus, platform_component, ARRAY_SIZE(platform_component), codecs, num_codecs, - asoc_sdw_rtd_init, &sdw_ops); + 1, asoc_sdw_rtd_init, &sdw_ops); /* * SoundWire DAILINKs use 'stream' functions and Bank Switch operations @@ -285,7 +285,7 @@ static int create_sdw_dailinks(struct snd_soc_card *card, } static int create_dmic_dailinks(struct snd_soc_card *card, - struct snd_soc_dai_link **dai_links, int *be_id) + struct snd_soc_dai_link **dai_links, int *be_id, int no_pcm) { struct device *dev = card->dev; int ret; @@ -294,7 +294,7 @@ static int create_dmic_dailinks(struct snd_soc_card *card, 0, 1, // DMIC only supports capture "acp-sof-dmic", platform_component->name, ARRAY_SIZE(platform_component), - "dmic-codec", "dmic-hifi", + "dmic-codec", "dmic-hifi", no_pcm, asoc_sdw_dmic_init, NULL); if (ret) return ret; @@ -377,7 +377,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) if (ctx->ignore_internal_dmic) { dev_warn(dev, "Ignoring ACP DMIC\n"); } else { - ret = create_dmic_dailinks(card, &dai_links, &be_id); + ret = create_dmic_dailinks(card, &dai_links, &be_id, 1); if (ret) return ret; } diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 5614e706a0bbed..9ca284a1d66663 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -790,7 +790,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, asoc_sdw_init_dai_link(dev, *dai_links, be_id, name, playback, capture, cpus, num_cpus, platform_component, ARRAY_SIZE(platform_component), codecs, num_codecs, - asoc_sdw_rtd_init, &sdw_ops); + 1, asoc_sdw_rtd_init, &sdw_ops); /* * SoundWire DAILINKs use 'stream' functions and Bank Switch operations @@ -867,7 +867,7 @@ static int create_ssp_dailinks(struct snd_soc_card *card, playback, capture, cpu_dai_name, platform_component->name, ARRAY_SIZE(platform_component), codec_name, - ssp_info->dais[0].dai_name, NULL, + ssp_info->dais[0].dai_name, 1, NULL, ssp_info->ops); if (ret) return ret; @@ -892,7 +892,7 @@ static int create_dmic_dailinks(struct snd_soc_card *card, 0, 1, // DMIC only supports capture "DMIC01 Pin", platform_component->name, ARRAY_SIZE(platform_component), - "dmic-codec", "dmic-hifi", + "dmic-codec", "dmic-hifi", 1, asoc_sdw_dmic_init, NULL); if (ret) return ret; @@ -903,7 +903,7 @@ static int create_dmic_dailinks(struct snd_soc_card *card, 0, 1, // DMIC only supports capture "DMIC16k Pin", platform_component->name, ARRAY_SIZE(platform_component), - "dmic-codec", "dmic-hifi", + "dmic-codec", "dmic-hifi", 1, /* don't call asoc_sdw_dmic_init() twice */ NULL, NULL); if (ret) @@ -947,7 +947,7 @@ static int create_hdmi_dailinks(struct snd_soc_card *card, 1, 0, // HDMI only supports playback cpu_dai_name, platform_component->name, ARRAY_SIZE(platform_component), - codec_name, codec_dai_name, + codec_name, codec_dai_name, 1, i == 0 ? sof_sdw_hdmi_init : NULL, NULL); if (ret) return ret; @@ -975,7 +975,7 @@ static int create_bt_dailinks(struct snd_soc_card *card, 1, 1, cpu_dai_name, platform_component->name, ARRAY_SIZE(platform_component), snd_soc_dummy_dlc.name, snd_soc_dummy_dlc.dai_name, - NULL, NULL); + 1, NULL, NULL); if (ret) return ret; diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index 6610efe8af1851..e7f5938701effa 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -1015,7 +1015,8 @@ void asoc_sdw_init_dai_link(struct device *dev, struct snd_soc_dai_link *dai_lin struct snd_soc_dai_link_component *cpus, int cpus_num, struct snd_soc_dai_link_component *platform_component, int num_platforms, struct snd_soc_dai_link_component *codecs, - int codecs_num, int (*init)(struct snd_soc_pcm_runtime *rtd), + int codecs_num, int no_pcm, + int (*init)(struct snd_soc_pcm_runtime *rtd), const struct snd_soc_ops *ops) { dev_dbg(dev, "create dai link %s, id %d\n", name, *be_id); @@ -1023,7 +1024,7 @@ void asoc_sdw_init_dai_link(struct device *dev, struct snd_soc_dai_link *dai_lin dai_links->name = name; dai_links->platforms = platform_component; dai_links->num_platforms = num_platforms; - dai_links->no_pcm = 1; + dai_links->no_pcm = no_pcm; dai_links->cpus = cpus; dai_links->num_cpus = cpus_num; dai_links->codecs = codecs; @@ -1039,7 +1040,7 @@ int asoc_sdw_init_simple_dai_link(struct device *dev, struct snd_soc_dai_link *d int *be_id, char *name, int playback, int capture, const char *cpu_dai_name, const char *platform_comp_name, int num_platforms, const char *codec_name, - const char *codec_dai_name, + const char *codec_dai_name, int no_pcm, int (*init)(struct snd_soc_pcm_runtime *rtd), const struct snd_soc_ops *ops) { @@ -1058,7 +1059,7 @@ int asoc_sdw_init_simple_dai_link(struct device *dev, struct snd_soc_dai_link *d asoc_sdw_init_dai_link(dev, dai_links, be_id, name, playback, capture, &dlc[0], 1, &dlc[1], num_platforms, - &dlc[2], 1, init, ops); + &dlc[2], 1, no_pcm, init, ops); return 0; } From d280cf5fbfe3cdd373c98e858834ff87b6ea64de Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 1 Nov 2024 07:38:02 +0530 Subject: [PATCH 15/97] ASoC: sdw_utils: Update stream_name in dai_links structure For sof stack, dai_link->stream name will be assigned. For legacy(No DSP enabled) stack, dai_link->stream name should be updated explicitly. Update the stream_name in dai_link structure. Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Liam Girdwood Reviewed-by: Ranjani Sridharan Link: https://patch.msgid.link/20241101020802.1103181-3-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sdw_utils/soc_sdw_utils.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index e7f5938701effa..19bd02e2cd6d67 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -1022,6 +1022,7 @@ void asoc_sdw_init_dai_link(struct device *dev, struct snd_soc_dai_link *dai_lin dev_dbg(dev, "create dai link %s, id %d\n", name, *be_id); dai_links->id = (*be_id)++; dai_links->name = name; + dai_links->stream_name = name; dai_links->platforms = platform_component; dai_links->num_platforms = num_platforms; dai_links->no_pcm = no_pcm; From 1d534bfb2b2ecec4e67a1667c67169f7a22e46f5 Mon Sep 17 00:00:00 2001 From: Weidong Wang Date: Thu, 24 Oct 2024 17:03:23 +0800 Subject: [PATCH 16/97] ASoC: dt-bindings: Add schema for "awinic,aw88081" Add the awinic,aw88081 property to support the aw88081 chip, which is an I2S/TDM input, high efficiency digital Smart K audio amplifie. Signed-off-by: Weidong Wang Reviewed-by: Rob Herring (Arm) Link: https://patch.msgid.link/20241024090324.131731-2-wangweidong.a@awinic.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/awinic,aw88395.yaml | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml b/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml index ac5f2e0f42cbd5..3b0b743e49c4c8 100644 --- a/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml +++ b/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml @@ -17,8 +17,9 @@ description: properties: compatible: enum: - - awinic,aw88395 + - awinic,aw88081 - awinic,aw88261 + - awinic,aw88395 - awinic,aw88399 reg: @@ -56,6 +57,7 @@ allOf: compatible: contains: enum: + - awinic,aw88081 - awinic,aw88261 then: properties: From 88264e4f0b6695245cea2810bf54bebf1c98c070 Mon Sep 17 00:00:00 2001 From: Weidong Wang Date: Thu, 24 Oct 2024 17:03:24 +0800 Subject: [PATCH 17/97] ASoC: codecs: Add aw88081 amplifier driver The driver is for amplifiers aw88081 of Awinic Technology Corporation. The awinic AW88081 is an I2S/TDM input, high efficiency digital Smart K audio amplifier Signed-off-by: Weidong Wang Reviewed-by: anish kumar Link: https://patch.msgid.link/20241024090324.131731-3-wangweidong.a@awinic.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/aw88081.c | 1087 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/aw88081.h | 286 ++++++++++ 4 files changed, 1387 insertions(+) create mode 100644 sound/soc/codecs/aw88081.c create mode 100644 sound/soc/codecs/aw88081.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d3cef4e497f3c8..7dead36be02c4b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,6 +57,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_AW8738 imply SND_SOC_AW87390 imply SND_SOC_AW88395 + imply SND_SOC_AW88081 imply SND_SOC_AW88261 imply SND_SOC_AW88399 imply SND_SOC_BT_SCO @@ -689,6 +690,17 @@ config SND_SOC_AW88261 boost converter can be adjusted smartly according to the input amplitude. +config SND_SOC_AW88081 + tristate "Soc Audio for awinic aw88081" + depends on I2C + select REGMAP_I2C + select SND_SOC_AW88395_LIB + help + This option enables support for aw88081 Smart PA. + The awinic AW88081 is an I2S/TDM input, high efficiency + digital Smart K audio amplifier. Due to its 9uV noise + floor and ultra-low distortion, clean listening is guaranteed. + config SND_SOC_AW87390 tristate "Soc Audio for awinic aw87390" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 2c69df06677e54..ea93968f6bf627 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -49,6 +49,7 @@ snd-soc-arizona-y := arizona.o arizona-jack.o snd-soc-audio-iio-aux-y := audio-iio-aux.o snd-soc-aw8738-y := aw8738.o snd-soc-aw87390-y := aw87390.o +snd-soc-aw88081-y := aw88081.o snd-soc-aw88395-lib-y := aw88395/aw88395_lib.o snd-soc-aw88395-y := aw88395/aw88395.o \ aw88395/aw88395_device.o @@ -465,6 +466,7 @@ obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_AUDIO_IIO_AUX) += snd-soc-audio-iio-aux.o obj-$(CONFIG_SND_SOC_AW8738) += snd-soc-aw8738.o obj-$(CONFIG_SND_SOC_AW87390) += snd-soc-aw87390.o +obj-$(CONFIG_SND_SOC_AW88081) += snd-soc-aw88081.o obj-$(CONFIG_SND_SOC_AW88395_LIB) += snd-soc-aw88395-lib.o obj-$(CONFIG_SND_SOC_AW88395) +=snd-soc-aw88395.o obj-$(CONFIG_SND_SOC_AW88261) +=snd-soc-aw88261.o diff --git a/sound/soc/codecs/aw88081.c b/sound/soc/codecs/aw88081.c new file mode 100644 index 00000000000000..58b8e002d76f08 --- /dev/null +++ b/sound/soc/codecs/aw88081.c @@ -0,0 +1,1087 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// aw88081.c -- AW88081 ALSA SoC Audio driver +// +// Copyright (c) 2024 awinic Technology CO., LTD +// +// Author: Weidong Wang +// + +#include +#include +#include +#include +#include "aw88081.h" +#include "aw88395/aw88395_device.h" + +struct aw88081 { + struct aw_device *aw_pa; + struct mutex lock; + struct delayed_work start_work; + struct regmap *regmap; + struct aw_container *aw_cfg; + + bool phase_sync; +}; + +static const struct regmap_config aw88081_regmap_config = { + .val_bits = 16, + .reg_bits = 8, + .max_register = AW88081_REG_MAX, + .reg_format_endian = REGMAP_ENDIAN_LITTLE, + .val_format_endian = REGMAP_ENDIAN_BIG, +}; + +static int aw88081_dev_get_iis_status(struct aw_device *aw_dev) +{ + unsigned int reg_val; + int ret; + + ret = regmap_read(aw_dev->regmap, AW88081_SYSST_REG, ®_val); + if (ret) + return ret; + if ((reg_val & AW88081_BIT_PLL_CHECK) != AW88081_BIT_PLL_CHECK) { + dev_err(aw_dev->dev, "check pll lock fail,reg_val:0x%04x", reg_val); + return -EINVAL; + } + + return 0; +} + +static int aw88081_dev_check_mode1_pll(struct aw_device *aw_dev) +{ + int ret, i; + + for (i = 0; i < AW88081_DEV_SYSST_CHECK_MAX; i++) { + ret = aw88081_dev_get_iis_status(aw_dev); + if (ret) { + dev_err(aw_dev->dev, "mode1 iis signal check error"); + usleep_range(AW88081_2000_US, AW88081_2000_US + 10); + } else { + return 0; + } + } + + return -EPERM; +} + +static int aw88081_dev_check_mode2_pll(struct aw_device *aw_dev) +{ + unsigned int reg_val; + int ret, i; + + ret = regmap_read(aw_dev->regmap, AW88081_PLLCTRL1_REG, ®_val); + if (ret) + return ret; + + reg_val &= (~AW88081_CCO_MUX_MASK); + if (reg_val == AW88081_CCO_MUX_DIVIDED_VALUE) { + dev_dbg(aw_dev->dev, "CCO_MUX is already divider"); + return -EPERM; + } + + /* change mode2 */ + ret = regmap_update_bits(aw_dev->regmap, AW88081_PLLCTRL1_REG, + ~AW88081_CCO_MUX_MASK, AW88081_CCO_MUX_DIVIDED_VALUE); + if (ret) + return ret; + + for (i = 0; i < AW88081_DEV_SYSST_CHECK_MAX; i++) { + ret = aw88081_dev_get_iis_status(aw_dev); + if (ret) { + dev_err(aw_dev->dev, "mode2 iis check error"); + usleep_range(AW88081_2000_US, AW88081_2000_US + 10); + } else { + break; + } + } + + /* change mode1 */ + ret = regmap_update_bits(aw_dev->regmap, AW88081_PLLCTRL1_REG, + ~AW88081_CCO_MUX_MASK, AW88081_CCO_MUX_BYPASS_VALUE); + if (ret == 0) { + usleep_range(AW88081_2000_US, AW88081_2000_US + 10); + for (i = 0; i < AW88081_DEV_SYSST_CHECK_MAX; i++) { + ret = aw88081_dev_check_mode1_pll(aw_dev); + if (ret) { + dev_err(aw_dev->dev, "mode2 switch to mode1, iis check error"); + usleep_range(AW88081_2000_US, AW88081_2000_US + 10); + } else { + break; + } + } + } + + return ret; +} + +static int aw88081_dev_check_syspll(struct aw_device *aw_dev) +{ + int ret; + + ret = aw88081_dev_check_mode1_pll(aw_dev); + if (ret) { + dev_dbg(aw_dev->dev, "mode1 check iis failed try switch to mode2 check"); + ret = aw88081_dev_check_mode2_pll(aw_dev); + if (ret) { + dev_err(aw_dev->dev, "mode2 check iis failed"); + return ret; + } + } + + return 0; +} + +static int aw88081_dev_check_sysst(struct aw_device *aw_dev) +{ + unsigned int check_val; + unsigned int reg_val; + unsigned int value; + int ret, i; + + ret = regmap_read(aw_dev->regmap, AW88081_PWMCTRL4_REG, ®_val); + if (ret) + return ret; + + if (reg_val & (~AW88081_NOISE_GATE_EN_MASK)) + check_val = AW88081_NO_SWS_SYSST_CHECK; + else + check_val = AW88081_SWS_SYSST_CHECK; + + for (i = 0; i < AW88081_DEV_SYSST_CHECK_MAX; i++) { + ret = regmap_read(aw_dev->regmap, AW88081_SYSST_REG, ®_val); + if (ret) + return ret; + + value = reg_val & (~AW88081_BIT_SYSST_CHECK_MASK) & check_val; + if (value != check_val) { + dev_err(aw_dev->dev, "check sysst fail, reg_val=0x%04x, check:0x%x", + reg_val, check_val); + usleep_range(AW88081_2000_US, AW88081_2000_US + 10); + } else { + return 0; + } + } + + return -EPERM; +} + +static void aw88081_dev_i2s_tx_enable(struct aw_device *aw_dev, bool flag) +{ + if (flag) + regmap_update_bits(aw_dev->regmap, AW88081_I2SCTRL3_REG, + ~AW88081_I2STXEN_MASK, AW88081_I2STXEN_ENABLE_VALUE); + else + regmap_update_bits(aw_dev->regmap, AW88081_I2SCTRL3_REG, + ~AW88081_I2STXEN_MASK, AW88081_I2STXEN_DISABLE_VALUE); +} + +static void aw88081_dev_pwd(struct aw_device *aw_dev, bool pwd) +{ + if (pwd) + regmap_update_bits(aw_dev->regmap, AW88081_SYSCTRL_REG, + ~AW88081_PWDN_MASK, AW88081_PWDN_POWER_DOWN_VALUE); + else + regmap_update_bits(aw_dev->regmap, AW88081_SYSCTRL_REG, + ~AW88081_PWDN_MASK, AW88081_PWDN_WORKING_VALUE); +} + +static void aw88081_dev_amppd(struct aw_device *aw_dev, bool amppd) +{ + if (amppd) + regmap_update_bits(aw_dev->regmap, AW88081_SYSCTRL_REG, + ~AW88081_EN_PA_MASK, AW88081_EN_PA_POWER_DOWN_VALUE); + else + regmap_update_bits(aw_dev->regmap, AW88081_SYSCTRL_REG, + ~AW88081_EN_PA_MASK, AW88081_EN_PA_WORKING_VALUE); +} + +static void aw88081_dev_clear_int_status(struct aw_device *aw_dev) +{ + unsigned int int_status; + + /* read int status and clear */ + regmap_read(aw_dev->regmap, AW88081_SYSINT_REG, &int_status); + /* make sure int status is clear */ + regmap_read(aw_dev->regmap, AW88081_SYSINT_REG, &int_status); + + dev_dbg(aw_dev->dev, "read interrupt reg = 0x%04x", int_status); +} + +static void aw88081_dev_set_volume(struct aw_device *aw_dev, unsigned int value) +{ + struct aw_volume_desc *vol_desc = &aw_dev->volume_desc; + unsigned int volume; + + volume = min((value + vol_desc->init_volume), (unsigned int)AW88081_MUTE_VOL); + + regmap_update_bits(aw_dev->regmap, AW88081_SYSCTRL2_REG, ~AW88081_VOL_MASK, volume); +} + +static void aw88081_dev_fade_in(struct aw_device *aw_dev) +{ + struct aw_volume_desc *desc = &aw_dev->volume_desc; + int fade_in_vol = desc->ctl_volume; + int fade_step = aw_dev->fade_step; + int i; + + if (fade_step == 0 || aw_dev->fade_in_time == 0) { + aw88081_dev_set_volume(aw_dev, fade_in_vol); + return; + } + + for (i = AW88081_MUTE_VOL; i >= fade_in_vol; i -= fade_step) { + aw88081_dev_set_volume(aw_dev, i); + usleep_range(aw_dev->fade_in_time, aw_dev->fade_in_time + 10); + } + + if (i != fade_in_vol) + aw88081_dev_set_volume(aw_dev, fade_in_vol); +} + +static void aw88081_dev_fade_out(struct aw_device *aw_dev) +{ + struct aw_volume_desc *desc = &aw_dev->volume_desc; + int fade_step = aw_dev->fade_step; + int i; + + if (fade_step == 0 || aw_dev->fade_out_time == 0) { + aw88081_dev_set_volume(aw_dev, AW88081_MUTE_VOL); + return; + } + + for (i = desc->ctl_volume; i <= AW88081_MUTE_VOL; i += fade_step) { + aw88081_dev_set_volume(aw_dev, i); + usleep_range(aw_dev->fade_out_time, aw_dev->fade_out_time + 10); + } + + if (i != AW88081_MUTE_VOL) + aw88081_dev_set_volume(aw_dev, AW88081_MUTE_VOL); +} + +static void aw88081_dev_mute(struct aw_device *aw_dev, bool is_mute) +{ + if (is_mute) { + aw88081_dev_fade_out(aw_dev); + regmap_update_bits(aw_dev->regmap, AW88081_SYSCTRL_REG, + ~AW88081_HMUTE_MASK, AW88081_HMUTE_ENABLE_VALUE); + } else { + regmap_update_bits(aw_dev->regmap, AW88081_SYSCTRL_REG, + ~AW88081_HMUTE_MASK, AW88081_HMUTE_DISABLE_VALUE); + aw88081_dev_fade_in(aw_dev); + } +} + +static void aw88081_dev_uls_hmute(struct aw_device *aw_dev, bool uls_hmute) +{ + if (uls_hmute) + regmap_update_bits(aw_dev->regmap, AW88081_SYSCTRL_REG, + ~AW88081_ULS_HMUTE_MASK, + AW88081_ULS_HMUTE_ENABLE_VALUE); + else + regmap_update_bits(aw_dev->regmap, AW88081_SYSCTRL_REG, + ~AW88081_ULS_HMUTE_MASK, + AW88081_ULS_HMUTE_DISABLE_VALUE); +} + +static int aw88081_dev_reg_update(struct aw88081 *aw88081, + unsigned char *data, unsigned int len) +{ + struct aw_device *aw_dev = aw88081->aw_pa; + struct aw_volume_desc *vol_desc = &aw_dev->volume_desc; + unsigned int read_vol; + int data_len, i, ret; + int16_t *reg_data; + u16 reg_val; + u8 reg_addr; + + if (!len || !data) { + dev_err(aw_dev->dev, "reg data is null or len is 0"); + return -EINVAL; + } + + reg_data = (int16_t *)data; + data_len = len >> 1; + + if (data_len & 0x1) { + dev_err(aw_dev->dev, "data len:%d unsupported", data_len); + return -EINVAL; + } + + for (i = 0; i < data_len; i += 2) { + reg_addr = reg_data[i]; + reg_val = reg_data[i + 1]; + + if (reg_addr == AW88081_SYSCTRL_REG) { + reg_val &= ~(~AW88081_EN_PA_MASK | + ~AW88081_PWDN_MASK | + ~AW88081_HMUTE_MASK | + ~AW88081_ULS_HMUTE_MASK); + + reg_val |= AW88081_EN_PA_POWER_DOWN_VALUE | + AW88081_PWDN_POWER_DOWN_VALUE | + AW88081_HMUTE_ENABLE_VALUE | + AW88081_ULS_HMUTE_ENABLE_VALUE; + } + + if (reg_addr == AW88081_SYSCTRL2_REG) { + read_vol = (reg_val & (~AW88081_VOL_MASK)) >> + AW88081_VOL_START_BIT; + aw_dev->volume_desc.init_volume = read_vol; + } + + /* i2stxen */ + if (reg_addr == AW88081_I2SCTRL3_REG) { + /* close tx */ + reg_val &= AW88081_I2STXEN_MASK; + reg_val |= AW88081_I2STXEN_DISABLE_VALUE; + } + + ret = regmap_write(aw_dev->regmap, reg_addr, reg_val); + if (ret) + return ret; + } + + if (aw_dev->prof_cur != aw_dev->prof_index) + vol_desc->ctl_volume = 0; + + /* keep min volume */ + aw88081_dev_set_volume(aw_dev, vol_desc->mute_volume); + + return 0; +} + +static int aw88081_dev_get_prof_name(struct aw_device *aw_dev, int index, char **prof_name) +{ + struct aw_prof_info *prof_info = &aw_dev->prof_info; + struct aw_prof_desc *prof_desc; + + if ((index >= aw_dev->prof_info.count) || (index < 0)) { + dev_err(aw_dev->dev, "index[%d] overflow count[%d]", + index, aw_dev->prof_info.count); + return -EINVAL; + } + + prof_desc = &aw_dev->prof_info.prof_desc[index]; + + *prof_name = prof_info->prof_name_list[prof_desc->id]; + + return 0; +} + +static int aw88081_dev_get_prof_data(struct aw_device *aw_dev, int index, + struct aw_prof_desc **prof_desc) +{ + if ((index >= aw_dev->prof_info.count) || (index < 0)) { + dev_err(aw_dev->dev, "%s: index[%d] overflow count[%d]\n", + __func__, index, aw_dev->prof_info.count); + return -EINVAL; + } + + *prof_desc = &aw_dev->prof_info.prof_desc[index]; + + return 0; +} + +static int aw88081_dev_fw_update(struct aw88081 *aw88081) +{ + struct aw_device *aw_dev = aw88081->aw_pa; + struct aw_prof_desc *prof_index_desc; + struct aw_sec_data_desc *sec_desc; + char *prof_name; + int ret; + + ret = aw88081_dev_get_prof_name(aw_dev, aw_dev->prof_index, &prof_name); + if (ret) { + dev_err(aw_dev->dev, "get prof name failed"); + return -EINVAL; + } + + dev_dbg(aw_dev->dev, "start update %s", prof_name); + + ret = aw88081_dev_get_prof_data(aw_dev, aw_dev->prof_index, &prof_index_desc); + if (ret) + return ret; + + /* update reg */ + sec_desc = prof_index_desc->sec_desc; + ret = aw88081_dev_reg_update(aw88081, sec_desc[AW88395_DATA_TYPE_REG].data, + sec_desc[AW88395_DATA_TYPE_REG].len); + if (ret) { + dev_err(aw_dev->dev, "update reg failed"); + return ret; + } + + aw_dev->prof_cur = aw_dev->prof_index; + + return 0; +} + +static int aw88081_dev_start(struct aw88081 *aw88081) +{ + struct aw_device *aw_dev = aw88081->aw_pa; + int ret; + + if (aw_dev->status == AW88081_DEV_PW_ON) { + dev_dbg(aw_dev->dev, "already power on"); + return 0; + } + + /* power on */ + aw88081_dev_pwd(aw_dev, false); + usleep_range(AW88081_2000_US, AW88081_2000_US + 10); + + ret = aw88081_dev_check_syspll(aw_dev); + if (ret) { + dev_err(aw_dev->dev, "pll check failed cannot start"); + goto pll_check_fail; + } + + /* amppd on */ + aw88081_dev_amppd(aw_dev, false); + usleep_range(AW88081_1000_US, AW88081_1000_US + 50); + + /* check i2s status */ + ret = aw88081_dev_check_sysst(aw_dev); + if (ret) { + dev_err(aw_dev->dev, "sysst check failed"); + goto sysst_check_fail; + } + + /* enable tx feedback */ + aw88081_dev_i2s_tx_enable(aw_dev, true); + + /* close uls mute */ + aw88081_dev_uls_hmute(aw_dev, false); + + /* close mute */ + aw88081_dev_mute(aw_dev, false); + + /* clear inturrupt */ + aw88081_dev_clear_int_status(aw_dev); + aw_dev->status = AW88081_DEV_PW_ON; + + return 0; + +sysst_check_fail: + aw88081_dev_i2s_tx_enable(aw_dev, false); + aw88081_dev_clear_int_status(aw_dev); + aw88081_dev_amppd(aw_dev, true); +pll_check_fail: + aw88081_dev_pwd(aw_dev, true); + aw_dev->status = AW88081_DEV_PW_OFF; + + return ret; +} + +static int aw88081_dev_stop(struct aw_device *aw_dev) +{ + if (aw_dev->status == AW88081_DEV_PW_OFF) { + dev_dbg(aw_dev->dev, "already power off"); + return 0; + } + + aw_dev->status = AW88081_DEV_PW_OFF; + + /* clear inturrupt */ + aw88081_dev_clear_int_status(aw_dev); + + aw88081_dev_uls_hmute(aw_dev, true); + /* set mute */ + aw88081_dev_mute(aw_dev, true); + + /* close tx feedback */ + aw88081_dev_i2s_tx_enable(aw_dev, false); + usleep_range(AW88081_1000_US, AW88081_1000_US + 100); + + /* enable amppd */ + aw88081_dev_amppd(aw_dev, true); + + /* set power down */ + aw88081_dev_pwd(aw_dev, true); + + return 0; +} + +static int aw88081_reg_update(struct aw88081 *aw88081, bool force) +{ + struct aw_device *aw_dev = aw88081->aw_pa; + int ret; + + if (force) { + ret = regmap_write(aw_dev->regmap, + AW88081_ID_REG, AW88081_SOFT_RESET_VALUE); + if (ret) + return ret; + + ret = aw88081_dev_fw_update(aw88081); + if (ret) + return ret; + } else { + if (aw_dev->prof_cur != aw_dev->prof_index) { + ret = aw88081_dev_fw_update(aw88081); + if (ret) + return ret; + } + } + + aw_dev->prof_cur = aw_dev->prof_index; + + return 0; +} + +static void aw88081_start_pa(struct aw88081 *aw88081) +{ + int ret, i; + + for (i = 0; i < AW88081_START_RETRIES; i++) { + ret = aw88081_reg_update(aw88081, aw88081->phase_sync); + if (ret) { + dev_err(aw88081->aw_pa->dev, "fw update failed, cnt:%d\n", i); + continue; + } + ret = aw88081_dev_start(aw88081); + if (ret) { + dev_err(aw88081->aw_pa->dev, "aw88081 device start failed. retry = %d", i); + continue; + } else { + dev_dbg(aw88081->aw_pa->dev, "start success\n"); + break; + } + } +} + +static void aw88081_startup_work(struct work_struct *work) +{ + struct aw88081 *aw88081 = + container_of(work, struct aw88081, start_work.work); + + mutex_lock(&aw88081->lock); + aw88081_start_pa(aw88081); + mutex_unlock(&aw88081->lock); +} + +static void aw88081_start(struct aw88081 *aw88081, bool sync_start) +{ + if (aw88081->aw_pa->fw_status != AW88081_DEV_FW_OK) + return; + + if (aw88081->aw_pa->status == AW88081_DEV_PW_ON) + return; + + if (sync_start == AW88081_SYNC_START) + aw88081_start_pa(aw88081); + else + queue_delayed_work(system_wq, + &aw88081->start_work, + AW88081_START_WORK_DELAY_MS); +} + +static struct snd_soc_dai_driver aw88081_dai[] = { + { + .name = "aw88081-aif", + .id = 1, + .playback = { + .stream_name = "Speaker_Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AW88081_RATES, + .formats = AW88081_FORMATS, + }, + .capture = { + .stream_name = "Speaker_Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AW88081_RATES, + .formats = AW88081_FORMATS, + }, + }, +}; + +static int aw88081_get_fade_in_time(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct aw88081 *aw88081 = snd_soc_component_get_drvdata(component); + struct aw_device *aw_dev = aw88081->aw_pa; + + ucontrol->value.integer.value[0] = aw_dev->fade_in_time; + + return 0; +} + +static int aw88081_set_fade_in_time(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct aw88081 *aw88081 = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct aw_device *aw_dev = aw88081->aw_pa; + int time; + + time = ucontrol->value.integer.value[0]; + + if (time < mc->min || time > mc->max) + return -EINVAL; + + if (time != aw_dev->fade_in_time) { + aw_dev->fade_in_time = time; + return 1; + } + + return 0; +} + +static int aw88081_get_fade_out_time(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct aw88081 *aw88081 = snd_soc_component_get_drvdata(component); + struct aw_device *aw_dev = aw88081->aw_pa; + + ucontrol->value.integer.value[0] = aw_dev->fade_out_time; + + return 0; +} + +static int aw88081_set_fade_out_time(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct aw88081 *aw88081 = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct aw_device *aw_dev = aw88081->aw_pa; + int time; + + time = ucontrol->value.integer.value[0]; + if (time < mc->min || time > mc->max) + return -EINVAL; + + if (time != aw_dev->fade_out_time) { + aw_dev->fade_out_time = time; + return 1; + } + + return 0; +} + +static int aw88081_dev_set_profile_index(struct aw_device *aw_dev, int index) +{ + /* check the index whether is valid */ + if ((index >= aw_dev->prof_info.count) || (index < 0)) + return -EINVAL; + /* check the index whether change */ + if (aw_dev->prof_index == index) + return -EPERM; + + aw_dev->prof_index = index; + + return 0; +} + +static int aw88081_profile_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct aw88081 *aw88081 = snd_soc_component_get_drvdata(codec); + char *prof_name; + int count, ret; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + + count = aw88081->aw_pa->prof_info.count; + if (count <= 0) { + uinfo->value.enumerated.items = 0; + return 0; + } + + uinfo->value.enumerated.items = count; + + if (uinfo->value.enumerated.item >= count) + uinfo->value.enumerated.item = count - 1; + + count = uinfo->value.enumerated.item; + + ret = aw88081_dev_get_prof_name(aw88081->aw_pa, count, &prof_name); + if (ret) { + strscpy(uinfo->value.enumerated.name, "null", + sizeof(uinfo->value.enumerated.name)); + return 0; + } + + strscpy(uinfo->value.enumerated.name, prof_name, sizeof(uinfo->value.enumerated.name)); + + return 0; +} + +static int aw88081_profile_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct aw88081 *aw88081 = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = aw88081->aw_pa->prof_index; + + return 0; +} + +static int aw88081_profile_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct aw88081 *aw88081 = snd_soc_component_get_drvdata(codec); + int ret; + + /* pa stop or stopping just set profile */ + mutex_lock(&aw88081->lock); + ret = aw88081_dev_set_profile_index(aw88081->aw_pa, ucontrol->value.integer.value[0]); + if (ret) { + dev_dbg(codec->dev, "profile index does not change"); + mutex_unlock(&aw88081->lock); + return 0; + } + + if (aw88081->aw_pa->status) { + aw88081_dev_stop(aw88081->aw_pa); + aw88081_start(aw88081, AW88081_SYNC_START); + } + + mutex_unlock(&aw88081->lock); + + return 1; +} + +static int aw88081_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct aw88081 *aw88081 = snd_soc_component_get_drvdata(codec); + struct aw_volume_desc *vol_desc = &aw88081->aw_pa->volume_desc; + + ucontrol->value.integer.value[0] = vol_desc->ctl_volume; + + return 0; +} + +static int aw88081_volume_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct aw88081 *aw88081 = snd_soc_component_get_drvdata(codec); + struct aw_volume_desc *vol_desc = &aw88081->aw_pa->volume_desc; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int value; + + value = ucontrol->value.integer.value[0]; + + if (value < mc->min || value > mc->max) + return -EINVAL; + + if (vol_desc->ctl_volume != value) { + vol_desc->ctl_volume = value; + aw88081_dev_set_volume(aw88081->aw_pa, vol_desc->ctl_volume); + return 1; + } + + return 0; +} + +static int aw88081_get_fade_step(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct aw88081 *aw88081 = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = aw88081->aw_pa->fade_step; + + return 0; +} + +static int aw88081_set_fade_step(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct aw88081 *aw88081 = snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int value; + + value = ucontrol->value.integer.value[0]; + if (value < mc->min || value > mc->max) + return -EINVAL; + + if (aw88081->aw_pa->fade_step != value) { + aw88081->aw_pa->fade_step = value; + return 1; + } + + return 0; +} + +static const struct snd_kcontrol_new aw88081_controls[] = { + SOC_SINGLE_EXT("PCM Playback Volume", AW88081_SYSCTRL2_REG, + 0, AW88081_MUTE_VOL, 0, aw88081_volume_get, + aw88081_volume_set), + SOC_SINGLE_EXT("Fade Step", 0, 0, AW88081_MUTE_VOL, 0, + aw88081_get_fade_step, aw88081_set_fade_step), + SOC_SINGLE_EXT("Volume Ramp Up Step", 0, 0, FADE_TIME_MAX, 0, + aw88081_get_fade_in_time, aw88081_set_fade_in_time), + SOC_SINGLE_EXT("Volume Ramp Down Step", 0, 0, FADE_TIME_MAX, 0, + aw88081_get_fade_out_time, aw88081_set_fade_out_time), + AW88081_PROFILE_EXT("Profile Set", aw88081_profile_info, + aw88081_profile_get, aw88081_profile_set), +}; + +static void aw88081_parse_channel_dt(struct aw88081 *aw88081) +{ + struct aw_device *aw_dev = aw88081->aw_pa; + struct device_node *np = aw_dev->dev->of_node; + u32 channel_value = AW88081_DEV_DEFAULT_CH; + + of_property_read_u32(np, "awinic,audio-channel", &channel_value); + aw88081->phase_sync = of_property_read_bool(np, "awinic,sync-flag"); + + aw_dev->channel = channel_value; +} + +static int aw88081_init(struct aw88081 *aw88081, struct i2c_client *i2c, struct regmap *regmap) +{ + struct aw_device *aw_dev; + unsigned int chip_id; + int ret; + + /* read chip id */ + ret = regmap_read(regmap, AW88081_ID_REG, &chip_id); + if (ret) { + dev_err(&i2c->dev, "%s read chipid error. ret = %d", __func__, ret); + return ret; + } + if (chip_id != AW88081_CHIP_ID) { + dev_err(&i2c->dev, "unsupported device"); + return -ENXIO; + } + + dev_dbg(&i2c->dev, "chip id = %x\n", chip_id); + + aw_dev = devm_kzalloc(&i2c->dev, sizeof(*aw_dev), GFP_KERNEL); + if (!aw_dev) + return -ENOMEM; + + aw88081->aw_pa = aw_dev; + aw_dev->i2c = i2c; + aw_dev->regmap = regmap; + aw_dev->dev = &i2c->dev; + aw_dev->chip_id = AW88081_CHIP_ID; + aw_dev->acf = NULL; + aw_dev->prof_info.prof_desc = NULL; + aw_dev->prof_info.prof_type = AW88395_DEV_NONE_TYPE_ID; + aw_dev->fade_step = AW88081_VOLUME_STEP_DB; + aw_dev->volume_desc.mute_volume = AW88081_MUTE_VOL; + aw88081_parse_channel_dt(aw88081); + + return 0; +} + +static int aw88081_dev_init(struct aw88081 *aw88081, struct aw_container *aw_cfg) +{ + struct aw_device *aw_dev = aw88081->aw_pa; + int ret; + + ret = aw88395_dev_cfg_load(aw_dev, aw_cfg); + if (ret) { + dev_err(aw_dev->dev, "aw_dev acf parse failed"); + return -EINVAL; + } + + ret = regmap_write(aw_dev->regmap, AW88081_ID_REG, AW88081_SOFT_RESET_VALUE); + if (ret) + return ret; + + aw_dev->fade_in_time = AW88081_500_US; + aw_dev->fade_out_time = AW88081_500_US; + aw_dev->prof_cur = AW88081_INIT_PROFILE; + aw_dev->prof_index = AW88081_INIT_PROFILE; + + ret = aw88081_dev_fw_update(aw88081); + if (ret) { + dev_err(aw_dev->dev, "fw update failed ret = %d\n", ret); + return ret; + } + + aw88081_dev_clear_int_status(aw_dev); + + aw88081_dev_uls_hmute(aw_dev, true); + + aw88081_dev_mute(aw_dev, true); + + usleep_range(AW88081_5000_US, AW88081_5000_US + 10); + + aw88081_dev_i2s_tx_enable(aw_dev, false); + + usleep_range(AW88081_1000_US, AW88081_1000_US + 100); + + aw88081_dev_amppd(aw_dev, true); + + aw88081_dev_pwd(aw_dev, true); + + return 0; +} + +static int aw88081_request_firmware_file(struct aw88081 *aw88081) +{ + const struct firmware *cont = NULL; + int ret; + + aw88081->aw_pa->fw_status = AW88081_DEV_FW_FAILED; + + ret = request_firmware(&cont, AW88081_ACF_FILE, aw88081->aw_pa->dev); + if (ret) + return ret; + + dev_dbg(aw88081->aw_pa->dev, "loaded %s - size: %zu\n", + AW88081_ACF_FILE, cont ? cont->size : 0); + + aw88081->aw_cfg = devm_kzalloc(aw88081->aw_pa->dev, cont->size + sizeof(int), GFP_KERNEL); + if (!aw88081->aw_cfg) { + release_firmware(cont); + return -ENOMEM; + } + aw88081->aw_cfg->len = (int)cont->size; + memcpy(aw88081->aw_cfg->data, cont->data, cont->size); + release_firmware(cont); + + ret = aw88395_dev_load_acf_check(aw88081->aw_pa, aw88081->aw_cfg); + if (ret) + return ret; + + mutex_lock(&aw88081->lock); + ret = aw88081_dev_init(aw88081, aw88081->aw_cfg); + mutex_unlock(&aw88081->lock); + + return ret; +} + +static int aw88081_playback_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct aw88081 *aw88081 = snd_soc_component_get_drvdata(component); + + mutex_lock(&aw88081->lock); + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + aw88081_start(aw88081, AW88081_ASYNC_START); + break; + case SND_SOC_DAPM_POST_PMD: + aw88081_dev_stop(aw88081->aw_pa); + break; + default: + break; + } + mutex_unlock(&aw88081->lock); + + return 0; +} + +static const struct snd_soc_dapm_widget aw88081_dapm_widgets[] = { + /* playback */ + SND_SOC_DAPM_AIF_IN_E("AIF_RX", "Speaker_Playback", 0, SND_SOC_NOPM, 0, 0, + aw88081_playback_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("DAC Output"), + + /* capture */ + SND_SOC_DAPM_AIF_OUT("AIF_TX", "Speaker_Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_INPUT("ADC Input"), +}; + +static const struct snd_soc_dapm_route aw88081_audio_map[] = { + {"DAC Output", NULL, "AIF_RX"}, + {"AIF_TX", NULL, "ADC Input"}, +}; + +static int aw88081_codec_probe(struct snd_soc_component *component) +{ + struct aw88081 *aw88081 = snd_soc_component_get_drvdata(component); + int ret; + + INIT_DELAYED_WORK(&aw88081->start_work, aw88081_startup_work); + + ret = aw88081_request_firmware_file(aw88081); + if (ret) + dev_err(aw88081->aw_pa->dev, "%s: request firmware failed\n", __func__); + + return ret; +} + +static void aw88081_codec_remove(struct snd_soc_component *aw_codec) +{ + struct aw88081 *aw88081 = snd_soc_component_get_drvdata(aw_codec); + + cancel_delayed_work_sync(&aw88081->start_work); +} + +static const struct snd_soc_component_driver soc_codec_dev_aw88081 = { + .probe = aw88081_codec_probe, + .remove = aw88081_codec_remove, + .dapm_widgets = aw88081_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aw88081_dapm_widgets), + .dapm_routes = aw88081_audio_map, + .num_dapm_routes = ARRAY_SIZE(aw88081_audio_map), + .controls = aw88081_controls, + .num_controls = ARRAY_SIZE(aw88081_controls), +}; + +static int aw88081_i2c_probe(struct i2c_client *i2c) +{ + struct aw88081 *aw88081; + int ret; + + ret = i2c_check_functionality(i2c->adapter, I2C_FUNC_I2C); + if (!ret) + return dev_err_probe(&i2c->dev, -ENXIO, "check_functionality failed"); + + aw88081 = devm_kzalloc(&i2c->dev, sizeof(*aw88081), GFP_KERNEL); + if (!aw88081) + return -ENOMEM; + + mutex_init(&aw88081->lock); + + i2c_set_clientdata(i2c, aw88081); + + aw88081->regmap = devm_regmap_init_i2c(i2c, &aw88081_regmap_config); + if (IS_ERR(aw88081->regmap)) + return dev_err_probe(&i2c->dev, PTR_ERR(aw88081->regmap), + "failed to init regmap\n"); + + /* aw pa init */ + ret = aw88081_init(aw88081, i2c, aw88081->regmap); + if (ret) + return ret; + + return devm_snd_soc_register_component(&i2c->dev, + &soc_codec_dev_aw88081, + aw88081_dai, ARRAY_SIZE(aw88081_dai)); +} + +static const struct i2c_device_id aw88081_i2c_id[] = { + { AW88081_I2C_NAME }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aw88081_i2c_id); + +static struct i2c_driver aw88081_i2c_driver = { + .driver = { + .name = AW88081_I2C_NAME, + }, + .probe = aw88081_i2c_probe, + .id_table = aw88081_i2c_id, +}; +module_i2c_driver(aw88081_i2c_driver); + +MODULE_DESCRIPTION("ASoC AW88081 Smart PA Driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/aw88081.h b/sound/soc/codecs/aw88081.h new file mode 100644 index 00000000000000..b4bf7288021abc --- /dev/null +++ b/sound/soc/codecs/aw88081.h @@ -0,0 +1,286 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// aw88081.h -- AW88081 ALSA SoC Audio driver +// +// Copyright (c) 2024 awinic Technology CO., LTD +// +// Author: Weidong Wang +// + +#ifndef __AW88081_H__ +#define __AW88081_H__ + +#define AW88081_ID_REG (0x00) +#define AW88081_SYSST_REG (0x01) +#define AW88081_SYSINT_REG (0x02) +#define AW88081_SYSINTM_REG (0x03) +#define AW88081_SYSCTRL_REG (0x04) +#define AW88081_SYSCTRL2_REG (0x05) +#define AW88081_I2SCTRL1_REG (0x06) +#define AW88081_I2SCTRL2_REG (0x07) +#define AW88081_I2SCTRL3_REG (0x08) +#define AW88081_DACCFG1_REG (0x09) +#define AW88081_DACCFG2_REG (0x0A) +#define AW88081_DACCFG3_REG (0x0B) +#define AW88081_DACCFG4_REG (0x0C) +#define AW88081_DACCFG5_REG (0x0D) +#define AW88081_DACCFG6_REG (0x0E) +#define AW88081_DACCFG7_REG (0x11) +#define AW88081_PWMCTRL1_REG (0x13) +#define AW88081_PWMCTRL2_REG (0x14) +#define AW88081_PWMCTRL3_REG (0x15) +#define AW88081_PWMCTRL4_REG (0x16) +#define AW88081_I2SCFG1_REG (0x17) +#define AW88081_DBGCTRL_REG (0x18) +#define AW88081_PDMCTRL_REG (0x19) +#define AW88081_DACST_REG (0x20) +#define AW88081_PATTERNST_REG (0x21) +#define AW88081_I2SINT_REG (0x26) +#define AW88081_I2SCAPCNT_REG (0x27) +#define AW88081_ANASTA1_REG (0x28) +#define AW88081_ANASTA2_REG (0x29) +#define AW88081_ANASTA3_REG (0x2A) +#define AW88081_VBAT_REG (0x21) +#define AW88081_TEMP_REG (0x22) +#define AW88081_PVDD_REG (0x23) +#define AW88081_ISNDAT_REG (0x24) +#define AW88081_VSNDAT_REG (0x25) +#define AW88081_DSMCFG1_REG (0x30) +#define AW88081_DSMCFG2_REG (0x31) +#define AW88081_DSMCFG3_REG (0x32) +#define AW88081_DSMCFG4_REG (0x33) +#define AW88081_DSMCFG5_REG (0x34) +#define AW88081_DSMCFG6_REG (0x35) +#define AW88081_DSMCFG7_REG (0x36) +#define AW88081_DSMCFG8_REG (0x37) +#define AW88081_TESTIN_REG (0x38) +#define AW88081_TESTOUT_REG (0x39) +#define AW88081_BOPCTRL1_REG (0x40) +#define AW88081_BOPCTRL2_REG (0x41) +#define AW88081_BOPCTRL3_REG (0x42) +#define AW88081_BOPSTA_REG (0x43) +#define AW88081_PLLCTRL1_REG (0x54) +#define AW88081_PLLCTRL2_REG (0x55) +#define AW88081_PLLCTRL3_REG (0x56) +#define AW88081_CDACTRL1_REG (0x57) +#define AW88081_CDACTRL2_REG (0x58) +#define AW88081_CDACTRL3_REG (0x59) +#define AW88081_DITHERCFG1_REG (0x5A) +#define AW88081_DITHERCFG2_REG (0x5B) +#define AW88081_DITHERCFG3_REG (0x5C) +#define AW88081_TM_REG (0x6E) +#define AW88081_TM2_REG (0x6F) +#define AW88081_TESTCTRL1_REG (0x70) +#define AW88081_TESTCTRL2_REG (0x71) + +#define AW88081_REG_MAX (0x72) + +#define AW88081_UVLS_START_BIT (14) +#define AW88081_UVLS_UVLO (1) +#define AW88081_UVLS_UVLO_VALUE \ + (AW88081_UVLS_UVLO << AW88081_UVLS_START_BIT) + +#define AW88081_SWS_START_BIT (8) +#define AW88081_SWS_SWITCHING (1) +#define AW88081_SWS_SWITCHING_VALUE \ + (AW88081_SWS_SWITCHING << AW88081_SWS_START_BIT) + +#define AW88081_NOCLKS_START_BIT (5) +#define AW88081_NOCLKS_NO_CLOCK (1) +#define AW88081_NOCLKS_NO_CLOCK_VALUE \ + (AW88081_NOCLKS_NO_CLOCK << AW88081_NOCLKS_START_BIT) + +#define AW88081_CLKS_START_BIT (4) +#define AW88081_CLKS_STABLE (1) +#define AW88081_CLKS_STABLE_VALUE \ + (AW88081_CLKS_STABLE << AW88081_CLKS_START_BIT) + +#define AW88081_OCDS_START_BIT (3) +#define AW88081_OCDS_OC (1) +#define AW88081_OCDS_OC_VALUE \ + (AW88081_OCDS_OC << AW88081_OCDS_START_BIT) + +#define AW88081_OTHS_START_BIT (1) +#define AW88081_OTHS_OT (1) +#define AW88081_OTHS_OT_VALUE \ + (AW88081_OTHS_OT << AW88081_OTHS_START_BIT) + +#define AW88081_PLLS_START_BIT (0) +#define AW88081_PLLS_LOCKED (1) +#define AW88081_PLLS_LOCKED_VALUE \ + (AW88081_PLLS_LOCKED << AW88081_PLLS_START_BIT) + +#define AW88081_BIT_PLL_CHECK \ + (AW88081_CLKS_STABLE_VALUE | \ + AW88081_PLLS_LOCKED_VALUE) + +#define AW88081_BIT_SYSST_CHECK_MASK \ + (~(AW88081_UVLS_UVLO_VALUE | \ + AW88081_SWS_SWITCHING_VALUE | \ + AW88081_NOCLKS_NO_CLOCK_VALUE | \ + AW88081_CLKS_STABLE_VALUE | \ + AW88081_OCDS_OC_VALUE | \ + AW88081_OTHS_OT_VALUE | \ + AW88081_PLLS_LOCKED_VALUE)) + +#define AW88081_NO_SWS_SYSST_CHECK \ + (AW88081_CLKS_STABLE_VALUE | \ + AW88081_PLLS_LOCKED_VALUE) + +#define AW88081_SWS_SYSST_CHECK \ + (AW88081_SWS_SWITCHING_VALUE | \ + AW88081_CLKS_STABLE_VALUE | \ + AW88081_PLLS_LOCKED_VALUE) + +#define AW88081_ULS_HMUTE_START_BIT (14) +#define AW88081_ULS_HMUTE_BITS_LEN (1) +#define AW88081_ULS_HMUTE_MASK \ + (~(((1< Date: Thu, 24 Oct 2024 01:29:13 +0000 Subject: [PATCH 18/97] ASoC: rename rtd->num to rtd->id Current rtd has "num". It sounds/looks like size of rtd or something, but it will be mainly used at snd_pcm_new() as "device index". This naming is confusable. Let's rename it to "id" Some drivers are using rtd->num, so let's keep it so far, and remove it if all user was switched. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87zfmub85z.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 4 ++-- include/sound/soc.h | 9 +++++---- sound/soc/soc-compress.c | 10 +++++----- sound/soc/soc-core.c | 15 ++++++++------- sound/soc/soc-dai.c | 4 ++-- sound/soc/soc-pcm.c | 16 ++++++++-------- 6 files changed, 30 insertions(+), 28 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 0d1b215f24f4f0..9dbeedf6da13bc 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -217,7 +217,7 @@ void snd_soc_dai_shutdown(struct snd_soc_dai *dai, void snd_soc_dai_suspend(struct snd_soc_dai *dai); void snd_soc_dai_resume(struct snd_soc_dai *dai); int snd_soc_dai_compress_new(struct snd_soc_dai *dai, - struct snd_soc_pcm_runtime *rtd, int num); + struct snd_soc_pcm_runtime *rtd, int id); bool snd_soc_dai_stream_valid(const struct snd_soc_dai *dai, int stream); void snd_soc_dai_action(struct snd_soc_dai *dai, int stream, int action); @@ -275,7 +275,7 @@ struct snd_soc_dai_ops { int (*probe)(struct snd_soc_dai *dai); int (*remove)(struct snd_soc_dai *dai); /* compress dai */ - int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num); + int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int id); /* Optional Callback used at pcm creation*/ int (*pcm_new)(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); diff --git a/include/sound/soc.h b/include/sound/soc.h index 5c240ea340276a..51840ceb3cd420 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -486,11 +486,11 @@ struct snd_soc_component *snd_soc_lookup_component_nolocked(struct device *dev, struct snd_soc_component *snd_soc_lookup_component(struct device *dev, const char *driver_name); -int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int id); #ifdef CONFIG_SND_SOC_COMPRESS -int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int id); #else -static inline int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +static inline int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int id) { return 0; } @@ -1195,7 +1195,8 @@ struct snd_soc_pcm_runtime { struct dentry *debugfs_dpcm_root; #endif - unsigned int num; /* 0-based and monotonic increasing */ + unsigned int num; /* REMOVE ME */ + unsigned int id; /* 0-based and monotonic increasing */ struct list_head list; /* rtd list of the soc card */ /* function mark */ diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index a0c55246f424b6..fb664c775dda50 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -537,11 +537,11 @@ static struct snd_compr_ops soc_compr_dyn_ops = { * snd_soc_new_compress - create a new compress. * * @rtd: The runtime for which we will create compress - * @num: the device index number (zero based - shared with normal PCMs) + * @id: the device index number (zero based - shared with normal PCMs) * * Return: 0 for success, else error. */ -int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int id) { struct snd_soc_component *component; struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); @@ -617,7 +617,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) snprintf(new_name, sizeof(new_name), "(%s)", rtd->dai_link->stream_name); - ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, + ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, id, playback, capture, &be_pcm); if (ret < 0) { dev_err(rtd->card->dev, @@ -638,7 +638,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else { snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); + rtd->dai_link->stream_name, codec_dai->name, id); memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); } @@ -652,7 +652,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) break; } - ret = snd_compress_new(rtd->card->snd_card, num, direction, + ret = snd_compress_new(rtd->card->snd_card, id, direction, new_name, compr); if (ret < 0) { component = snd_soc_rtd_to_codec(rtd, 0)->component; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f04b671ce33ea6..3cb7482791669c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -558,7 +558,8 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( */ rtd->card = card; rtd->dai_link = dai_link; - rtd->num = card->num_rtd++; + rtd->id = card->num_rtd++; + rtd->num = rtd->id; /* REMOVE ME */ rtd->pmdown_time = pmdown_time; /* default power off timeout */ /* see for_each_card_rtds */ @@ -1458,7 +1459,7 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct snd_soc_component *component; - int ret, num, i; + int ret, id, i; /* do machine specific initialization */ ret = snd_soc_link_init(rtd); @@ -1473,7 +1474,7 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, /* add DPCM sysfs entries */ soc_dpcm_debugfs_add(rtd); - num = rtd->num; + id = rtd->id; /* * most drivers will register their PCMs using DAI link ordering but @@ -1485,18 +1486,18 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, continue; if (rtd->dai_link->no_pcm) - num += component->driver->be_pcm_base; + id += component->driver->be_pcm_base; else - num = rtd->dai_link->id; + id = rtd->dai_link->id; } /* create compress_device if possible */ - ret = snd_soc_dai_compress_new(cpu_dai, rtd, num); + ret = snd_soc_dai_compress_new(cpu_dai, rtd, id); if (ret != -ENOTSUPP) goto err; /* create the pcm */ - ret = soc_new_pcm(rtd, num); + ret = soc_new_pcm(rtd, id); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 4a1c85ad5a8d60..2feb76bf57bb72 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -457,12 +457,12 @@ void snd_soc_dai_shutdown(struct snd_soc_dai *dai, } int snd_soc_dai_compress_new(struct snd_soc_dai *dai, - struct snd_soc_pcm_runtime *rtd, int num) + struct snd_soc_pcm_runtime *rtd, int id) { int ret = -ENOTSUPP; if (dai->driver->ops && dai->driver->ops->compress_new) - ret = dai->driver->ops->compress_new(rtd, num); + ret = dai->driver->ops->compress_new(rtd, id); return soc_dai_ret(dai, ret); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 678400e76e53b7..81b63e547a0996 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2891,7 +2891,7 @@ static int soc_get_playback_capture(struct snd_soc_pcm_runtime *rtd, static int soc_create_pcm(struct snd_pcm **pcm, struct snd_soc_pcm_runtime *rtd, - int playback, int capture, int num) + int playback, int capture, int id) { char new_name[64]; int ret; @@ -2901,13 +2901,13 @@ static int soc_create_pcm(struct snd_pcm **pcm, snprintf(new_name, sizeof(new_name), "codec2codec(%s)", rtd->dai_link->stream_name); - ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, + ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, id, playback, capture, pcm); } else if (rtd->dai_link->no_pcm) { snprintf(new_name, sizeof(new_name), "(%s)", rtd->dai_link->stream_name); - ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, + ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, id, playback, capture, pcm); } else { if (rtd->dai_link->dynamic) @@ -2916,9 +2916,9 @@ static int soc_create_pcm(struct snd_pcm **pcm, else snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, - soc_codec_dai_name(rtd), num); + soc_codec_dai_name(rtd), id); - ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback, + ret = snd_pcm_new(rtd->card->snd_card, new_name, id, playback, capture, pcm); } if (ret < 0) { @@ -2926,13 +2926,13 @@ static int soc_create_pcm(struct snd_pcm **pcm, new_name, rtd->dai_link->name, ret); return ret; } - dev_dbg(rtd->card->dev, "ASoC: registered pcm #%d %s\n",num, new_name); + dev_dbg(rtd->card->dev, "ASoC: registered pcm #%d %s\n", id, new_name); return 0; } /* create a new pcm */ -int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int id) { struct snd_soc_component *component; struct snd_pcm *pcm; @@ -2943,7 +2943,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (ret < 0) return ret; - ret = soc_create_pcm(&pcm, rtd, playback, capture, num); + ret = soc_create_pcm(&pcm, rtd, playback, capture, id); if (ret < 0) return ret; From eae33f737c7a929d92b559fe1a1002d597b7b903 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 24 Oct 2024 01:29:20 +0000 Subject: [PATCH 19/97] ASoC: fsl: switch to use rtd->id from rtd->num Now rtd->num is renamed to rtd->id. Let's switch. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87y12eb85r.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/fsl/imx-card.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index 0f11f20dc51a43..95a57fda025039 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -275,7 +275,7 @@ static unsigned long akcodec_get_mclk_rate(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct imx_card_data *data = snd_soc_card_get_drvdata(rtd->card); const struct imx_card_plat_data *plat_data = data->plat_data; - struct dai_link_data *link_data = &data->link_data[rtd->num]; + struct dai_link_data *link_data = &data->link_data[rtd->id]; unsigned int width = slots * slot_width; unsigned int rate = params_rate(params); int i; @@ -313,7 +313,7 @@ static int imx_aif_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct snd_soc_card *card = rtd->card; struct imx_card_data *data = snd_soc_card_get_drvdata(card); - struct dai_link_data *link_data = &data->link_data[rtd->num]; + struct dai_link_data *link_data = &data->link_data[rtd->id]; struct imx_card_plat_data *plat_data = data->plat_data; struct device *dev = card->dev; struct snd_soc_dai *codec_dai; @@ -435,7 +435,7 @@ static int imx_aif_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct imx_card_data *data = snd_soc_card_get_drvdata(card); - struct dai_link_data *link_data = &data->link_data[rtd->num]; + struct dai_link_data *link_data = &data->link_data[rtd->id]; static struct snd_pcm_hw_constraint_list constraint_rates; static struct snd_pcm_hw_constraint_list constraint_channels; int ret = 0; From b19f75df8fa9f8d4aa8b5886dca0f2d832a76baa Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 24 Oct 2024 01:29:27 +0000 Subject: [PATCH 20/97] ASoC: meson: switch to use rtd->id from rtd->num Now rtd->num is renamed to rtd->id. Let's switch. Signed-off-by: Kuninori Morimoto Acked-by: Jerome Brunet Link: https://patch.msgid.link/87wmhyb85l.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-card.c | 6 +++--- sound/soc/meson/gx-card.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 5ebf287fe7004e..a2dfccb7990f3a 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -43,7 +43,7 @@ static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); struct axg_dai_link_tdm_data *be = - (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->id]; return meson_card_i2s_set_sysclk(substream, params, be->mclk_fs); } @@ -56,7 +56,7 @@ static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd) { struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); struct axg_dai_link_tdm_data *be = - (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->id]; struct snd_soc_dai *codec_dai; int ret, i; @@ -86,7 +86,7 @@ static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd) { struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); struct axg_dai_link_tdm_data *be = - (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->id]; int ret; /* The loopback rx_mask is the pad tx_mask */ diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c index 455f6bfc9f8fa5..b408cc2bbc9193 100644 --- a/sound/soc/meson/gx-card.c +++ b/sound/soc/meson/gx-card.c @@ -32,7 +32,7 @@ static int gx_card_i2s_be_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); struct gx_dai_link_i2s_data *be = - (struct gx_dai_link_i2s_data *)priv->link_data[rtd->num]; + (struct gx_dai_link_i2s_data *)priv->link_data[rtd->id]; return meson_card_i2s_set_sysclk(substream, params, be->mclk_fs); } From 970a874b76d09d6a5880e8832e572850cfcb4008 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 24 Oct 2024 01:29:34 +0000 Subject: [PATCH 21/97] ASoC: sh: switch to use rtd->id from rtd->num Now rtd->num is renamed to rtd->id. Let's switch. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87v7xib85e.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/renesas/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/renesas/rcar/core.c b/sound/soc/renesas/rcar/core.c index c32e88d6a141ed..e2234928c9e881 100644 --- a/sound/soc/renesas/rcar/core.c +++ b/sound/soc/renesas/rcar/core.c @@ -1843,7 +1843,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = name, .info = rsnd_kctrl_info, - .index = rtd->num, + .index = rtd->id, .get = rsnd_kctrl_get, .put = rsnd_kctrl_put, }; From 742e622db67efc32affb5893fdcc0149f374533e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 24 Oct 2024 01:29:39 +0000 Subject: [PATCH 22/97] ASoC: generic: switch to use rtd->id from rtd->num Now rtd->num is renamed to rtd->id. Let's switch. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87ttd2b858.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index fedae7f6f70cc5..d47c372228b34c 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -296,7 +296,7 @@ int simple_util_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct simple_util_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *props = simple_priv_to_props(priv, rtd->num); + struct simple_dai_props *props = simple_priv_to_props(priv, rtd->id); struct simple_util_dai *dai; unsigned int fixed_sysclk = 0; int i1, i2, i; @@ -357,7 +357,7 @@ void simple_util_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct simple_util_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *props = simple_priv_to_props(priv, rtd->num); + struct simple_dai_props *props = simple_priv_to_props(priv, rtd->id); struct simple_util_dai *dai; int i; @@ -448,7 +448,7 @@ int simple_util_hw_params(struct snd_pcm_substream *substream, struct simple_util_dai *pdai; struct snd_soc_dai *sdai; struct simple_util_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *props = simple_priv_to_props(priv, rtd->num); + struct simple_dai_props *props = simple_priv_to_props(priv, rtd->id); unsigned int mclk, mclk_fs = 0; int i, ret; @@ -517,7 +517,7 @@ int simple_util_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct simple_util_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->id); struct simple_util_data *data = &dai_props->adata; struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); @@ -628,7 +628,7 @@ static int simple_init_for_codec2codec(struct snd_soc_pcm_runtime *rtd, int simple_util_dai_init(struct snd_soc_pcm_runtime *rtd) { struct simple_util_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *props = simple_priv_to_props(priv, rtd->num); + struct simple_dai_props *props = simple_priv_to_props(priv, rtd->id); struct simple_util_dai *dai; int i, ret; From c59db5ed233a19f6aadd086fb89149ec5f6fa855 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 24 Oct 2024 01:29:45 +0000 Subject: [PATCH 23/97] ASoC: remove rtd->num No one is using rtd->num. Let's remove it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87sesmb852.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 1 - sound/soc/soc-core.c | 1 - 2 files changed, 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 51840ceb3cd420..21a50a8057eb60 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1195,7 +1195,6 @@ struct snd_soc_pcm_runtime { struct dentry *debugfs_dpcm_root; #endif - unsigned int num; /* REMOVE ME */ unsigned int id; /* 0-based and monotonic increasing */ struct list_head list; /* rtd list of the soc card */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3cb7482791669c..233c91e60f0cbb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -559,7 +559,6 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( rtd->card = card; rtd->dai_link = dai_link; rtd->id = card->num_rtd++; - rtd->num = rtd->id; /* REMOVE ME */ rtd->pmdown_time = pmdown_time; /* default power off timeout */ /* see for_each_card_rtds */ From cb18cd26039f5cdecb0ac53fb447b6f0859f3d1c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 24 Oct 2024 01:29:52 +0000 Subject: [PATCH 24/97] ASoC: soc-core: do rtd->id trick at snd_soc_add_pcm_runtime() qcom/qdsp6 want to use irregular rtd->id because of its topology. Current code is calculating it at soc_init_pcm_runtime() which calls soc_new_pcm(), and it doesn't save it to rtd->id. Let's calculate and save it to rtd at snd_soc_add_pcm_runtime() which create rtd and connect related components. But, this feature should be implemented by using "dai_link" instead of "component". Add FIXME as comment. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87r086b84w.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 42 ++++++++++++++++++++++++------------------ 1 file changed, 24 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 233c91e60f0cbb..4f0bfe73fe15ef 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1166,7 +1166,7 @@ static int snd_soc_add_pcm_runtime(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai_link_component *codec, *platform, *cpu; struct snd_soc_component *component; - int i, ret; + int i, id, ret; lockdep_assert_held(&client_mutex); @@ -1225,6 +1225,28 @@ static int snd_soc_add_pcm_runtime(struct snd_soc_card *card, } } + /* + * Most drivers will register their PCMs using DAI link ordering but + * topology based drivers can use the DAI link id field to set PCM + * device number and then use rtd + a base offset of the BEs. + * + * FIXME + * + * This should be implemented by using "dai_link" feature instead of + * "component" feature. + */ + id = rtd->id; + for_each_rtd_components(rtd, i, component) { + if (!component->driver->use_dai_pcm_id) + continue; + + if (rtd->dai_link->no_pcm) + id += component->driver->be_pcm_base; + else + id = rtd->dai_link->id; + } + rtd->id = id; + return 0; _err_defer: @@ -1457,8 +1479,7 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); - struct snd_soc_component *component; - int ret, id, i; + int ret, id; /* do machine specific initialization */ ret = snd_soc_link_init(rtd); @@ -1475,21 +1496,6 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, id = rtd->id; - /* - * most drivers will register their PCMs using DAI link ordering but - * topology based drivers can use the DAI link id field to set PCM - * device number and then use rtd + a base offset of the BEs. - */ - for_each_rtd_components(rtd, i, component) { - if (!component->driver->use_dai_pcm_id) - continue; - - if (rtd->dai_link->no_pcm) - id += component->driver->be_pcm_base; - else - id = rtd->dai_link->id; - } - /* create compress_device if possible */ ret = snd_soc_dai_compress_new(cpu_dai, rtd, id); if (ret != -ENOTSUPP) From 8b12da9a18f4dd53e4b3a7393829a555e84f073c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 24 Oct 2024 01:29:58 +0000 Subject: [PATCH 25/97] ASoC: cleanup function parameter for rtd and its id some functions had parameter like below xxx(..., rtd, ..., id); This "id" is rtd->id. We don't need to have "id" on each functions because we can get it from "rtd". Let's cleanup it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87plnqb84p.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 5 ++--- include/sound/soc.h | 6 +++--- sound/soc/soc-compress.c | 9 ++++----- sound/soc/soc-core.c | 8 +++----- sound/soc/soc-dai.c | 4 ++-- sound/soc/soc-pcm.c | 16 ++++++++-------- 6 files changed, 22 insertions(+), 26 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 9dbeedf6da13bc..b275201b02f608 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -216,8 +216,7 @@ void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, int rollback); void snd_soc_dai_suspend(struct snd_soc_dai *dai); void snd_soc_dai_resume(struct snd_soc_dai *dai); -int snd_soc_dai_compress_new(struct snd_soc_dai *dai, - struct snd_soc_pcm_runtime *rtd, int id); +int snd_soc_dai_compress_new(struct snd_soc_dai *dai, struct snd_soc_pcm_runtime *rtd); bool snd_soc_dai_stream_valid(const struct snd_soc_dai *dai, int stream); void snd_soc_dai_action(struct snd_soc_dai *dai, int stream, int action); @@ -275,7 +274,7 @@ struct snd_soc_dai_ops { int (*probe)(struct snd_soc_dai *dai); int (*remove)(struct snd_soc_dai *dai); /* compress dai */ - int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int id); + int (*compress_new)(struct snd_soc_pcm_runtime *rtd); /* Optional Callback used at pcm creation*/ int (*pcm_new)(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); diff --git a/include/sound/soc.h b/include/sound/soc.h index 21a50a8057eb60..4f5d411e3823f6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -486,11 +486,11 @@ struct snd_soc_component *snd_soc_lookup_component_nolocked(struct device *dev, struct snd_soc_component *snd_soc_lookup_component(struct device *dev, const char *driver_name); -int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int id); +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd); #ifdef CONFIG_SND_SOC_COMPRESS -int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int id); +int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd); #else -static inline int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int id) +static inline int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd) { return 0; } diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index fb664c775dda50..3c514703fa33d6 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -537,11 +537,10 @@ static struct snd_compr_ops soc_compr_dyn_ops = { * snd_soc_new_compress - create a new compress. * * @rtd: The runtime for which we will create compress - * @id: the device index number (zero based - shared with normal PCMs) * * Return: 0 for success, else error. */ -int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int id) +int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component; struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); @@ -617,7 +616,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int id) snprintf(new_name, sizeof(new_name), "(%s)", rtd->dai_link->stream_name); - ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, id, + ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, rtd->id, playback, capture, &be_pcm); if (ret < 0) { dev_err(rtd->card->dev, @@ -638,7 +637,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int id) memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else { snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, id); + rtd->dai_link->stream_name, codec_dai->name, rtd->id); memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); } @@ -652,7 +651,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int id) break; } - ret = snd_compress_new(rtd->card->snd_card, id, direction, + ret = snd_compress_new(rtd->card->snd_card, rtd->id, direction, new_name, compr); if (ret < 0) { component = snd_soc_rtd_to_codec(rtd, 0)->component; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4f0bfe73fe15ef..a1dace4bb61664 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1479,7 +1479,7 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); - int ret, id; + int ret; /* do machine specific initialization */ ret = snd_soc_link_init(rtd); @@ -1494,15 +1494,13 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, /* add DPCM sysfs entries */ soc_dpcm_debugfs_add(rtd); - id = rtd->id; - /* create compress_device if possible */ - ret = snd_soc_dai_compress_new(cpu_dai, rtd, id); + ret = snd_soc_dai_compress_new(cpu_dai, rtd); if (ret != -ENOTSUPP) goto err; /* create the pcm */ - ret = soc_new_pcm(rtd, id); + ret = soc_new_pcm(rtd); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 2feb76bf57bb72..34ba1a93a4c95a 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -457,12 +457,12 @@ void snd_soc_dai_shutdown(struct snd_soc_dai *dai, } int snd_soc_dai_compress_new(struct snd_soc_dai *dai, - struct snd_soc_pcm_runtime *rtd, int id) + struct snd_soc_pcm_runtime *rtd) { int ret = -ENOTSUPP; if (dai->driver->ops && dai->driver->ops->compress_new) - ret = dai->driver->ops->compress_new(rtd, id); + ret = dai->driver->ops->compress_new(rtd); return soc_dai_ret(dai, ret); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 81b63e547a0996..fb7f25fd8ec5bd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2891,7 +2891,7 @@ static int soc_get_playback_capture(struct snd_soc_pcm_runtime *rtd, static int soc_create_pcm(struct snd_pcm **pcm, struct snd_soc_pcm_runtime *rtd, - int playback, int capture, int id) + int playback, int capture) { char new_name[64]; int ret; @@ -2901,13 +2901,13 @@ static int soc_create_pcm(struct snd_pcm **pcm, snprintf(new_name, sizeof(new_name), "codec2codec(%s)", rtd->dai_link->stream_name); - ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, id, + ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, rtd->id, playback, capture, pcm); } else if (rtd->dai_link->no_pcm) { snprintf(new_name, sizeof(new_name), "(%s)", rtd->dai_link->stream_name); - ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, id, + ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, rtd->id, playback, capture, pcm); } else { if (rtd->dai_link->dynamic) @@ -2916,9 +2916,9 @@ static int soc_create_pcm(struct snd_pcm **pcm, else snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, - soc_codec_dai_name(rtd), id); + soc_codec_dai_name(rtd), rtd->id); - ret = snd_pcm_new(rtd->card->snd_card, new_name, id, playback, + ret = snd_pcm_new(rtd->card->snd_card, new_name, rtd->id, playback, capture, pcm); } if (ret < 0) { @@ -2926,13 +2926,13 @@ static int soc_create_pcm(struct snd_pcm **pcm, new_name, rtd->dai_link->name, ret); return ret; } - dev_dbg(rtd->card->dev, "ASoC: registered pcm #%d %s\n", id, new_name); + dev_dbg(rtd->card->dev, "ASoC: registered pcm #%d %s\n", rtd->id, new_name); return 0; } /* create a new pcm */ -int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int id) +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component; struct snd_pcm *pcm; @@ -2943,7 +2943,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int id) if (ret < 0) return ret; - ret = soc_create_pcm(&pcm, rtd, playback, capture, id); + ret = soc_create_pcm(&pcm, rtd, playback, capture); if (ret < 0) return ret; From 953e549471cabc9d4980f1da2e9fa79f4c23da06 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Fri, 1 Nov 2024 18:55:53 +0200 Subject: [PATCH 26/97] regmap: irq: Set lockdep class for hierarchical IRQ domains Lockdep gives a false positive splat as it can't distinguish the lock which is taken by different IRQ descriptors from different IRQ chips that are organized in a way of a hierarchy: ====================================================== WARNING: possible circular locking dependency detected 6.12.0-rc5-next-20241101-00148-g9fabf8160b53 #562 Tainted: G W ------------------------------------------------------ modprobe/141 is trying to acquire lock: ffff899446947868 (intel_soc_pmic_bxtwc:502:(&bxtwc_regmap_config)->lock){+.+.}-{4:4}, at: regmap_update_bits_base+0x33/0x90 but task is already holding lock: ffff899446947c68 (&d->lock){+.+.}-{4:4}, at: __setup_irq+0x682/0x790 which lock already depends on the new lock. -> #3 (&d->lock){+.+.}-{4:4}: -> #2 (&desc->request_mutex){+.+.}-{4:4}: -> #1 (ipclock){+.+.}-{4:4}: -> #0 (intel_soc_pmic_bxtwc:502:(&bxtwc_regmap_config)->lock){+.+.}-{4:4}: Chain exists of: intel_soc_pmic_bxtwc:502:(&bxtwc_regmap_config)->lock --> &desc->request_mutex --> &d->lock Possible unsafe locking scenario: CPU0 CPU1 ---- ---- lock(&d->lock); lock(&desc->request_mutex); lock(&d->lock); lock(intel_soc_pmic_bxtwc:502:(&bxtwc_regmap_config)->lock); *** DEADLOCK *** 3 locks held by modprobe/141: #0: ffff8994419368f8 (&dev->mutex){....}-{4:4}, at: __driver_attach+0xf6/0x250 #1: ffff89944690b250 (&desc->request_mutex){+.+.}-{4:4}, at: __setup_irq+0x1a2/0x790 #2: ffff899446947c68 (&d->lock){+.+.}-{4:4}, at: __setup_irq+0x682/0x790 Set a lockdep class when we map the IRQ so that it doesn't warn about a lockdep bug that doesn't exist. Fixes: 4af8be67fd99 ("regmap: Convert regmap_irq to use irq_domain") Signed-off-by: Andy Shevchenko Link: https://patch.msgid.link/20241101165553.4055617-1-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- drivers/base/regmap/regmap-irq.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/drivers/base/regmap/regmap-irq.c b/drivers/base/regmap/regmap-irq.c index 33ec28e3a80252..0bcd81389a29f8 100644 --- a/drivers/base/regmap/regmap-irq.c +++ b/drivers/base/regmap/regmap-irq.c @@ -511,12 +511,16 @@ static irqreturn_t regmap_irq_thread(int irq, void *d) return IRQ_NONE; } +static struct lock_class_key regmap_irq_lock_class; +static struct lock_class_key regmap_irq_request_class; + static int regmap_irq_map(struct irq_domain *h, unsigned int virq, irq_hw_number_t hw) { struct regmap_irq_chip_data *data = h->host_data; irq_set_chip_data(virq, data); + irq_set_lockdep_class(virq, ®map_irq_lock_class, ®map_irq_request_class); irq_set_chip(virq, &data->irq_chip); irq_set_nested_thread(virq, 1); irq_set_parent(virq, data->irq); From c2d188e137e77294323132a760a4608321a36a70 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Nov 2024 11:07:34 +0100 Subject: [PATCH 27/97] ALSA: ump: Don't enumeration invalid groups for legacy rawmidi The legacy rawmidi tries to enumerate all possible UMP groups belonging to the UMP endpoint. But currently it shows all 16 ports when the UMP endpoint is configured with static blocks, although most of them may be unused. There was already a fix for the sequencer client side to ignore such groups in the commit 3bfd7c0ba184 ("ALSA: seq: ump: Skip useless ports for static blocks"), and this commit is a similar fix for UMP rawmidi devices; it adds simply the check for the validity of each group that has been already parsed. (Note that the group info was moved to snd_ump_endpoint.groups[] by the commit 0642a3c5cacc0321c755 ("ALSA: ump: Update substream name from assigned FB names")). Link: https://patch.msgid.link/20241104100735.16127-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/ump.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/ump.c b/sound/core/ump.c index cf22a17e38dd50..7d59a0a9b037ad 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -1233,7 +1233,7 @@ static int fill_legacy_mapping(struct snd_ump_endpoint *ump) num = 0; for (i = 0; i < SNDRV_UMP_MAX_GROUPS; i++) - if (group_maps & (1U << i)) + if ((group_maps & (1U << i)) && ump->groups[i].valid) ump->legacy_mapping[num++] = i; return num; From 8abbf1f01d6a2ef9f911f793e30f7382154b5a3a Mon Sep 17 00:00:00 2001 From: Murad Masimov Date: Fri, 1 Nov 2024 21:55:13 +0300 Subject: [PATCH 28/97] ALSA: firewire-lib: fix return value on fail in amdtp_tscm_init() If amdtp_stream_init() fails in amdtp_tscm_init(), the latter returns zero, though it's supposed to return error code, which is checked inside init_stream() in file tascam-stream.c. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 47faeea25ef3 ("ALSA: firewire-tascam: add data block processing layer") Signed-off-by: Murad Masimov Reviewed-by: Takashi Sakamoto Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20241101185517.1819-1-m.masimov@maxima.ru --- sound/firewire/tascam/amdtp-tascam.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c index 0b42d65590081a..079afa4bd3811b 100644 --- a/sound/firewire/tascam/amdtp-tascam.c +++ b/sound/firewire/tascam/amdtp-tascam.c @@ -238,7 +238,7 @@ int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit, err = amdtp_stream_init(s, unit, dir, flags, fmt, process_ctx_payloads, sizeof(struct amdtp_tscm)); if (err < 0) - return 0; + return err; if (dir == AMDTP_OUT_STREAM) { // Use fixed value for FDF field. From cac99f73f0752e1c83674e12fb2c605dca9ce474 Mon Sep 17 00:00:00 2001 From: Heiner Kallweit Date: Thu, 31 Oct 2024 20:32:52 +0100 Subject: [PATCH 29/97] ALSA: hda: intel: Don't free interrupt when suspending There's no need to free/re-request the interrupt on system suspend. PCI core takes care, using functions like pci_restore_msi_state(). Signed-off-by: Heiner Kallweit Link: https://patch.msgid.link/1b7e109b-eb69-4542-8022-4ac8f9116474@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b4540c5cd2a6f9..9fc5e6c5d80030 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1040,14 +1040,6 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); azx_shutdown_chip(chip); - if (bus->irq >= 0) { - free_irq(bus->irq, chip); - bus->irq = -1; - chip->card->sync_irq = -1; - } - - if (chip->msi) - pci_disable_msi(chip->pci); trace_azx_suspend(chip); return 0; @@ -1062,11 +1054,6 @@ static int __maybe_unused azx_resume(struct device *dev) return 0; chip = card->private_data; - if (chip->msi) - if (pci_enable_msi(chip->pci) < 0) - chip->msi = 0; - if (azx_acquire_irq(chip, 1) < 0) - return -EIO; __azx_runtime_resume(chip); From 149cb7d9537e241b43056fb4133f56832ac51b7a Mon Sep 17 00:00:00 2001 From: Heiner Kallweit Date: Thu, 31 Oct 2024 20:41:12 +0100 Subject: [PATCH 30/97] ALSA: hda: intel: Switch to pci_alloc_irq_vectors API Switch from legacy pci_msi_enable()/pci_intx() API to the pci_alloc_irq_vectors API. Signed-off-by: Heiner Kallweit Link: https://patch.msgid.link/11c60429-9435-4666-8e27-77160abef68e@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9fc5e6c5d80030..fc329b6a70f55c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -773,6 +773,14 @@ static void azx_clear_irq_pending(struct azx *chip) static int azx_acquire_irq(struct azx *chip, int do_disconnect) { struct hdac_bus *bus = azx_bus(chip); + int ret; + + if (!chip->msi || pci_alloc_irq_vectors(chip->pci, 1, 1, PCI_IRQ_MSI) < 0) { + ret = pci_alloc_irq_vectors(chip->pci, 1, 1, PCI_IRQ_INTX); + if (ret < 0) + return ret; + chip->msi = 0; + } if (request_irq(chip->pci->irq, azx_interrupt, chip->msi ? 0 : IRQF_SHARED, @@ -786,7 +794,6 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect) } bus->irq = chip->pci->irq; chip->card->sync_irq = bus->irq; - pci_intx(chip->pci, !chip->msi); return 0; } @@ -1879,13 +1886,9 @@ static int azx_first_init(struct azx *chip) chip->gts_present = true; #endif - if (chip->msi) { - if (chip->driver_caps & AZX_DCAPS_NO_MSI64) { - dev_dbg(card->dev, "Disabling 64bit MSI\n"); - pci->no_64bit_msi = true; - } - if (pci_enable_msi(pci) < 0) - chip->msi = 0; + if (chip->msi && chip->driver_caps & AZX_DCAPS_NO_MSI64) { + dev_dbg(card->dev, "Disabling 64bit MSI\n"); + pci->no_64bit_msi = true; } pci_set_master(pci); @@ -2037,7 +2040,7 @@ static int disable_msi_reset_irq(struct azx *chip) free_irq(bus->irq, chip); bus->irq = -1; chip->card->sync_irq = -1; - pci_disable_msi(chip->pci); + pci_free_irq_vectors(chip->pci); chip->msi = 0; err = azx_acquire_irq(chip, 1); if (err < 0) From fe09de2db2365eed8b44b572cff7d421eaf1754a Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Mon, 4 Nov 2024 18:00:55 +0800 Subject: [PATCH 31/97] ASoC: tas2781: Add new driver version for tas2563 & tas2781 qfn chip Add new driver version to support tas2563 & tas2781 qfn chip Signed-off-by: Shenghao Ding Link: https://patch.msgid.link/20241104100055.48-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2781-fmwlib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index ae360c97fe1efb..0aeb88abbf52f1 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1992,6 +1992,7 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, break; case 0x202: case 0x400: + case 0x401: tas_priv->fw_parse_variable_header = fw_parse_variable_header_git; tas_priv->fw_parse_program_data = From 8ae4c65d7ae82fead83202448453e47078ddfde7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Nov 2024 20:06:53 +0100 Subject: [PATCH 32/97] ALSA: hda: Fix unused variable warning The previous code cleanup made a variable not really used, which now leads to a compile warning. Let's fix it. Fixes: cac99f73f075 ("ALSA: hda: intel: Don't free interrupt when suspending") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202411050247.3esQz7Am-lkp@intel.com/ Link: https://patch.msgid.link/20241104190654.32216-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fc329b6a70f55c..6e271777feb9c9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1039,13 +1039,11 @@ static int azx_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; - struct hdac_bus *bus; if (!azx_is_pm_ready(card)) return 0; chip = card->private_data; - bus = azx_bus(chip); azx_shutdown_chip(chip); trace_azx_suspend(chip); From dabc44c28f118910dea96244d903f0c270225669 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2024 13:02:17 +0100 Subject: [PATCH 33/97] ALSA: usb-audio: Add quirk for HP 320 FHD Webcam HP 320 FHD Webcam (03f0:654a) seems to have flaky firmware like other webcam devices that don't like the frequency inquiries. Also, Mic Capture Volume has an invalid resolution, hence fix it to be 16 (as a blind shot). Link: https://bugzilla.suse.com/show_bug.cgi?id=1232768 Cc: Link: https://patch.msgid.link/20241105120220.5740-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 1 + sound/usb/quirks.c | 2 ++ 2 files changed, 3 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 9945ae55b0d08b..bd67027c767751 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1205,6 +1205,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, } break; case USB_ID(0x1bcf, 0x2283): /* NexiGo N930AF FHD Webcam */ + case USB_ID(0x03f0, 0x654a): /* HP 320 FHD Webcam */ if (!strcmp(kctl->id.name, "Mic Capture Volume")) { usb_audio_info(chip, "set resolution quirk: cval->res = 16\n"); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e6278a24579559..c5fd180357d1e8 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2114,6 +2114,8 @@ struct usb_audio_quirk_flags_table { static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { /* Device matches */ + DEVICE_FLG(0x03f0, 0x654a, /* HP 320 FHD Webcam */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x041e, 0x3000, /* Creative SB Extigy */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x041e, 0x4080, /* Creative Live Cam VF0610 */ From d6e6b9218ced5249b9136833ef5ec3f554ec7fde Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2024 13:02:18 +0100 Subject: [PATCH 34/97] ALSA: usb-audio: Make mic volume workarounds globally applicable It seems that many webcams have buggy firmware and don't expose the mic capture volume with the proper resolution. We have workarounds in mixer.c, but judging from the numbers, those can be better managed as global quirk flags. Link: https://patch.msgid.link/20241105120220.5740-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 58 ++++++++++++-------------------------------- sound/usb/quirks.c | 31 +++++++++++++++++++---- sound/usb/usbaudio.h | 4 +++ 3 files changed, 45 insertions(+), 48 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 341b32f5ddd0c1..66976be06bc088 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1084,6 +1084,21 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, struct snd_kcontrol *kctl) { struct snd_usb_audio *chip = cval->head.mixer->chip; + + if (chip->quirk_flags & QUIRK_FLAG_MIC_RES_384) { + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + usb_audio_info(chip, + "set resolution quirk: cval->res = 384\n"); + cval->res = 384; + } + } else if (chip->quirk_flags & QUIRK_FLAG_MIC_RES_16) { + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + usb_audio_info(chip, + "set resolution quirk: cval->res = 16\n"); + cval->res = 16; + } + } + switch (chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ @@ -1168,27 +1183,6 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, } break; - case USB_ID(0x046d, 0x0807): /* Logitech Webcam C500 */ - case USB_ID(0x046d, 0x0808): - case USB_ID(0x046d, 0x0809): - case USB_ID(0x046d, 0x0819): /* Logitech Webcam C210 */ - case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ - case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ - case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ - case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ - case USB_ID(0x046d, 0x08ca): /* Logitech Quickcam Fusion */ - case USB_ID(0x046d, 0x0991): - case USB_ID(0x046d, 0x09a2): /* QuickCam Communicate Deluxe/S7500 */ - /* Most audio usb devices lie about volume resolution. - * Most Logitech webcams have res = 384. - * Probably there is some logitech magic behind this number --fishor - */ - if (!strcmp(kctl->id.name, "Mic Capture Volume")) { - usb_audio_info(chip, - "set resolution quirk: cval->res = 384\n"); - cval->res = 384; - } - break; case USB_ID(0x0495, 0x3042): /* ESS Technology Asus USB DAC */ if ((strstr(kctl->id.name, "Playback Volume") != NULL) || strstr(kctl->id.name, "Capture Volume") != NULL) { @@ -1197,28 +1191,6 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 1; } break; - case USB_ID(0x1224, 0x2a25): /* Jieli Technology USB PHY 2.0 */ - if (!strcmp(kctl->id.name, "Mic Capture Volume")) { - usb_audio_info(chip, - "set resolution quirk: cval->res = 16\n"); - cval->res = 16; - } - break; - case USB_ID(0x1bcf, 0x2283): /* NexiGo N930AF FHD Webcam */ - case USB_ID(0x03f0, 0x654a): /* HP 320 FHD Webcam */ - if (!strcmp(kctl->id.name, "Mic Capture Volume")) { - usb_audio_info(chip, - "set resolution quirk: cval->res = 16\n"); - cval->res = 16; - } - break; - case USB_ID(0x1bcf, 0x2281): /* HD Webcam */ - if (!strcmp(kctl->id.name, "Mic Capture Volume")) { - usb_audio_info(chip, - "set resolution quirk: cval->res = 16\n"); - cval->res = 16; - } - break; } } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index c5fd180357d1e8..cbfbb064a9c23e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2115,7 +2115,7 @@ struct usb_audio_quirk_flags_table { static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { /* Device matches */ DEVICE_FLG(0x03f0, 0x654a, /* HP 320 FHD Webcam */ - QUIRK_FLAG_GET_SAMPLE_RATE), + QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_MIC_RES_16), DEVICE_FLG(0x041e, 0x3000, /* Creative SB Extigy */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x041e, 0x4080, /* Creative Live Cam VF0610 */ @@ -2123,10 +2123,31 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { DEVICE_FLG(0x045e, 0x083c, /* MS USB Link headset */ QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_CTL_MSG_DELAY | QUIRK_FLAG_DISABLE_AUTOSUSPEND), + DEVICE_FLG(0x046d, 0x0807, /* Logitech Webcam C500 */ + QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_MIC_RES_384), + DEVICE_FLG(0x046d, 0x0808, /* Logitech Webcam C600 */ + QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_MIC_RES_384), + DEVICE_FLG(0x046d, 0x0809, + QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_MIC_RES_384), + DEVICE_FLG(0x046d, 0x0819, /* Logitech Webcam C210 */ + QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_MIC_RES_384), + DEVICE_FLG(0x046d, 0x081b, /* HD Webcam c310 */ + QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_MIC_RES_384), + DEVICE_FLG(0x046d, 0x081d, /* HD Webcam c510 */ + QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_MIC_RES_384), + DEVICE_FLG(0x046d, 0x0825, /* HD Webcam c270 */ + QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_MIC_RES_384), + DEVICE_FLG(0x046d, 0x0826, /* HD Webcam c525 */ + QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_MIC_RES_384), DEVICE_FLG(0x046d, 0x084c, /* Logitech ConferenceCam Connect */ QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_CTL_MSG_DELAY_1M), + DEVICE_FLG(0x046d, 0x08ca, /* Logitech Quickcam Fusion */ + QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_MIC_RES_384), DEVICE_FLG(0x046d, 0x0991, /* Logitech QuickCam Pro */ - QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), + QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR | + QUIRK_FLAG_MIC_RES_384), + DEVICE_FLG(0x046d, 0x09a2, /* QuickCam Communicate Deluxe/S7500 */ + QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_MIC_RES_384), DEVICE_FLG(0x046d, 0x09a4, /* Logitech QuickCam E 3500 */ QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x0499, 0x1509, /* Steinberg UR22 */ @@ -2194,7 +2215,7 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { DEVICE_FLG(0x0fd9, 0x0008, /* Hauppauge HVR-950Q */ QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x1224, 0x2a25, /* Jieli Technology USB PHY 2.0 */ - QUIRK_FLAG_GET_SAMPLE_RATE), + QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_MIC_RES_16), DEVICE_FLG(0x1395, 0x740a, /* Sennheiser DECT */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x1397, 0x0507, /* Behringer UMC202HD */ @@ -2232,9 +2253,9 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { DEVICE_FLG(0x19f7, 0x0035, /* RODE NT-USB+ */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x1bcf, 0x2281, /* HD Webcam */ - QUIRK_FLAG_GET_SAMPLE_RATE), + QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_MIC_RES_16), DEVICE_FLG(0x1bcf, 0x2283, /* NexiGo N930AF FHD Webcam */ - QUIRK_FLAG_GET_SAMPLE_RATE), + QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_MIC_RES_16), DEVICE_FLG(0x2040, 0x7200, /* Hauppauge HVR-950Q */ QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x2040, 0x7201, /* Hauppauge HVR-950Q-MXL */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b0f042c996087e..158ec053dc44dd 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -194,6 +194,8 @@ extern bool snd_usb_skip_validation; * QUIRK_FLAG_FIXED_RATE * Do not set PCM rate (frequency) when only one rate is available * for the given endpoint. + * QUIRK_FLAG_MIC_RES_16 and QUIRK_FLAG_MIC_RES_384 + * Set the fixed resolution for Mic Capture Volume (mostly for webcams) */ #define QUIRK_FLAG_GET_SAMPLE_RATE (1U << 0) @@ -218,5 +220,7 @@ extern bool snd_usb_skip_validation; #define QUIRK_FLAG_IFACE_SKIP_CLOSE (1U << 19) #define QUIRK_FLAG_FORCE_IFACE_RESET (1U << 20) #define QUIRK_FLAG_FIXED_RATE (1U << 21) +#define QUIRK_FLAG_MIC_RES_16 (1U << 22) +#define QUIRK_FLAG_MIC_RES_384 (1U << 23) #endif /* __USBAUDIO_H */ From 224b898f7c5ff575b02911e21383f271761bdeb6 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Mon, 4 Nov 2024 14:43:10 +0530 Subject: [PATCH 35/97] ASoC: amd: acp: Fix for ACP SOF dmic tplg component load failure Stream name mismatch with topology file causes tplg load failure. As SOF framework assigns dailink->stream name, overriding stream name other than link name causes SOF dmic component load failure. [ 35.474995] snd_sof_amd_acp70 0000:c4:00.5: error: can't connect DAI ACPDMIC0.IN stream acp-dmic-codec [ 35.475001] snd_sof_amd_acp70 0000:c4:00.5: failed to add widget type 28 name : ACPDMIC0.IN stream acp-dmic-codec [ 35.475013] sof_mach acp70-dsp: ASoC: failed to load widget ACPDMIC0.IN [ 35.475018] sof_mach acp70-dsp: ASoC: topology: could not load header: -22 [ 35.475072] snd_sof_amd_acp70 0000:c4:00.5: error: tplg component load failed -22 [ 35.475083] snd_sof_amd_acp70 0000:c4:00.5: error: failed to load DSP topology -22 [ 35.475090] snd_sof_amd_acp70 0000:c4:00.5: ASoC: error at snd_soc_component_probe on 0000:c4:00.5: -22 [ 35.475117] sof_mach acp70-dsp: ASoC: failed to instantiate card -22 [ 35.475254] sof_mach acp70-dsp: error -EINVAL: Failed to register card(sof-acp70-dsp) [ 35.475261] sof_mach acp70-dsp: probe with driver sof_mach failed with error -22 Fixes: b2385de2ae11 ("ASoC: amd: acp: Add stream name to ACP PDM DMIC devices") Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20241104091312.1108299-1-venkataprasad.potturu@amd.com Reviewed-by: Mario Limonciello Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-mach-common.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 67aa0ad83486a3..d314253207d5c3 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -1561,7 +1561,6 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) if (drv_data->dmic_cpu_id == DMIC) { links[i].name = "acp-dmic-codec"; - links[i].stream_name = "DMIC capture"; links[i].id = DMIC_BE_ID; links[i].codecs = dmic_codec; links[i].num_codecs = ARRAY_SIZE(dmic_codec); From 82e54d65416b8e7cae422bee1755dd203c95d500 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sun, 27 Oct 2024 23:07:49 -0300 Subject: [PATCH 36/97] ASoC: dt-bindings: fsl_spdif: Document imx6sl/sx compatible fallback i.MX6SL and i.MX6SX SPDIF blocks are compatible with i.MX35. Document 'fsl,imx35-spdif' as a fallback compatible for these two chip variants. This fixes the following dt-schema warnings: compatible: ['fsl,imx6sl-spdif', 'fsl,imx35-spdif'] is too long compatible: ['fsl,imx6sx-spdif', 'fsl,imx35-spdif'] is too long Signed-off-by: Fabio Estevam Link: https://patch.msgid.link/20241028020749.36972-1-festevam@gmail.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,spdif.yaml | 27 ++++++++++++------- 1 file changed, 17 insertions(+), 10 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml index 204f361cea27ab..5654e9f61abaec 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -16,16 +16,23 @@ description: | properties: compatible: - enum: - - fsl,imx35-spdif - - fsl,vf610-spdif - - fsl,imx6sx-spdif - - fsl,imx8qm-spdif - - fsl,imx8qxp-spdif - - fsl,imx8mq-spdif - - fsl,imx8mm-spdif - - fsl,imx8mn-spdif - - fsl,imx8ulp-spdif + oneOf: + - items: + - enum: + - fsl,imx35-spdif + - fsl,imx6sx-spdif + - fsl,imx8mm-spdif + - fsl,imx8mn-spdif + - fsl,imx8mq-spdif + - fsl,imx8qm-spdif + - fsl,imx8qxp-spdif + - fsl,imx8ulp-spdif + - fsl,vf610-spdif + - items: + - enum: + - fsl,imx6sl-spdif + - fsl,imx6sx-spdif + - const: fsl,imx35-spdif reg: maxItems: 1 From 8f5fab5329b7e966344d59fd1c17adbf9f025c52 Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Thu, 31 Oct 2024 14:02:53 +0800 Subject: [PATCH 37/97] ASoC: codecs: ES8326: Reduce pop noise We modify the value of ES8326_ANA_MICBIAS to reduce the pop noise Signed-off-by: Zhang Yi Link: https://patch.msgid.link/20241031060253.21001-1-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index aa3e364827c8a5..a5603b6176889a 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -616,7 +616,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) 0x0F, 0x0F); if (es8326->version > ES8326_VERSION_B) { regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x00); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); } } } else { @@ -1082,7 +1082,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x03); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x13); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); } From 159098859bf6d46b34a1e1f7d44d28987b875878 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 23 Oct 2024 14:41:52 +0200 Subject: [PATCH 38/97] ASoC: qcom: x1e80100: Support boards with two speakers Some Qualcomm X1E laptops have only two speakers. Regardless whether this sound card driver is suitable for them (we could re-use one for some older SoC), we should set reasonable channel map depending on the number of channels, not always 4-speaker setup. This change is necessary for bringing audio support on Lenovo Thinkpad T14s with Qualcomm X1E78100 and only two speakers. Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20241023124152.130706-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/x1e80100.c | 40 ++++++++++++++++++++++++++++++++++----- 1 file changed, 35 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/x1e80100.c b/sound/soc/qcom/x1e80100.c index 898b5c26bf1ee6..8eb57fc12f0dab 100644 --- a/sound/soc/qcom/x1e80100.c +++ b/sound/soc/qcom/x1e80100.c @@ -95,23 +95,53 @@ static int x1e80100_snd_hw_params(struct snd_pcm_substream *substream, return qcom_snd_sdw_hw_params(substream, params, &data->sruntime[cpu_dai->id]); } +static int x1e80100_snd_hw_map_channels(unsigned int *ch_map, int num) +{ + switch (num) { + case 1: + ch_map[0] = PCM_CHANNEL_FC; + break; + case 2: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + break; + case 3: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + ch_map[2] = PCM_CHANNEL_FC; + break; + case 4: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_LB; + ch_map[2] = PCM_CHANNEL_FR; + ch_map[3] = PCM_CHANNEL_RB; + break; + default: + return -EINVAL; + } + + return 0; +} + static int x1e80100_snd_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; - const unsigned int rx_slot[4] = { PCM_CHANNEL_FL, - PCM_CHANNEL_LB, - PCM_CHANNEL_FR, - PCM_CHANNEL_RB }; + unsigned int channels = substream->runtime->channels; + unsigned int rx_slot[4]; int ret; switch (cpu_dai->id) { case WSA_CODEC_DMA_RX_0: case WSA_CODEC_DMA_RX_1: + ret = x1e80100_snd_hw_map_channels(rx_slot, channels); + if (ret) + return ret; + ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, - ARRAY_SIZE(rx_slot), rx_slot); + channels, rx_slot); if (ret) return ret; break; From 1157733344651ca505e259d6554591ff156922fa Mon Sep 17 00:00:00 2001 From: Qiu-ji Chen Date: Mon, 30 Sep 2024 18:12:16 +0800 Subject: [PATCH 39/97] ASoC: codecs: Fix atomicity violation in snd_soc_component_get_drvdata() An atomicity violation occurs when the validity of the variables da7219->clk_src and da7219->mclk_rate is being assessed. Since the entire assessment is not protected by a lock, the da7219 variable might still be in flux during the assessment, rendering this check invalid. To fix this issue, we recommend adding a lock before the block if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq)) so that the legitimacy check for da7219->clk_src and da7219->mclk_rate is protected by the lock, ensuring the validity of the check. This possible bug is found by an experimental static analysis tool developed by our team. This tool analyzes the locking APIs to extract function pairs that can be concurrently executed, and then analyzes the instructions in the paired functions to identify possible concurrency bugs including data races and atomicity violations. Fixes: 6d817c0e9fd7 ("ASoC: codecs: Add da7219 codec driver") Cc: stable@vger.kernel.org Signed-off-by: Qiu-ji Chen Link: https://patch.msgid.link/20240930101216.23723-1-chenqiuji666@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/da7219.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 311ea7918b3124..e2da3e317b5a3e 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1167,17 +1167,20 @@ static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); int ret = 0; - if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq)) + mutex_lock(&da7219->pll_lock); + + if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq)) { + mutex_unlock(&da7219->pll_lock); return 0; + } if ((freq < 2000000) || (freq > 54000000)) { + mutex_unlock(&da7219->pll_lock); dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", freq); return -EINVAL; } - mutex_lock(&da7219->pll_lock); - switch (clk_id) { case DA7219_CLKSRC_MCLK_SQR: snd_soc_component_update_bits(component, DA7219_PLL_CTRL, From 28f7aa0c015036db260adbec37891984a31ed4c2 Mon Sep 17 00:00:00 2001 From: Suraj Sonawane Date: Sat, 2 Nov 2024 18:06:30 +0530 Subject: [PATCH 40/97] ASoC: bcm63xx-pcm-whistler: fix uninit-value in i2s_dma_isr Fix an issue detected by the Smatch tool: sound/soc/bcm/bcm63xx-pcm-whistler.c:264 i2s_dma_isr() error: uninitialized symbol 'val_1'. sound/soc/bcm/bcm63xx-pcm-whistler.c:264 i2s_dma_isr() error: uninitialized symbol 'val_2'. These errors were triggered because the variables 'val_1' and 'val_2' could remain uninitialized if 'offlevel' is zero, meaning the loop that assigns values to them does not execute. In this case, 'dma_addr_next' would use uninitialized data, potentially leading to undefined behavior. To resolve this, a conditional update for 'dma_addr_next' is added, ensuring it is assigned only when 'val_1' and 'val_2' are read. A new boolean variable 'val_read' flags when the values have been retrieved, setting 'dma_addr_next' only if valid data is available. This solution prevents the use of uninitialized data, maintaining defined behavior for 'dma_addr_next' in all cases, and aligns with expected usage of I2S RX descriptor data. Signed-off-by: Suraj Sonawane Link: https://patch.msgid.link/20241102123630.25446-1-surajsonawane0215@gmail.com Signed-off-by: Mark Brown --- sound/soc/bcm/bcm63xx-pcm-whistler.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/bcm/bcm63xx-pcm-whistler.c b/sound/soc/bcm/bcm63xx-pcm-whistler.c index 018f2372e892c2..e3a4fcc63a56dc 100644 --- a/sound/soc/bcm/bcm63xx-pcm-whistler.c +++ b/sound/soc/bcm/bcm63xx-pcm-whistler.c @@ -256,12 +256,16 @@ static irqreturn_t i2s_dma_isr(int irq, void *bcm_i2s_priv) offlevel = (int_status & I2S_RX_DESC_OFF_LEVEL_MASK) >> I2S_RX_DESC_OFF_LEVEL_SHIFT; + bool val_read = false; while (offlevel) { regmap_read(regmap_i2s, I2S_RX_DESC_OFF_ADDR, &val_1); regmap_read(regmap_i2s, I2S_RX_DESC_OFF_LEN, &val_2); + val_read = true; offlevel--; } - prtd->dma_addr_next = val_1 + val_2; + if (val_read) + prtd->dma_addr_next = val_1 + val_2; + ifflevel = (int_status & I2S_RX_DESC_IFF_LEVEL_MASK) >> I2S_RX_DESC_IFF_LEVEL_SHIFT; From 101c9023594ac937b11739aab149a0c14ab901b6 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 25 Oct 2024 14:29:35 +0800 Subject: [PATCH 41/97] ASoC: fsl_mqs: Support accessing registers by scmi interface On i.MX95, the MQS module in Always-on (AON) domain only can be accessed by System Controller Management Interface (SCMI) MISC Protocol. So define a specific regmap_config for the case. Signed-off-by: Shengjiu Wang Link: https://patch.msgid.link/20241025062935.1071408-1-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_mqs.c | 41 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 42 insertions(+) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index e283751abfefe8..8e88830e8e577c 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -30,6 +30,7 @@ config SND_SOC_FSL_MQS tristate "Medium Quality Sound (MQS) module support" depends on SND_SOC_FSL_SAI select REGMAP_MMIO + select IMX_SCMI_MISC_DRV if IMX_SCMI_MISC_EXT !=n help Say Y if you want to add Medium Quality Sound (MQS) support for the Freescale CPUs. diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index 145f9ca15e43cb..0513e9e8402e82 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -6,6 +6,7 @@ // Copyright 2019 NXP #include +#include #include #include #include @@ -74,6 +75,29 @@ struct fsl_mqs { #define FSL_MQS_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) #define FSL_MQS_FORMATS SNDRV_PCM_FMTBIT_S16_LE +static int fsl_mqs_sm_read(void *context, unsigned int reg, unsigned int *val) +{ + struct fsl_mqs *mqs_priv = context; + int num = 1; + + if (IS_ENABLED(CONFIG_IMX_SCMI_MISC_DRV) && + mqs_priv->soc->ctrl_off == reg) + return scmi_imx_misc_ctrl_get(SCMI_IMX_CTRL_MQS1_SETTINGS, &num, val); + + return -EINVAL; +}; + +static int fsl_mqs_sm_write(void *context, unsigned int reg, unsigned int val) +{ + struct fsl_mqs *mqs_priv = context; + + if (IS_ENABLED(CONFIG_IMX_SCMI_MISC_DRV) && + mqs_priv->soc->ctrl_off == reg) + return scmi_imx_misc_ctrl_set(SCMI_IMX_CTRL_MQS1_SETTINGS, val); + + return -EINVAL; +}; + static int fsl_mqs_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -188,6 +212,13 @@ static const struct regmap_config fsl_mqs_regmap_config = { .cache_type = REGCACHE_NONE, }; +static const struct regmap_config fsl_mqs_sm_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_read = fsl_mqs_sm_read, + .reg_write = fsl_mqs_sm_write, +}; + static int fsl_mqs_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; @@ -219,6 +250,16 @@ static int fsl_mqs_probe(struct platform_device *pdev) dev_err(&pdev->dev, "failed to get gpr regmap\n"); return PTR_ERR(mqs_priv->regmap); } + } else if (mqs_priv->soc->type == TYPE_REG_SM) { + mqs_priv->regmap = devm_regmap_init(&pdev->dev, + NULL, + mqs_priv, + &fsl_mqs_sm_regmap); + if (IS_ERR(mqs_priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap: %ld\n", + PTR_ERR(mqs_priv->regmap)); + return PTR_ERR(mqs_priv->regmap); + } } else { regs = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(regs)) From a80aedeb816c81e86e3a59384f010da3414479dd Mon Sep 17 00:00:00 2001 From: Stanislav Jakubek Date: Wed, 30 Oct 2024 18:48:38 +0100 Subject: [PATCH 42/97] ASoC: dt-bindings: sprd,pcm-platform: convert to YAML Convert the Spreadtrum DMA platform bindings to DT schema. Adjust filename to match compatible. Signed-off-by: Stanislav Jakubek Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/9fc646b70a73e7a6c513771d69b0edcd140f09d7.1730310275.git.stano.jakubek@gmail.com Signed-off-by: Mark Brown --- .../bindings/sound/sprd,pcm-platform.yaml | 56 +++++++++++++++++++ .../devicetree/bindings/sound/sprd-pcm.txt | 23 -------- 2 files changed, 56 insertions(+), 23 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sprd,pcm-platform.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sprd-pcm.txt diff --git a/Documentation/devicetree/bindings/sound/sprd,pcm-platform.yaml b/Documentation/devicetree/bindings/sound/sprd,pcm-platform.yaml new file mode 100644 index 00000000000000..c15c01bbb884af --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sprd,pcm-platform.yaml @@ -0,0 +1,56 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sprd,pcm-platform.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Spreadtrum DMA platform + +maintainers: + - Orson Zhai + - Baolin Wang + - Chunyan Zhang + +properties: + compatible: + const: sprd,pcm-platform + + dmas: + maxItems: 10 + + dma-names: + items: + - const: normal_p_l + - const: normal_p_r + - const: normal_c_l + - const: normal_c_r + - const: voice_c + - const: fast_p + - const: loop_c + - const: loop_p + - const: voip_c + - const: voip_p + +required: + - compatible + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + platform { + compatible = "sprd,pcm-platform"; + dmas = <&agcp_dma 1 1>, <&agcp_dma 2 2>, + <&agcp_dma 3 3>, <&agcp_dma 4 4>, + <&agcp_dma 5 5>, <&agcp_dma 6 6>, + <&agcp_dma 7 7>, <&agcp_dma 8 8>, + <&agcp_dma 9 9>, <&agcp_dma 10 10>; + dma-names = "normal_p_l", "normal_p_r", + "normal_c_l", "normal_c_r", + "voice_c", "fast_p", + "loop_c", "loop_p", + "voip_c", "voip_p"; + }; +... diff --git a/Documentation/devicetree/bindings/sound/sprd-pcm.txt b/Documentation/devicetree/bindings/sound/sprd-pcm.txt deleted file mode 100644 index fbbcade2181d80..00000000000000 --- a/Documentation/devicetree/bindings/sound/sprd-pcm.txt +++ /dev/null @@ -1,23 +0,0 @@ -* Spreadtrum DMA platform bindings - -Required properties: -- compatible: Should be "sprd,pcm-platform". -- dmas: Specify the list of DMA controller phandle and DMA request line ordered pairs. -- dma-names: Identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. - -Example: - - audio_platform:platform@0 { - compatible = "sprd,pcm-platform"; - dmas = <&agcp_dma 1 1>, <&agcp_dma 2 2>, - <&agcp_dma 3 3>, <&agcp_dma 4 4>, - <&agcp_dma 5 5>, <&agcp_dma 6 6>, - <&agcp_dma 7 7>, <&agcp_dma 8 8>, - <&agcp_dma 9 9>, <&agcp_dma 10 10>; - dma-names = "normal_p_l", "normal_p_r", - "normal_c_l", "normal_c_r", - "voice_c", "fast_p", - "loop_c", "loop_p", - "voip_c", "voip_p"; - }; From 310558120e5eaf48025c3947fc91b4d02bd90fac Mon Sep 17 00:00:00 2001 From: Stanislav Jakubek Date: Wed, 30 Oct 2024 18:49:22 +0100 Subject: [PATCH 43/97] ASoC: dt-bindings: sprd,sc9860-mcdt: convert to YAML Convert the Spreadtrum Multi-Channel Data Transfer controller bindings to DT schema. Adjust filename to match compatible. Signed-off-by: Stanislav Jakubek Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/140ee384c1c351ffa3abefa8dd3246d1625dda8d.1730310275.git.stano.jakubek@gmail.com Signed-off-by: Mark Brown --- .../bindings/sound/sprd,sc9860-mcdt.yaml | 47 +++++++++++++++++++ .../devicetree/bindings/sound/sprd-mcdt.txt | 19 -------- 2 files changed, 47 insertions(+), 19 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sprd,sc9860-mcdt.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sprd-mcdt.txt diff --git a/Documentation/devicetree/bindings/sound/sprd,sc9860-mcdt.yaml b/Documentation/devicetree/bindings/sound/sprd,sc9860-mcdt.yaml new file mode 100644 index 00000000000000..3b66bedeff9731 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sprd,sc9860-mcdt.yaml @@ -0,0 +1,47 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sprd,sc9860-mcdt.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Spreadtrum Multi-Channel Data Transfer controller + +description: + The Multi-channel data transfer controller is used for sound stream + transmission between the audio subsystem and other AP/CP subsystem. It + supports 10 DAC channels and 10 ADC channels, and each channel can be + configured with DMA mode or interrupt mode. + +maintainers: + - Orson Zhai + - Baolin Wang + - Chunyan Zhang + +properties: + compatible: + const: sprd,sc9860-mcdt + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + +additionalProperties: false + +examples: + - | + #include + #include + + mcdt@41490000 { + compatible = "sprd,sc9860-mcdt"; + reg = <0x41490000 0x170>; + interrupts = ; + }; +... diff --git a/Documentation/devicetree/bindings/sound/sprd-mcdt.txt b/Documentation/devicetree/bindings/sound/sprd-mcdt.txt deleted file mode 100644 index 274ba0acbfd642..00000000000000 --- a/Documentation/devicetree/bindings/sound/sprd-mcdt.txt +++ /dev/null @@ -1,19 +0,0 @@ -Spreadtrum Multi-Channel Data Transfer Binding - -The Multi-channel data transfer controller is used for sound stream -transmission between audio subsystem and other AP/CP subsystem. It -supports 10 DAC channel and 10 ADC channel, and each channel can be -configured with DMA mode or interrupt mode. - -Required properties: -- compatible: Should be "sprd,sc9860-mcdt". -- reg: Should contain registers address and length. -- interrupts: Should contain one interrupt shared by all channel. - -Example: - -mcdt@41490000 { - compatible = "sprd,sc9860-mcdt"; - reg = <0 0x41490000 0 0x170>; - interrupts = ; -}; From 393de01870bcf2ea1eadd21ad12f927d78cbb726 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 1 Nov 2024 17:51:58 +0100 Subject: [PATCH 44/97] ASoC: dt-bindings: qcom,sm8250: Add SM8750 sound card Add bindings for SM8750 sound card, compatible with older SM8450 variant. Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20241101165159.370619-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/qcom,sm8250.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index 2e2e01493a5f4f..b9e33a7429b0c0 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -25,6 +25,7 @@ properties: - enum: - qcom,sm8550-sndcard - qcom,sm8650-sndcard + - qcom,sm8750-sndcard - const: qcom,sm8450-sndcard - enum: - qcom,apq8096-sndcard From 4b9f02b6c5376b65dac398c4f06804c914cbb7be Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 1 Nov 2024 17:51:59 +0100 Subject: [PATCH 45/97] ASoC: qcom: sc8280xp Add SM8750 sound card Add OF device ID entry for SM8750 sound card with its own model name, used to load proper Audioreach topology file. The sound card is compatible with SM8450 and newer family. Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20241101165159.370619-2-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/sc8280xp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c index 922ecada1cd8d9..3113773171761a 100644 --- a/sound/soc/qcom/sc8280xp.c +++ b/sound/soc/qcom/sc8280xp.c @@ -190,6 +190,7 @@ static const struct of_device_id snd_sc8280xp_dt_match[] = { {.compatible = "qcom,sm8450-sndcard", "sm8450"}, {.compatible = "qcom,sm8550-sndcard", "sm8550"}, {.compatible = "qcom,sm8650-sndcard", "sm8650"}, + {.compatible = "qcom,sm8750-sndcard", "sm8750"}, {} }; From adf7ea48ce05a6c5c44f0f9d3f81e83e5cb70c3e Mon Sep 17 00:00:00 2001 From: Frank Li Date: Mon, 28 Oct 2024 15:49:31 -0400 Subject: [PATCH 46/97] ASoC: dt-bindings: fsl-esai: allow fsl,imx8qm-esai fallback to fsl,imx6ull-esai The ESAI of i.MX8QM is the same as i.MX6ULL. So allow fsl,imx8qm-esai fallback to fsl,imx6ull-esai. Signed-off-by: Frank Li Acked-by: Rob Herring (Arm) Link: https://patch.msgid.link/20241028-esai_fix-v1-1-3c1432a5613c@nxp.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,esai.yaml | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.yaml b/Documentation/devicetree/bindings/sound/fsl,esai.yaml index d1b4e23f1c95f5..27c34ce4c2e22a 100644 --- a/Documentation/devicetree/bindings/sound/fsl,esai.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,esai.yaml @@ -18,11 +18,15 @@ description: properties: compatible: - enum: - - fsl,imx35-esai - - fsl,imx6ull-esai - - fsl,imx8qm-esai - - fsl,vf610-esai + oneOf: + - enum: + - fsl,imx35-esai + - fsl,imx6ull-esai + - fsl,vf610-esai + - items: + - enum: + - fsl,imx8qm-esai + - const: fsl,imx6ull-esai reg: maxItems: 1 From 08a3b241adfd90361c16c3e92f5275b816a73f04 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 5 Nov 2024 01:00:00 +0000 Subject: [PATCH 47/97] MAINTAINERS: Generic Sound Card section ALSA SoC Sound has Generic Sound Card (Simple-Card, Audio-Graph-Card, Audio-Graph-Card2). Adds its Maintainer. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87ikt2a41c.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- MAINTAINERS | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 9d6272c00fbd79..388626c3df1f64 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -21697,6 +21697,15 @@ S: Supported W: https://github.com/thesofproject/linux/ F: sound/soc/sof/ +SOUND - GENERIC SOUND CARD (Simple-Audio-Card, Audio-Graph-Card) +M: Kuninori Morimoto +S: Supported +L: linux-sound@vger.kernel.org +F: sound/soc/generic/ +F: include/sound/simple_card* +F: Documentation/devicetree/bindings/sound/simple-card.yaml +F: Documentation/devicetree/bindings/sound/audio-graph*.yaml + SOUNDWIRE SUBSYSTEM M: Vinod Koul M: Bard Liao From 9bb4af400c386374ab1047df44c508512c08c31f Mon Sep 17 00:00:00 2001 From: Amelie Delaunay Date: Tue, 5 Nov 2024 15:02:42 +0100 Subject: [PATCH 48/97] ASoC: stm32: spdifrx: fix dma channel release in stm32_spdifrx_remove In case of error when requesting ctrl_chan DMA channel, ctrl_chan is not null. So the release of the dma channel leads to the following issue: [ 4.879000] st,stm32-spdifrx 500d0000.audio-controller: dma_request_slave_channel error -19 [ 4.888975] Unable to handle kernel NULL pointer dereference at virtual address 000000000000003d [...] [ 5.096577] Call trace: [ 5.099099] dma_release_channel+0x24/0x100 [ 5.103235] stm32_spdifrx_remove+0x24/0x60 [snd_soc_stm32_spdifrx] [ 5.109494] stm32_spdifrx_probe+0x320/0x4c4 [snd_soc_stm32_spdifrx] To avoid this issue, release channel only if the pointer is valid. Fixes: 794df9448edb ("ASoC: stm32: spdifrx: manage rebind issue") Signed-off-by: Amelie Delaunay Signed-off-by: Olivier Moysan Link: https://patch.msgid.link/20241105140242.527279-1-olivier.moysan@foss.st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_spdifrx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index d1b32ba1e1a210..9e30852de93cdf 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -939,7 +939,7 @@ static void stm32_spdifrx_remove(struct platform_device *pdev) { struct stm32_spdifrx_data *spdifrx = platform_get_drvdata(pdev); - if (spdifrx->ctrl_chan) + if (!IS_ERR(spdifrx->ctrl_chan)) dma_release_channel(spdifrx->ctrl_chan); if (spdifrx->dmab) From 93b763a5ab13509e9a2bcbcac3002607736e601b Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Tue, 5 Nov 2024 18:05:57 +0800 Subject: [PATCH 49/97] ASoC: rt722: change the interrupt mask for jack type detection This patch changed the interrupt mask from XU to GE. Signed-off-by: Shuming Fan Link: https://patch.msgid.link/20241105100557.1987917-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt722-sdca-sdw.c | 12 ++++-------- sound/soc/codecs/rt722-sdca.c | 7 ++++--- 2 files changed, 8 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index 87354bb1564e8d..0abbd92cbc7e2a 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -177,7 +177,7 @@ static int rt722_sdca_update_status(struct sdw_slave *slave, * This also could sync with the cache value as the rt722_sdca_jack_init set. */ sdw_write_no_pm(rt722->slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_6); + SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(rt722->slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); } @@ -308,12 +308,8 @@ static int rt722_sdca_interrupt_callback(struct sdw_slave *slave, SDW_SCP_SDCA_INT_SDCA_0, SDW_SCP_SDCA_INT_SDCA_0); if (ret < 0) goto io_error; - } else if (ret & SDW_SCP_SDCA_INTMASK_SDCA_6) { - ret = sdw_update_no_pm(rt722->slave, SDW_SCP_SDCA_INT1, - SDW_SCP_SDCA_INT_SDCA_6, SDW_SCP_SDCA_INT_SDCA_6); - if (ret < 0) - goto io_error; } + ret = sdw_read_no_pm(rt722->slave, SDW_SCP_SDCA_INT2); if (ret < 0) goto io_error; @@ -444,7 +440,7 @@ static int __maybe_unused rt722_sdca_dev_system_suspend(struct device *dev) mutex_lock(&rt722_sdca->disable_irq_lock); rt722_sdca->disable_irq = true; ret1 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6, 0); + SDW_SCP_SDCA_INTMASK_SDCA_0, 0); ret2 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8, 0); mutex_unlock(&rt722_sdca->disable_irq_lock); @@ -471,7 +467,7 @@ static int __maybe_unused rt722_sdca_dev_resume(struct device *dev) if (!slave->unattach_request) { mutex_lock(&rt722->disable_irq_lock); if (rt722->disable_irq == true) { - sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); rt722->disable_irq = false; } diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c index f9f7512ca36087..908846e994df34 100644 --- a/sound/soc/codecs/rt722-sdca.c +++ b/sound/soc/codecs/rt722-sdca.c @@ -190,8 +190,8 @@ static void rt722_sdca_jack_detect_handler(struct work_struct *work) if (!rt722->component->card || !rt722->component->card->instantiated) return; - /* SDW_SCP_SDCA_INT_SDCA_6 is used for jack detection */ - if (rt722->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_6) { + /* SDW_SCP_SDCA_INT_SDCA_0 is used for jack detection */ + if (rt722->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_0) { ret = rt722_sdca_headset_detect(rt722); if (ret < 0) return; @@ -294,7 +294,7 @@ static void rt722_sdca_jack_init(struct rt722_sdca_priv *rt722) if (rt722->hs_jack) { /* set SCP_SDCA_IntMask1[0]=1 */ sdw_write_no_pm(rt722->slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6); + SDW_SCP_SDCA_INTMASK_SDCA_0); /* set SCP_SDCA_IntMask2[0]=1 */ sdw_write_no_pm(rt722->slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); @@ -308,6 +308,7 @@ static void rt722_sdca_jack_init(struct rt722_sdca_priv *rt722) regmap_write(rt722->regmap, SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT722_SDCA_ENT_XU0D, RT722_SDCA_CTL_SELECTED_MODE, 0), 0); + rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_GE_RELATED_CTL1, 0x0000); /* trigger GE interrupt */ rt722_sdca_index_update_bits(rt722, RT722_VENDOR_HDA_CTL, RT722_GE_RELATED_CTL2, 0x4000, 0x4000); From af23d38caae5841bd7aa754a7e7205ab719f568d Mon Sep 17 00:00:00 2001 From: Deep Harsora Date: Tue, 5 Nov 2024 19:10:56 +0800 Subject: [PATCH 50/97] ASoC: Intel: sof_sdw: Add missing quirks from some new Dell Add missing quirks for some new Dell laptops using cs42l43's speaker outputs. Signed-off-by: Deep Harsora Signed-off-by: Bard Liao Reviewed-by: Peter Ujfalusi Reviewed-by: Charles Keepax Link: https://patch.msgid.link/20241105111057.182076-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 48 ++++++++++++++++++++++++++++++++ 1 file changed, 48 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 5614e706a0bbed..b12f700e842fd1 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -480,6 +480,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { .driver_data = (void *)(SOF_SDW_TGL_HDMI | RT711_JD2), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CF6") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, { .callback = sof_sdw_quirk_cb, .matches = { @@ -488,6 +496,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(SOC_SDW_CODEC_SPKR), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CFA") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, /* MeteorLake devices */ { .callback = sof_sdw_quirk_cb, @@ -572,6 +588,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(SOC_SDW_CODEC_SPKR), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0D36") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, { .callback = sof_sdw_quirk_cb, .matches = { @@ -647,6 +671,30 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(SOC_SDW_CODEC_SPKR), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CF3") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CF4") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CF5") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, /* Pantherlake devices*/ { .callback = sof_sdw_quirk_cb, From ed4bcfbcf45d02fa81c77cff86e914d71c1b3c1f Mon Sep 17 00:00:00 2001 From: Fei Shao Date: Tue, 5 Nov 2024 17:11:36 +0800 Subject: [PATCH 51/97] ASoC: dt-bindings: mediatek,mt8188-mt6359: Add mediatek,adsp property On some MediaTek SoCs, an Audio DSP (ADSP) is integrated as a separate hardware block that leverages Sound Open Firmware (SOF) and provides additional audio functionalities. This hardware is optional, and the audio subsystem will still function normally when it's not present. To enable ADSP support, a 'mediatek,adsp' property is required in the sound card node to pass the ADSP phandle. This allows AFE to link to ADSP when the sound card is probed. MT8188 has ADSP integrated, so add the 'mediatek,adsp' property to allow using it in the audio subsystem. This fixes dtbs_check error: Unevaluated properties are not allowed ('mediatek,adsp' was unexpected) Signed-off-by: Fei Shao Reviewed-by: Rob Herring (Arm) Link: https://patch.msgid.link/20241105091246.3944946-1-fshao@chromium.org Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml index f94ad0715e3239..ba482747f0e664 100644 --- a/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml @@ -29,6 +29,13 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of MT8188 ASoC platform. + mediatek,adsp: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the MT8188 ADSP platform, which is the optional Audio DSP + hardware that provides additional audio functionalities if present. + The AFE will link to ADSP when the phandle is provided. + patternProperties: "^dai-link-[0-9]+$": type: object From b3cb7f2a3a1732a775861a2279d951e79c0e614c Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Tue, 5 Nov 2024 08:51:32 +0000 Subject: [PATCH 52/97] ASoC: rt721-sdca: change interrupt mask from XU to GE Change interrupt mask from XU to GE to fix jack detection interrupt issue. Signed-off-by: Jack Yu Link: https://patch.msgid.link/cbc81e324673467a96b70e4e219766b5@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt721-sdca-sdw.c | 13 ++++--------- sound/soc/codecs/rt721-sdca.c | 6 ++++-- sound/soc/codecs/rt721-sdca.h | 1 + 3 files changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt721-sdca-sdw.c b/sound/soc/codecs/rt721-sdca-sdw.c index c0f8cccae3b2bb..c71453da088a01 100644 --- a/sound/soc/codecs/rt721-sdca-sdw.c +++ b/sound/soc/codecs/rt721-sdca-sdw.c @@ -203,7 +203,7 @@ static int rt721_sdca_update_status(struct sdw_slave *slave, * This also could sync with the cache value as the rt721_sdca_jack_init set. */ sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_6); + SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); } @@ -280,7 +280,7 @@ static int rt721_sdca_read_prop(struct sdw_slave *slave) } /* set the timeout values */ - prop->clk_stop_timeout = 900; + prop->clk_stop_timeout = 1380; /* wake-up event */ prop->wake_capable = 1; @@ -337,11 +337,6 @@ static int rt721_sdca_interrupt_callback(struct sdw_slave *slave, SDW_SCP_SDCA_INT_SDCA_0, SDW_SCP_SDCA_INT_SDCA_0); if (ret < 0) goto io_error; - } else if (ret & SDW_SCP_SDCA_INTMASK_SDCA_6) { - ret = sdw_update_no_pm(rt721->slave, SDW_SCP_SDCA_INT1, - SDW_SCP_SDCA_INT_SDCA_6, SDW_SCP_SDCA_INT_SDCA_6); - if (ret < 0) - goto io_error; } ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT2); if (ret < 0) @@ -475,7 +470,7 @@ static int __maybe_unused rt721_sdca_dev_system_suspend(struct device *dev) mutex_lock(&rt721_sdca->disable_irq_lock); rt721_sdca->disable_irq = true; ret1 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6, 0); + SDW_SCP_SDCA_INTMASK_SDCA_0, 0); ret2 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8, 0); mutex_unlock(&rt721_sdca->disable_irq_lock); @@ -502,7 +497,7 @@ static int __maybe_unused rt721_sdca_dev_resume(struct device *dev) if (!slave->unattach_request) { mutex_lock(&rt721->disable_irq_lock); if (rt721->disable_irq == true) { - sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); rt721->disable_irq = false; } diff --git a/sound/soc/codecs/rt721-sdca.c b/sound/soc/codecs/rt721-sdca.c index bdd160b80b6465..1c9f32e405cf95 100644 --- a/sound/soc/codecs/rt721-sdca.c +++ b/sound/soc/codecs/rt721-sdca.c @@ -39,7 +39,7 @@ static void rt721_sdca_jack_detect_handler(struct work_struct *work) return; /* SDW_SCP_SDCA_INT_SDCA_6 is used for jack detection */ - if (rt721->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_6) { + if (rt721->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_0) { rt721->jack_type = rt_sdca_headset_detect(rt721->regmap, RT721_SDCA_ENT_GE49); if (rt721->jack_type < 0) @@ -286,7 +286,7 @@ static void rt721_sdca_jack_init(struct rt721_sdca_priv *rt721) mutex_lock(&rt721->calibrate_mutex); if (rt721->hs_jack) { sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6); + SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); dev_dbg(&rt721->slave->dev, "in %s enable\n", __func__); @@ -298,6 +298,8 @@ static void rt721_sdca_jack_init(struct rt721_sdca_priv *rt721) regmap_write(rt721->regmap, SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_XU0D, RT721_SDCA_CTL_SELECTED_MODE, 0), 0); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_XU_REL_CTRL, 0x0000); rt_sdca_index_update_bits(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, RT721_GE_REL_CTRL1, 0x4000, 0x4000); } diff --git a/sound/soc/codecs/rt721-sdca.h b/sound/soc/codecs/rt721-sdca.h index e2f071909da863..0a82c107b19a20 100644 --- a/sound/soc/codecs/rt721-sdca.h +++ b/sound/soc/codecs/rt721-sdca.h @@ -133,6 +133,7 @@ struct rt721_sdca_dmic_kctrl_priv { #define RT721_HDA_LEGACY_UAJ_CTL 0x02 #define RT721_HDA_LEGACY_CTL1 0x05 #define RT721_HDA_LEGACY_RESET_CTL 0x06 +#define RT721_XU_REL_CTRL 0x0c #define RT721_GE_REL_CTRL1 0x0d #define RT721_HDA_LEGACY_GPIO_WAKE_EN_CTL 0x0e #define RT721_GE_SDCA_RST_CTRL 0x10 From 99348781d249817c8f96a7cbf636b7c6d74bd756 Mon Sep 17 00:00:00 2001 From: Fei Shao Date: Tue, 5 Nov 2024 17:18:11 +0800 Subject: [PATCH 53/97] ASoC: dt-bindings: everest,es8326: Document interrupt property The ES8326 audio codec has one interrupt pin for headset detection according to the datasheet. Document that in the binding. This fixes dtbs_check error: 'interrupts-extended' does not match any of the regexes: 'pinctrl-[0-9]+' Signed-off-by: Fei Shao Reviewed-by: Rob Herring (Arm) Link: https://patch.msgid.link/20241105091910.3984381-1-fshao@chromium.org Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/everest,es8326.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/everest,es8326.yaml b/Documentation/devicetree/bindings/sound/everest,es8326.yaml index d51431df7acf9c..b5594a9d508e87 100644 --- a/Documentation/devicetree/bindings/sound/everest,es8326.yaml +++ b/Documentation/devicetree/bindings/sound/everest,es8326.yaml @@ -24,6 +24,10 @@ properties: items: - const: mclk + interrupts: + maxItems: 1 + description: interrupt output for headset detection + "#sound-dai-cells": const: 0 From 9e096b3cbbecfeecb544ec5a94dbd6e10df2219b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 6 Nov 2024 09:53:12 +0200 Subject: [PATCH 54/97] ALSA: compress_offload: Use runtime pointer in snd_compr_poll() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit runtime is not used as seen with W=1 : sound/core/compress_offload.c: In function ‘snd_compr_poll’: sound/core/compress_offload.c:409:35: error: variable ‘runtime’ set but not used [-Werror=unused-but-set-variable] 409 | struct snd_compr_runtime *runtime; | ^~~~~~~ Instead of dropping the runtime, use it in the function in place of stream->runtime Fixes: 04177158cf98 ("ALSA: compress_offload: introduce accel operation mode") Signed-off-by: Peter Ujfalusi Reviewed-by: Jaroslav Kysela Link: https://patch.msgid.link/20241106075312.15601-1-peter.ujfalusi@linux.intel.com Signed-off-by: Takashi Iwai --- sound/core/compress_offload.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 5ecdad80a0d8cd..86ed2fbee0c867 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -418,7 +418,7 @@ static __poll_t snd_compr_poll(struct file *f, poll_table *wait) guard(mutex)(&stream->device->lock); - switch (stream->runtime->state) { + switch (runtime->state) { case SNDRV_PCM_STATE_OPEN: case SNDRV_PCM_STATE_XRUN: return snd_compr_get_poll(stream) | EPOLLERR; @@ -426,7 +426,7 @@ static __poll_t snd_compr_poll(struct file *f, poll_table *wait) break; } - poll_wait(f, &stream->runtime->sleep, wait); + poll_wait(f, &runtime->sleep, wait); #if IS_ENABLED(CONFIG_SND_COMPRESS_ACCEL) if (stream->direction == SND_COMPRESS_ACCEL) { @@ -445,18 +445,18 @@ static __poll_t snd_compr_poll(struct file *f, poll_table *wait) avail = snd_compr_get_avail(stream); pr_debug("avail is %ld\n", (unsigned long)avail); /* check if we have at least one fragment to fill */ - switch (stream->runtime->state) { + switch (runtime->state) { case SNDRV_PCM_STATE_DRAINING: /* stream has been woken up after drain is complete * draining done so set stream state to stopped */ retval = snd_compr_get_poll(stream); - stream->runtime->state = SNDRV_PCM_STATE_SETUP; + runtime->state = SNDRV_PCM_STATE_SETUP; break; case SNDRV_PCM_STATE_RUNNING: case SNDRV_PCM_STATE_PREPARED: case SNDRV_PCM_STATE_PAUSED: - if (avail >= stream->runtime->fragment_size) + if (avail >= runtime->fragment_size) retval = snd_compr_get_poll(stream); break; default: From b6bd3f3b6357f727a177f31861144788eceda3c1 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 6 Nov 2024 13:58:09 +0800 Subject: [PATCH 55/97] ASoC: Intel: Kconfig: make SND_SOC_ACPI_INTEL_MATCH depend on ACPI MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit SND_SOC_ACPI_INTEL_MATCH relies on ACPI functions. It will not work if ACPI is not selected. Signed-off-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Liam Girdwood Reviewed-by: Ranjani Sridharan Link: https://patch.msgid.link/20241106055810.10123-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 14461dee3e52ce..f644e94234280c 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -70,7 +70,8 @@ if SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL config SND_SOC_ACPI_INTEL_MATCH tristate - select SND_SOC_ACPI if ACPI + depends on ACPI + select SND_SOC_ACPI select SND_SOC_ACPI_INTEL_SDCA_QUIRKS # this option controls the compilation of ACPI matching tables and # helpers and is not meant to be selected by the user. From 845cb1ddf1fc2212f876db6df9d3277badd35709 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 6 Nov 2024 13:58:10 +0800 Subject: [PATCH 56/97] ASoC: Intel: Kconfig: select SND_SOC_SDCA by SND_SOC_ACPI_INTEL_SDCA_QUIRKS MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit SND_SOC_SDCA can't be optional when SND_SOC_ACPI_INTEL_SDCA_QUIRKS is selected. IS_REACHABLE can prevent the link error. However it is not suitable for this case. When CONFIG_SND_SOC_ACPI_INTEL_SDCA_QUIRKS is Y and CONFIG_SND_SOC_SDCA is M, the SDCA helpers will be empty and return false. But we need the helpers to do their jobs whenSND_SOC_SDCA is M. IOW, the SDCA library is not optional for Intel platforms where the SDCA_QUIRK is selected. Also, make SND_SOC_SDCA invisible. SND_SOC_SDCA should be selected if a device supports SDCA. User should not unselect it. Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202411021722.DiX1Y5sf-lkp@intel.com/ Suggested-by: Peter Ujfalusi Signed-off-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Liam Girdwood Reviewed-by: Ranjani Sridharan Link: https://patch.msgid.link/20241106055810.10123-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 2 +- sound/soc/sdca/Kconfig | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f644e94234280c..46b45f390ae9ea 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -78,7 +78,7 @@ config SND_SOC_ACPI_INTEL_MATCH config SND_SOC_ACPI_INTEL_SDCA_QUIRKS tristate - imply SND_SOC_SDCA + select SND_SOC_SDCA endif ## SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL diff --git a/sound/soc/sdca/Kconfig b/sound/soc/sdca/Kconfig index 07f6822fa614f3..ee20b9914aa1fa 100644 --- a/sound/soc/sdca/Kconfig +++ b/sound/soc/sdca/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_SOC_SDCA - tristate "ASoC SDCA library" + tristate depends on ACPI help This option enables support for the MIPI SoundWire Device From 82a1ccdf616d396c99f535febb6c997781c5c26c Mon Sep 17 00:00:00 2001 From: Kiseok Jo Date: Wed, 6 Nov 2024 09:57:59 +0900 Subject: [PATCH 57/97] ASoC: dt-bindings: irondevice,sma1307: Add initial DT This adds the schema binding for the Iron Device SMA1307 Amp Signed-off-by: Kiseok Jo Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20241106005800.7520-2-kiseok.jo@irondevice.com Signed-off-by: Mark Brown --- .../bindings/sound/irondevice,sma1307.yaml | 53 +++++++++++++++++++ 1 file changed, 53 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml diff --git a/Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml b/Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml new file mode 100644 index 00000000000000..1e2a038d004890 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml @@ -0,0 +1,53 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/irondevice,sma1307.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Iron Device SMA1307 Audio Amplifier + +maintainers: + - Kiseok Jo + +description: + SMA1307 boosted digital speaker amplifier with feedback-loop. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - irondevice,sma1307a + - irondevice,sma1307aq + description: + If a 'q' is added, it indicated the product is AEC-Q100 + qualified for automotive applications. SMA1307A supports + both WLCSP and QFN packages. However, SMA1307AQ only + supports the QFN package. + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@1e { + compatible = "irondevice,sma1307a"; + reg = <0x1e>; + #sound-dai-cells = <1>; + }; + }; From 576c57e6b4c1d734bcb7cc33dde9a99a9383b520 Mon Sep 17 00:00:00 2001 From: Kiseok Jo Date: Wed, 6 Nov 2024 09:58:00 +0900 Subject: [PATCH 58/97] ASoC: sma1307: Add driver for Iron Device SMA1307 The Iron Device SMA1307 is a boosted digital speaker amplifier Signed-off-by: Kiseok Jo Link: https://patch.msgid.link/20241106005800.7520-3-kiseok.jo@irondevice.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2052 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sma1307.h | 444 ++++++++ 4 files changed, 2508 insertions(+) create mode 100644 sound/soc/codecs/sma1307.c create mode 100644 sound/soc/codecs/sma1307.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7dead36be02c4b..0f2df7c91e186a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -240,6 +240,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_SIMPLE_AMPLIFIER imply SND_SOC_SIMPLE_MUX imply SND_SOC_SMA1303 + imply SND_SOC_SMA1307 imply SND_SOC_SPDIF imply SND_SOC_SRC4XXX_I2C imply SND_SOC_SSM2305 @@ -1873,6 +1874,15 @@ config SND_SOC_SMA1303 help Enable support for Iron Device SMA1303 Boosted Class-D amplifier +config SND_SOC_SMA1307 + tristate "Iron Device SMA1307 Audio Amplifier" + depends on I2C + help + Enable support for Iron Device SMA1307 boosted digital speaker + amplifier with feedback-loop. + If you are using a system with an SMA1307 amplifier connected + via I2C, enable this option. + config SND_SOC_SPDIF tristate "S/PDIF CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ea93968f6bf627..f37e82ddb7a104 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -279,6 +279,7 @@ snd-soc-sigmadsp-i2c-y := sigmadsp-i2c.o snd-soc-sigmadsp-regmap-y := sigmadsp-regmap.o snd-soc-si476x-y := si476x.o snd-soc-sma1303-y := sma1303.o +snd-soc-sma1307-y := sma1307.o snd-soc-spdif-tx-y := spdif_transmitter.o snd-soc-spdif-rx-y := spdif_receiver.o snd-soc-src4xxx-y := src4xxx.o @@ -689,6 +690,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SMA1303) += snd-soc-sma1303.o +obj-$(CONFIG_SND_SOC_SMA1307) += snd-soc-sma1307.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SRC4XXX) += snd-soc-src4xxx.o obj-$(CONFIG_SND_SOC_SRC4XXX_I2C) += snd-soc-src4xxx-i2c.o diff --git a/sound/soc/codecs/sma1307.c b/sound/soc/codecs/sma1307.c new file mode 100644 index 00000000000000..985a247b331072 --- /dev/null +++ b/sound/soc/codecs/sma1307.c @@ -0,0 +1,2052 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// sma1307.c -- sma1307 ALSA SoC Audio driver +// +// Copyright 2024 Iron Device Corporation +// +// Auther: Gyuhwa Park +// Auther: Kiseok Jo + +#include +#include +#include +#include +#include +#include +#include "sma1307.h" + +#define CHECK_PERIOD_TIME 1 /* sec per HZ */ +#define PLL_MATCH(_input_clk_name, _output_clk_name, _input_clk,\ + _post_n, _n, _vco, _p_cp)\ +{\ + .input_clk_name = _input_clk_name,\ + .output_clk_name = _output_clk_name,\ + .input_clk = _input_clk,\ + .post_n = _post_n,\ + .n = _n,\ + .vco = _vco,\ + .p_cp = _p_cp,\ +} + +static const char *setting_file = "sma1307_setting.bin"; +#define SMA1307_SETTING_CHECKSUM 0x100000 + +/* PLL clock setting Table */ +struct sma1307_pll_match { + char *input_clk_name; + char *output_clk_name; + unsigned int input_clk; + unsigned int post_n; + unsigned int n; + unsigned int vco; + unsigned int p_cp; +}; + +struct sma1307_data { + char *name; + void (*init)(struct regmap *regmap); +}; + +struct sma1307_priv { + bool check_fault_status; + bool force_mute_status; + bool sw_ot1_prot; + char *name; + enum sma1307_mode amp_mode; + int binary_mode; + int dapm_aif_in; + int dapm_aif_out0; + int dapm_aif_out1; + int dapm_sdo_en; + int dapm_sdo_setting; + int num_of_pll_matches; + int check_fault_period; + struct delayed_work check_fault_work; + struct device *dev; + struct kobject *kobj; + struct mutex default_lock; + struct regmap *regmap; + struct sma1307_setting_file set; + const struct sma1307_pll_match *pll_matches; + const struct sma1307_data *data; + unsigned int cur_vol; + unsigned int format; + unsigned int frame_size; + unsigned int init_vol; + unsigned int last_bclk; + unsigned int otp_trm2; + unsigned int otp_trm3; + unsigned int rev_num; + unsigned int sys_clk_id; + unsigned int tdm_slot0_rx; + unsigned int tdm_slot1_rx; + unsigned int tdm_slot0_tx; + unsigned int tdm_slot1_tx; + unsigned int tsdw_cnt; +}; + +static const struct sma1307_pll_match sma1307_pll_matches[] = { + /* in_clk_name, out_clk_name, input_clk post_n, n, vco, p_cp */ + PLL_MATCH("1.411MHz", "24.554MHz", + 1411200, 0x06, 0xD1, 0x88, 0x00), + PLL_MATCH("1.536MHz", "24.576MHz", + 1536000, 0x06, 0xC0, 0x88, 0x00), + PLL_MATCH("2.822MHz", "24.554MHz", + 2822400, 0x06, 0xD1, 0x88, 0x04), + PLL_MATCH("3.072MHz", "24.576MHz", + 3072000, 0x06, 0x60, 0x88, 0x00), + PLL_MATCH("6.144MHz", "24.576MHz", + 6144000, 0x06, 0x60, 0x88, 0x04), + PLL_MATCH("12.288MHz", "24.576MHz", + 12288000, 0x06, 0x60, 0x88, 0x08), + PLL_MATCH("19.2MHz", "24.48MHz", + 19200000, 0x06, 0x7B, 0x88, 0x0C), + PLL_MATCH("24.576MHz", "24.576MHz", + 24576000, 0x06, 0x60, 0x88, 0x0C), +}; + +static struct snd_soc_component *sma1307_amp_component; + +static void sma1307_startup(struct snd_soc_component *); +static void sma1307_shutdown(struct snd_soc_component *); +static void sma1307_reset(struct snd_soc_component *); +static void sma1307_set_binary(struct snd_soc_component *); +static void sma1307_set_default(struct snd_soc_component *); + +/* Initial register value - 6.0W SPK (8ohm load) */ +static const struct reg_default sma1307_reg_def[] = { + { 0x00, 0x80 }, + { 0x01, 0x00 }, + { 0x02, 0x52 }, + { 0x03, 0x4C }, + { 0x04, 0x47 }, + { 0x05, 0x42 }, + { 0x06, 0x40 }, + { 0x07, 0x40 }, + { 0x08, 0x3C }, + { 0x09, 0x2F }, + { 0x0A, 0x32 }, + { 0x0B, 0x50 }, + { 0x0C, 0x8C }, + { 0x0D, 0x00 }, + { 0x0E, 0x3F }, + { 0x0F, 0x00 }, + { 0x10, 0x00 }, + { 0x11, 0x00 }, + { 0x12, 0x00 }, + { 0x13, 0x09 }, + { 0x14, 0x12 }, + { 0x1C, 0x00 }, + { 0x1D, 0x85 }, + { 0x1E, 0xA1 }, + { 0x1F, 0x67 }, + { 0x22, 0x00 }, + { 0x23, 0x1F }, + { 0x24, 0x7A }, + { 0x25, 0x00 }, + { 0x26, 0xFF }, + { 0x27, 0x39 }, + { 0x28, 0x54 }, + { 0x29, 0x92 }, + { 0x2A, 0xB0 }, + { 0x2B, 0xED }, + { 0x2C, 0xED }, + { 0x2D, 0xFF }, + { 0x2E, 0xFF }, + { 0x2F, 0xFF }, + { 0x30, 0xFF }, + { 0x31, 0xFF }, + { 0x32, 0xFF }, + { 0x34, 0x01 }, + { 0x35, 0x17 }, + { 0x36, 0x92 }, + { 0x37, 0x00 }, + { 0x38, 0x01 }, + { 0x39, 0x10 }, + { 0x3E, 0x01 }, + { 0x3F, 0x08 }, + { 0x8B, 0x05 }, + { 0x8C, 0x50 }, + { 0x8D, 0x80 }, + { 0x8E, 0x10 }, + { 0x8F, 0x02 }, + { 0x90, 0x02 }, + { 0x91, 0x83 }, + { 0x92, 0xC0 }, + { 0x93, 0x00 }, + { 0x94, 0xA4 }, + { 0x95, 0x74 }, + { 0x96, 0x57 }, + { 0xA2, 0xCC }, + { 0xA3, 0x28 }, + { 0xA4, 0x40 }, + { 0xA5, 0x01 }, + { 0xA6, 0x41 }, + { 0xA7, 0x08 }, + { 0xA8, 0x04 }, + { 0xA9, 0x27 }, + { 0xAA, 0x10 }, + { 0xAB, 0x10 }, + { 0xAC, 0x10 }, + { 0xAD, 0x0F }, + { 0xAE, 0xCD }, + { 0xAF, 0x70 }, + { 0xB0, 0x03 }, + { 0xB1, 0xEF }, + { 0xB2, 0x03 }, + { 0xB3, 0xEF }, + { 0xB4, 0xF3 }, + { 0xB5, 0x3D }, +}; + +static bool sma1307_readable_register(struct device *dev, unsigned int reg) +{ + if (reg > SMA1307_FF_DEVICE_INDEX) + return false; + + switch (reg) { + case SMA1307_00_SYSTEM_CTRL ... SMA1307_1F_TONE_FINE_VOLUME: + case SMA1307_22_COMP_HYS_SEL ... SMA1307_32_BROWN_OUT_PROT19: + case SMA1307_34_OCP_SPK ... SMA1307_39_PMT_NZ_VAL: + case SMA1307_3B_TEST1 ... SMA1307_3F_ATEST2: + case SMA1307_8B_PLL_POST_N ... SMA1307_9A_OTP_TRM3: + case SMA1307_A0_PAD_CTRL0 ... SMA1307_BE_MCBS_CTRL2: + case SMA1307_F5_READY_FOR_V_SAR: + case SMA1307_F7_READY_FOR_T_SAR ... SMA1307_FF_DEVICE_INDEX: + break; + default: + return false; + } + return true; +} + +static bool sma1307_writeable_register(struct device *dev, unsigned int reg) +{ + if (reg > SMA1307_FF_DEVICE_INDEX) + return false; + + switch (reg) { + case SMA1307_00_SYSTEM_CTRL ... SMA1307_1F_TONE_FINE_VOLUME: + case SMA1307_22_COMP_HYS_SEL ... SMA1307_32_BROWN_OUT_PROT19: + case SMA1307_34_OCP_SPK ... SMA1307_39_PMT_NZ_VAL: + case SMA1307_3B_TEST1 ... SMA1307_3F_ATEST2: + case SMA1307_8B_PLL_POST_N ... SMA1307_9A_OTP_TRM3: + case SMA1307_A0_PAD_CTRL0 ... SMA1307_BE_MCBS_CTRL2: + break; + default: + return false; + } + return true; +} + +static bool sma1307_volatile_register(struct device *dev, unsigned int reg) +{ + if (reg > SMA1307_FF_DEVICE_INDEX) + return false; + + switch (reg) { + case SMA1307_F8_STATUS_T1 ... SMA1307_FF_DEVICE_INDEX: + break; + default: + return false; + } + return true; +} + +/* DB scale conversion of speaker volume */ +static const DECLARE_TLV_DB_SCALE(sma1307_spk_tlv, -6000, 50, 0); + +static const char *const sma1307_aif_in_source_text[] = { + "Mono", "Left", "Right" +}; + +static const char *const sma1307_sdo_setting_text[] = { + "Data_One_48k", "Data_Two_48k", "Data_Two_24k", + "Clk_PLL", "Clk_OSC" +}; + +static const char *const sma1307_aif_out_source_text[] = { + "Disable", "After_FmtC", "After_Mixer", "After_DSP", + "Vrms2_Avg", "Battery", "Temperature", "After_Delay" +}; + +static const char *const sma1307_tdm_slot_text[] = { + "Slot0", "Slot1", "Slot2", "Slot3", + "Slot4", "Slot5", "Slot6", "Slot7" +}; + +static const char *const sma1307_binary_mode_text[] = { + "Mode0", "Mode1", "Mode2", "Mode3", "Mode4" +}; + +static const char *const sma1307_reset_text[] = { + "Reset" +}; + +static const struct soc_enum sma1307_aif_in_source_enum = +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1307_aif_in_source_text), + sma1307_aif_in_source_text); +static const struct soc_enum sma1307_sdo_setting_enum = +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1307_sdo_setting_text), + sma1307_sdo_setting_text); +static const struct soc_enum sma1307_aif_out_source_enum = +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1307_aif_out_source_text), + sma1307_aif_out_source_text); +static const struct soc_enum sma1307_tdm_slot_enum = +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1307_tdm_slot_text), + sma1307_tdm_slot_text); +static const struct soc_enum sma1307_binary_mode_enum = +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1307_binary_mode_text), + sma1307_binary_mode_text); +static const struct soc_enum sma1307_reset_enum = +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1307_reset_text), + sma1307_reset_text); + +static int sma1307_force_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = (int)sma1307->force_mute_status; + + return 0; +} + +static int sma1307_force_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + bool change = false, val = (bool)ucontrol->value.integer.value[0]; + + if (sma1307->force_mute_status == val) { + change = false; + } else { + change = true; + sma1307->force_mute_status = val; + } + + return change; +} + +static int sma1307_tdm_slot_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + int val1, val2; + + regmap_read(sma1307->regmap, SMA1307_A5_TDM1, &val1); + regmap_read(sma1307->regmap, SMA1307_A6_TDM2, &val2); + + if (!strcmp(kcontrol->id.name, SMA1307_TDM_RX0_POS_NAME)) { + ucontrol->value.integer.value[0] + = (val1 & SMA1307_TDM_SLOT0_RX_POS_MASK) >> 3; + sma1307->tdm_slot0_rx = ucontrol->value.integer.value[0]; + } else if (!strcmp(kcontrol->id.name, SMA1307_TDM_RX1_POS_NAME)) { + ucontrol->value.integer.value[0] + = val1 & SMA1307_TDM_SLOT1_RX_POS_MASK; + sma1307->tdm_slot1_rx = ucontrol->value.integer.value[0]; + } else if (!strcmp(kcontrol->id.name, SMA1307_TDM_TX0_POS_NAME)) { + ucontrol->value.integer.value[0] + = (val2 & SMA1307_TDM_SLOT0_TX_POS_MASK) >> 3; + sma1307->tdm_slot0_tx = ucontrol->value.integer.value[0]; + } else if (!strcmp(kcontrol->id.name, SMA1307_TDM_TX1_POS_NAME)) { + ucontrol->value.integer.value[0] + = val2 & SMA1307_TDM_SLOT1_TX_POS_MASK; + sma1307->tdm_slot1_tx = ucontrol->value.integer.value[0]; + } else { + return -EINVAL; + } + + return 0; +} + +static int sma1307_tdm_slot_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + int val = (int)ucontrol->value.integer.value[0]; + bool change; + + if (!strcmp(kcontrol->id.name, SMA1307_TDM_RX0_POS_NAME)) { + if (sma1307->tdm_slot0_rx == val) + change = false; + else { + change = true; + sma1307->tdm_slot0_rx = val; + regmap_update_bits(sma1307->regmap, SMA1307_A5_TDM1, + SMA1307_TDM_SLOT0_RX_POS_MASK, val << 3); + } + } else if (!strcmp(kcontrol->id.name, SMA1307_TDM_RX1_POS_NAME)) { + if (sma1307->tdm_slot1_rx == val) + change = false; + else { + change = true; + sma1307->tdm_slot1_rx = val; + regmap_update_bits(sma1307->regmap, SMA1307_A5_TDM1, + SMA1307_TDM_SLOT1_RX_POS_MASK, val); + } + } else if (!strcmp(kcontrol->id.name, SMA1307_TDM_TX0_POS_NAME)) { + if (sma1307->tdm_slot0_tx == val) + change = false; + else { + change = true; + sma1307->tdm_slot0_tx = val; + regmap_update_bits(sma1307->regmap, SMA1307_A6_TDM2, + SMA1307_TDM_SLOT0_TX_POS_MASK, val << 3); + } + } else if (!strcmp(kcontrol->id.name, SMA1307_TDM_TX1_POS_NAME)) { + if (sma1307->tdm_slot1_tx == val) + change = false; + else { + change = true; + sma1307->tdm_slot1_tx = val; + regmap_update_bits(sma1307->regmap, SMA1307_A6_TDM2, + SMA1307_TDM_SLOT1_TX_POS_MASK, val); + } + } else { + dev_err(sma1307->dev, "%s: Invalid Control ID - %s\n", + __func__, kcontrol->id.name); + return -EINVAL; + } + + return change; +} + +static int sma1307_sw_ot1_prot_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = (int)sma1307->sw_ot1_prot; + + return 0; +} + +static int sma1307_sw_ot1_prot_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + bool change = false, val = (bool)ucontrol->value.integer.value[0]; + + if (sma1307->sw_ot1_prot == val) + change = false; + else { + change = true; + sma1307->sw_ot1_prot = val; + } + + return change; +} + +static int sma1307_check_fault_status_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = (int)sma1307->check_fault_status; + + return 0; +} + +static int sma1307_check_fault_status_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + bool change = false, val = (bool)ucontrol->value.integer.value[0]; + + if (sma1307->check_fault_status == val) { + change = false; + } else { + change = true; + sma1307->check_fault_status = val; + } + + return change; +} + +static int sma1307_check_fault_period_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = sma1307->check_fault_period; + + return 0; +} + +static int sma1307_check_fault_period_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + bool change = false; + int val = ucontrol->value.integer.value[0]; + + if (val < mc->min || val > mc->max) + return -EINVAL; + if (sma1307->check_fault_period == val) { + change = false; + } else { + change = true; + sma1307->check_fault_period = val; + } + + return change; +} + +static int sma1307_reset_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + regmap_update_bits(sma1307->regmap, SMA1307_00_SYSTEM_CTRL, + SMA1307_RESET_MASK, SMA1307_RESET_ON); + sma1307_reset(component); + + snd_ctl_notify(component->card->snd_card, SNDRV_CTL_EVENT_MASK_VALUE, + &kcontrol->id); + + return true; +} + +static int sma1307_binary_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_kcontrol_chip(kcontrol); + + sma1307->binary_mode = (int)ucontrol->value.enumerated.item[0]; + if (sma1307->set.status) + sma1307_set_binary(component); + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + +static void sma1307_startup(struct snd_soc_component *component) +{ + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + regmap_update_bits(sma1307->regmap, SMA1307_A2_TOP_MAN1, + SMA1307_PLL_MASK, SMA1307_PLL_ON); + regmap_update_bits(sma1307->regmap, SMA1307_00_SYSTEM_CTRL, + SMA1307_POWER_MASK, SMA1307_POWER_ON); + + if (sma1307->amp_mode == SMA1307_MONO_MODE) { + regmap_update_bits(sma1307->regmap, + SMA1307_10_SYSTEM_CTRL1, + SMA1307_SPK_MODE_MASK, + SMA1307_SPK_MONO); + } else { + regmap_update_bits(sma1307->regmap, + SMA1307_10_SYSTEM_CTRL1, + SMA1307_SPK_MODE_MASK, + SMA1307_SPK_STEREO); + } + + if (sma1307->check_fault_status) { + if (sma1307->check_fault_period > 0) + queue_delayed_work(system_freezable_wq, + &sma1307->check_fault_work, + sma1307->check_fault_period * HZ); + else + queue_delayed_work(system_freezable_wq, + &sma1307->check_fault_work, + CHECK_PERIOD_TIME * HZ); + } +} + +static void sma1307_shutdown(struct snd_soc_component *component) +{ + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + /* for SMA1307A */ + cancel_delayed_work_sync(&sma1307->check_fault_work); + + regmap_update_bits(sma1307->regmap, SMA1307_0E_MUTE_VOL_CTRL, + SMA1307_SPK_MUTE_MASK, SMA1307_SPK_MUTE); + /* Need to wait time for mute slope */ + msleep(55); + + regmap_update_bits(sma1307->regmap, SMA1307_10_SYSTEM_CTRL1, + SMA1307_SPK_MODE_MASK, SMA1307_SPK_OFF); + regmap_update_bits(sma1307->regmap, SMA1307_A2_TOP_MAN1, + SMA1307_PLL_MASK, SMA1307_PLL_OFF); + regmap_update_bits(sma1307->regmap, SMA1307_00_SYSTEM_CTRL, + SMA1307_POWER_MASK, SMA1307_POWER_OFF); +} + +static int sma1307_aif_in_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + unsigned int mux = sma1307->dapm_aif_in; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (mux) { + case SMA1307_MONO_MODE: + regmap_update_bits(sma1307->regmap, + SMA1307_11_SYSTEM_CTRL2, + SMA1307_MONOMIX_MASK, + SMA1307_MONOMIX_ON); + break; + case SMA1307_LEFT_MODE: + regmap_update_bits(sma1307->regmap, + SMA1307_11_SYSTEM_CTRL2, + SMA1307_MONOMIX_MASK, + SMA1307_MONOMIX_OFF); + regmap_update_bits(sma1307->regmap, + SMA1307_11_SYSTEM_CTRL2, + SMA1307_LR_DATA_SW_MASK, + SMA1307_LR_DATA_SW_NORMAL); + break; + case SMA1307_RIGHT_MODE: + regmap_update_bits(sma1307->regmap, + SMA1307_11_SYSTEM_CTRL2, + SMA1307_MONOMIX_MASK, + SMA1307_MONOMIX_OFF); + regmap_update_bits(sma1307->regmap, + SMA1307_11_SYSTEM_CTRL2, + SMA1307_LR_DATA_SW_MASK, + SMA1307_LR_DATA_SW_SWAP); + break; + default: + + dev_err(sma1307->dev, "%s: Invalid value (%d)\n", + __func__, mux); + return -EINVAL; + } + sma1307->amp_mode = mux; + break; + } + return 0; +} + +static int sma1307_sdo_setting_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + unsigned int mux = sma1307->dapm_sdo_setting; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (mux) { + case SMA1307_OUT_DATA_ONE_48K: + regmap_update_bits(sma1307->regmap, + SMA1307_A2_TOP_MAN1, + SMA1307_SDO_OUTPUT2_MASK, + SMA1307_ONE_SDO_PER_CH); + regmap_update_bits(sma1307->regmap, + SMA1307_A3_TOP_MAN2, + SMA1307_SDO_OUTPUT3_MASK + | + SMA1307_DATA_CLK_SEL_MASK, + SMA1307_SDO_OUTPUT3_DIS + | SMA1307_SDO_DATA); + break; + case SMA1307_OUT_DATA_TWO_48K: + regmap_update_bits(sma1307->regmap, + SMA1307_A2_TOP_MAN1, + SMA1307_SDO_OUTPUT2_MASK, + SMA1307_TWO_SDO_PER_CH); + regmap_update_bits(sma1307->regmap, + SMA1307_A3_TOP_MAN2, + SMA1307_SDO_OUTPUT3_MASK + | + SMA1307_DATA_CLK_SEL_MASK, + SMA1307_SDO_OUTPUT3_DIS + | SMA1307_SDO_DATA); + break; + case SMA1307_OUT_DATA_TWO_24K: + regmap_update_bits(sma1307->regmap, + SMA1307_A2_TOP_MAN1, + SMA1307_SDO_OUTPUT2_MASK, + SMA1307_TWO_SDO_PER_CH); + regmap_update_bits(sma1307->regmap, + SMA1307_A3_TOP_MAN2, + SMA1307_SDO_OUTPUT3_MASK + | + SMA1307_DATA_CLK_SEL_MASK, + SMA1307_TWO_SDO_PER_CH_24K + | SMA1307_SDO_DATA); + break; + case SMA1307_OUT_CLK_PLL: + regmap_update_bits(sma1307->regmap, + SMA1307_A3_TOP_MAN2, + SMA1307_DATA_CLK_SEL_MASK, + SMA1307_SDO_CLK_PLL); + + break; + case SMA1307_OUT_CLK_OSC: + regmap_update_bits(sma1307->regmap, + SMA1307_A3_TOP_MAN2, + SMA1307_DATA_CLK_SEL_MASK, + SMA1307_SDO_CLK_OSC); + + break; + default: + dev_err(sma1307->dev, "%s: Invalid value (%d)\n", + __func__, mux); + return -EINVAL; + } + break; + } + return 0; +} + +static int sma1307_aif_out_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + unsigned int mux = 0, val = 0, mask = 0; + + if (!strcmp(w->name, SMA1307_AIF_OUT0_NAME)) { + mux = sma1307->dapm_aif_out0; + val = mux; + mask = SMA1307_SDO_OUT0_SEL_MASK; + } else if (!strcmp(w->name, SMA1307_AIF_OUT1_NAME)) { + mux = sma1307->dapm_aif_out1; + val = mux << 3; + mask = SMA1307_SDO_OUT1_SEL_MASK; + } else { + dev_err(sma1307->dev, "%s: Invalid widget - %s\n", + __func__, w->name); + return -EINVAL; + } + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(sma1307->regmap, SMA1307_09_OUTPUT_CTRL, + mask, val); + break; + } + return 0; +} + +static int sma1307_sdo_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(sma1307->regmap, + SMA1307_09_OUTPUT_CTRL, + SMA1307_PORT_CONFIG_MASK, + SMA1307_OUTPUT_PORT_ENABLE); + regmap_update_bits(sma1307->regmap, + SMA1307_A3_TOP_MAN2, + SMA1307_SDO_OUTPUT_MASK, + SMA1307_LOGIC_OUTPUT); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(sma1307->regmap, + SMA1307_09_OUTPUT_CTRL, + SMA1307_PORT_CONFIG_MASK, + SMA1307_INPUT_PORT_ONLY); + regmap_update_bits(sma1307->regmap, + SMA1307_A3_TOP_MAN2, + SMA1307_SDO_OUTPUT_MASK, + SMA1307_HIGH_Z_OUTPUT); + break; + } + return 0; +} + +static int sma1307_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + sma1307_startup(component); + break; + case SND_SOC_DAPM_PRE_PMD: + sma1307_shutdown(component); + break; + } + return 0; +} + +static int sma1307_dapm_aif_in_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + + ucontrol->value.enumerated.item[0] = (unsigned int)sma1307->dapm_aif_in; + snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + return 0; +} + +static int sma1307_dapm_aif_in_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + int val = (int)ucontrol->value.enumerated.item[0]; + bool change; + + if ((val < 0) || (val >= ARRAY_SIZE(sma1307_aif_in_source_text))) { + dev_err(sma1307->dev, "%s: Out of range\n", __func__); + return -EINVAL; + } + + if (sma1307->dapm_aif_in != val) { + change = true; + sma1307->dapm_aif_in = val; + } else + change = false; + + snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + return change; +} + +static int sma1307_dapm_sdo_setting_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + + ucontrol->value.enumerated.item[0] = + (unsigned int)sma1307->dapm_sdo_setting; + snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + return 0; +} + +static int sma1307_dapm_sdo_setting_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + int val = (int)ucontrol->value.enumerated.item[0]; + bool change; + + if ((val < 0) || (val >= ARRAY_SIZE(sma1307_sdo_setting_text))) { + dev_err(sma1307->dev, "%s: Out of range\n", __func__); + return -EINVAL; + } + + if (sma1307->dapm_sdo_setting != val) { + change = true; + sma1307->dapm_sdo_setting = val; + } else + change = false; + + snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + return change; +} + +static int sma1307_dapm_aif_out_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + unsigned int val = 0; + + if (!strcmp(kcontrol->id.name, SMA1307_AIF_OUT0_NAME)) { + val = (unsigned int)sma1307->dapm_aif_out0; + } else if (!strcmp(kcontrol->id.name, SMA1307_AIF_OUT1_NAME)) { + val = (unsigned int)sma1307->dapm_aif_out1; + } else { + dev_err(sma1307->dev, "%s: Invalid Control ID - %s\n", + __func__, kcontrol->id.name); + return -EINVAL; + } + ucontrol->value.enumerated.item[0] = val; + snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + return 0; +} + +static int sma1307_dapm_aif_out_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + int val = (int)ucontrol->value.enumerated.item[0]; + bool change; + + if ((val < 0) || (val >= ARRAY_SIZE(sma1307_aif_out_source_text))) { + dev_err(sma1307->dev, "%s: Out of range\n", __func__); + return -EINVAL; + } + + if (!strcmp(kcontrol->id.name, SMA1307_AIF_OUT0_NAME)) { + if (sma1307->dapm_aif_out0 != val) { + change = true; + sma1307->dapm_aif_out0 = val; + } else + change = false; + } else if (!strcmp(kcontrol->id.name, SMA1307_AIF_OUT1_NAME)) { + if (sma1307->dapm_aif_out1 != val) { + change = true; + sma1307->dapm_aif_out1 = val; + } else + change = false; + } else { + dev_err(sma1307->dev, "%s: Invalid Control ID - %s\n", + __func__, kcontrol->id.name); + return -EINVAL; + } + + snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + return change; +} + +static int sma1307_dapm_sdo_enable_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + + ucontrol->value.integer.value[0] = (long)sma1307->dapm_sdo_en; + snd_soc_dapm_put_volsw(kcontrol, ucontrol); + + return 0; +} + +static int sma1307_dapm_sdo_enable_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + int val = (int)ucontrol->value.integer.value[0]; + bool change; + + if ((val < 0) || (val > 1)) { + dev_err(sma1307->dev, "%s: Out of range\n", __func__); + return -EINVAL; + } + + if (sma1307->dapm_sdo_en != val) { + change = true; + sma1307->dapm_sdo_en = val; + } else + change = false; + + snd_soc_dapm_put_volsw(kcontrol, ucontrol); + + return change; +} + +static const struct snd_kcontrol_new sma1307_aif_in_source_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SMA1307_AIF_IN_NAME, + .info = snd_soc_info_enum_double, + .get = sma1307_dapm_aif_in_get, + .put = sma1307_dapm_aif_in_put, + .private_value = (unsigned long)&sma1307_aif_in_source_enum +}; + +static const struct snd_kcontrol_new sma1307_sdo_setting_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SDO Setting", + .info = snd_soc_info_enum_double, + .get = sma1307_dapm_sdo_setting_get, + .put = sma1307_dapm_sdo_setting_put, + .private_value = (unsigned long)&sma1307_sdo_setting_enum +}; + +static const struct snd_kcontrol_new sma1307_aif_out0_source_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SMA1307_AIF_OUT0_NAME, + .info = snd_soc_info_enum_double, + .get = sma1307_dapm_aif_out_get, + .put = sma1307_dapm_aif_out_put, + .private_value = (unsigned long)&sma1307_aif_out_source_enum +}; + +static const struct snd_kcontrol_new sma1307_aif_out1_source_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SMA1307_AIF_OUT1_NAME, + .info = snd_soc_info_enum_double, + .get = sma1307_dapm_aif_out_get, + .put = sma1307_dapm_aif_out_put, + .private_value = (unsigned long)&sma1307_aif_out_source_enum +}; + +static const struct snd_kcontrol_new sma1307_sdo_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Switch", + .info = snd_soc_info_volsw, + .get = sma1307_dapm_sdo_enable_get, + .put = sma1307_dapm_sdo_enable_put, + .private_value = SOC_SINGLE_VALUE(SND_SOC_NOPM, 0, 1, 0, 0) +}; + +static const struct snd_kcontrol_new sma1307_enable_control = + SOC_DAPM_SINGLE("Switch", SMA1307_00_SYSTEM_CTRL, 0, 1, 0); + +static const struct snd_kcontrol_new sma1307_binary_mode_control[] = { + SOC_ENUM_EXT("Binary Mode", sma1307_binary_mode_enum, + snd_soc_get_enum_double, sma1307_binary_mode_put), +}; + +static const struct snd_kcontrol_new sma1307_snd_controls[] = { + SOC_SINGLE_TLV(SMA1307_VOL_CTRL_NAME, SMA1307_0A_SPK_VOL, + 0, 167, 1, sma1307_spk_tlv), + SOC_ENUM_EXT(SMA1307_TDM_RX0_POS_NAME, sma1307_tdm_slot_enum, + sma1307_tdm_slot_get, sma1307_tdm_slot_put), + SOC_ENUM_EXT(SMA1307_TDM_RX1_POS_NAME, sma1307_tdm_slot_enum, + sma1307_tdm_slot_get, sma1307_tdm_slot_put), + SOC_ENUM_EXT(SMA1307_TDM_TX0_POS_NAME, sma1307_tdm_slot_enum, + sma1307_tdm_slot_get, sma1307_tdm_slot_put), + SOC_ENUM_EXT(SMA1307_TDM_TX1_POS_NAME, sma1307_tdm_slot_enum, + sma1307_tdm_slot_get, sma1307_tdm_slot_put), + SOC_ENUM_EXT(SMA1307_RESET_CTRL_NAME, sma1307_reset_enum, + snd_soc_get_enum_double, sma1307_reset_put), + SOC_SINGLE_BOOL_EXT(SMA1307_FORCE_MUTE_CTRL_NAME, 0, + sma1307_force_mute_get, sma1307_force_mute_put), + SOC_SINGLE_BOOL_EXT(SMA1307_OT1_SW_PROT_CTRL_NAME, 0, + sma1307_sw_ot1_prot_get, sma1307_sw_ot1_prot_put), + SOC_SINGLE_BOOL_EXT(SMA1307_CHECK_FAULT_STATUS_NAME, 0, + sma1307_check_fault_status_get, + sma1307_check_fault_status_put), + SOC_SINGLE_EXT(SMA1307_CHECK_FAULT_PERIOD_NAME, SND_SOC_NOPM, 0, 600, 0, + sma1307_check_fault_period_get, + sma1307_check_fault_period_put), +}; + +static const struct snd_soc_dapm_widget sma1307_dapm_widgets[] = { + /* platform domain */ + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_INPUT("SDO"), + + /* path domain */ + SND_SOC_DAPM_MUX_E(SMA1307_AIF_IN_NAME, SND_SOC_NOPM, 0, 0, + &sma1307_aif_in_source_control, + sma1307_aif_in_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("SDO Setting", SND_SOC_NOPM, 0, 0, + &sma1307_sdo_setting_control, + sma1307_sdo_setting_event, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MUX_E(SMA1307_AIF_OUT0_NAME, SND_SOC_NOPM, 0, 0, + &sma1307_aif_out0_source_control, + sma1307_aif_out_event, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MUX_E(SMA1307_AIF_OUT1_NAME, SND_SOC_NOPM, 0, 0, + &sma1307_aif_out1_source_control, + sma1307_aif_out_event, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SWITCH_E("SDO Enable", SND_SOC_NOPM, 0, 0, + &sma1307_sdo_control, + sma1307_sdo_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER("Entry", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV_E("AMP Power", SND_SOC_NOPM, 0, 0, NULL, 0, + sma1307_power_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("AMP Enable", SND_SOC_NOPM, 0, 0, + &sma1307_enable_control), + + /* stream domain */ + SND_SOC_DAPM_AIF_IN("AIF IN", "Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF OUT", "Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route sma1307_audio_map[] = { + /* Playback */ + { "AIF IN Source", "Mono", "AIF IN" }, + { "AIF IN Source", "Left", "AIF IN" }, + { "AIF IN Source", "Right", "AIF IN" }, + + { "SDO Enable", "Switch", "AIF IN" }, + + { "SDO Setting", "Data_One_48k", "SDO Enable" }, + { "SDO Setting", "Data_Two_48k", "SDO Enable" }, + { "SDO Setting", "Data_Two_24k", "SDO Enable" }, + { "SDO Setting", "Clk_PLL", "SDO Enable" }, + { "SDO Setting", "Clk_OSC", "SDO Enable" }, + + { "AIF OUT0 Source", "Disable", "SDO Setting" }, + { "AIF OUT0 Source", "After_FmtC", "SDO Setting" }, + { "AIF OUT0 Source", "After_Mixer", "SDO Setting" }, + { "AIF OUT0 Source", "After_DSP", "SDO Setting" }, + { "AIF OUT0 Source", "Vrms2_Avg", "SDO Setting" }, + { "AIF OUT0 Source", "Battery", "SDO Setting" }, + { "AIF OUT0 Source", "Temperature", "SDO Setting" }, + { "AIF OUT0 Source", "After_Delay", "SDO Setting" }, + + { "AIF OUT1 Source", "Disable", "SDO Setting" }, + { "AIF OUT1 Source", "After_FmtC", "SDO Setting" }, + { "AIF OUT1 Source", "After_Mixer", "SDO Setting" }, + { "AIF OUT1 Source", "After_DSP", "SDO Setting" }, + { "AIF OUT1 Source", "Vrms2_Avg", "SDO Setting" }, + { "AIF OUT1 Source", "Battery", "SDO Setting" }, + { "AIF OUT1 Source", "Temperature", "SDO Setting" }, + { "AIF OUT1 Source", "After_Delay", "SDO Setting" }, + + { "Entry", NULL, "AIF OUT0 Source" }, + { "Entry", NULL, "AIF OUT1 Source" }, + { "Entry", NULL, "AIF IN Source" }, + + { "AMP Power", NULL, "Entry" }, + + { "AMP Enable", "Switch", "AMP Power" }, + { "SPK", NULL, "AMP Enable" }, + + /* Capture */ + { "AIF OUT", NULL, "AMP Enable" }, +}; + +static void sma1307_setup_pll(struct snd_soc_component *component, + unsigned int bclk) +{ + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + int i = 0; + + dev_dbg(component->dev, "%s: BCLK = %dHz\n", __func__, bclk); + + if (sma1307->sys_clk_id == SMA1307_PLL_CLKIN_MCLK) { + dev_warn(component->dev, "%s: MCLK is not supported\n", + __func__); + } else if (sma1307->sys_clk_id == SMA1307_PLL_CLKIN_BCLK) { + for (i = 0; i < sma1307->num_of_pll_matches; i++) { + if (sma1307->pll_matches[i].input_clk == bclk) + break; + } + if (i == sma1307->num_of_pll_matches) { + dev_warn(component->dev, + "%s: No matching value between pll table and SCK\n", + __func__); + return; + } + + regmap_update_bits(sma1307->regmap, + SMA1307_A2_TOP_MAN1, + SMA1307_PLL_MASK, SMA1307_PLL_ON); + } + + regmap_write(sma1307->regmap, SMA1307_8B_PLL_POST_N, + sma1307->pll_matches[i].post_n); + regmap_write(sma1307->regmap, SMA1307_8C_PLL_N, + sma1307->pll_matches[i].n); + regmap_write(sma1307->regmap, SMA1307_8D_PLL_A_SETTING, + sma1307->pll_matches[i].vco); + regmap_write(sma1307->regmap, SMA1307_8E_PLL_P_CP, + sma1307->pll_matches[i].p_cp); +} + +static int sma1307_dai_hw_params_amp(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + unsigned int bclk = 0; + int ret = 0; + + if (sma1307->format == SND_SOC_DAIFMT_DSP_A) + bclk = params_rate(params) * sma1307->frame_size; + else + bclk = params_rate(params) * params_physical_width(params) + * params_channels(params); + + dev_dbg(component->dev, + "%s: rate = %d : bit size = %d : channel = %d\n", + __func__, params_rate(params), params_width(params), + params_channels(params)); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (sma1307->sys_clk_id == SMA1307_PLL_CLKIN_BCLK) { + if (sma1307->last_bclk != bclk) { + sma1307_setup_pll(component, bclk); + sma1307->last_bclk = bclk; + } + } + + switch (params_rate(params)) { + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 44100: + case 48000: + break; + + case 96000: + dev_warn(component->dev, + "%s: %d rate not support SDO\n", __func__, + params_rate(params)); + break; + + default: + dev_err(component->dev, "%s: not support rate : %d\n", + __func__, params_rate(params)); + + return -EINVAL; + } + + /* substream->stream is SNDRV_PCM_STREAM_CAPTURE */ + } else { + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(sma1307->regmap, + SMA1307_A4_TOP_MAN3, + SMA1307_SCK_RATE_MASK + | + SMA1307_DATA_WIDTH_MASK, + SMA1307_SCK_32FS | + SMA1307_DATA_16BIT); + break; + + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(sma1307->regmap, + SMA1307_A4_TOP_MAN3, + SMA1307_SCK_RATE_MASK + | + SMA1307_DATA_WIDTH_MASK, + SMA1307_SCK_64FS | + SMA1307_DATA_24BIT); + break; + + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(sma1307->regmap, + SMA1307_A4_TOP_MAN3, + SMA1307_SCK_RATE_MASK + | + SMA1307_DATA_WIDTH_MASK, + SMA1307_SCK_64FS | + SMA1307_DATA_24BIT); + break; + default: + dev_err(component->dev, + "%s: not support data bit : %d\n", __func__, + params_format(params)); + return -EINVAL; + } + } + + switch (sma1307->format) { + case SND_SOC_DAIFMT_I2S: + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_I2S_MODE_MASK, + SMA1307_STANDARD_I2S); + regmap_update_bits(sma1307->regmap, + SMA1307_A4_TOP_MAN3, + SMA1307_INTERFACE_MASK, + SMA1307_I2S_FORMAT); + break; + case SND_SOC_DAIFMT_LEFT_J: + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_I2S_MODE_MASK, SMA1307_LJ); + regmap_update_bits(sma1307->regmap, + SMA1307_A4_TOP_MAN3, + SMA1307_INTERFACE_MASK, + SMA1307_LJ_FORMAT); + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_width(params)) { + case 16: + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_I2S_MODE_MASK, + SMA1307_RJ_16BIT); + break; + case 24: + case 32: + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_I2S_MODE_MASK, + SMA1307_RJ_24BIT); + break; + } + break; + case SND_SOC_DAIFMT_DSP_A: + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_I2S_MODE_MASK, + SMA1307_STANDARD_I2S); + regmap_update_bits(sma1307->regmap, + SMA1307_A4_TOP_MAN3, + SMA1307_INTERFACE_MASK, + SMA1307_TDM_FORMAT); + break; + } + + switch (params_width(params)) { + case 16: + case 24: + case 32: + break; + default: + dev_err(component->dev, + "%s: not support data bit : %d\n", __func__, + params_format(params)); + return -EINVAL; + } + if (ret < 0) + return -EINVAL; + + return 0; +} + +static int sma1307_dai_set_sysclk_amp(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + switch (clk_id) { + case SMA1307_EXTERNAL_CLOCK_19_2: + case SMA1307_EXTERNAL_CLOCK_24_576: + case SMA1307_PLL_CLKIN_MCLK: + case SMA1307_PLL_CLKIN_BCLK: + break; + default: + dev_err(component->dev, "%s: Invalid clk id: %d\n", + __func__, clk_id); + return -EINVAL; + } + sma1307->sys_clk_id = clk_id; + + return 0; +} + +static int sma1307_dai_set_fmt_amp(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + + case SND_SOC_DAIFMT_CBC_CFC: + dev_dbg(component->dev, + "%s: %s\n", __func__, "I2S/TDM Device mode"); + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_CONTROLLER_DEVICE_MASK, + SMA1307_DEVICE_MODE); + break; + + case SND_SOC_DAIFMT_CBP_CFP: + dev_dbg(component->dev, + "%s: %s\n", __func__, "I2S/TDM Controller mode"); + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_CONTROLLER_DEVICE_MASK, + SMA1307_CONTROLLER_MODE); + break; + + default: + dev_err(component->dev, + "%s: Unsupported Controller/Device : 0x%x\n", + __func__, fmt); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + sma1307->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + dev_err(component->dev, + "%s: Unsupported Audio Interface Format : 0x%x\n", + __func__, fmt); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + + case SND_SOC_DAIFMT_IB_NF: + dev_dbg(component->dev, "%s: %s\n", + __func__, "Invert BCLK + Normal Frame"); + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_SCK_RISING_MASK, + SMA1307_SCK_RISING_EDGE); + break; + case SND_SOC_DAIFMT_IB_IF: + dev_dbg(component->dev, "%s: %s\n", + __func__, "Invert BCLK + Invert Frame"); + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_LEFTPOL_MASK + | SMA1307_SCK_RISING_MASK, + SMA1307_HIGH_FIRST_CH + | SMA1307_SCK_RISING_EDGE); + break; + case SND_SOC_DAIFMT_NB_IF: + dev_dbg(component->dev, "%s: %s\n", + __func__, "Normal BCLK + Invert Frame"); + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_LEFTPOL_MASK, + SMA1307_HIGH_FIRST_CH); + break; + case SND_SOC_DAIFMT_NB_NF: + dev_dbg(component->dev, "%s: %s\n", + __func__, "Normal BCLK + Normal Frame"); + break; + default: + dev_err(component->dev, + "%s: Unsupported Bit & Frameclock : 0x%x\n", + __func__, fmt); + return -EINVAL; + } + + return 0; +} + +static int sma1307_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "%s: slots = %d, slot_width - %d\n", + __func__, slots, slot_width); + + sma1307->frame_size = slot_width * slots; + + regmap_update_bits(sma1307->regmap, + SMA1307_A4_TOP_MAN3, + SMA1307_INTERFACE_MASK, SMA1307_TDM_FORMAT); + + regmap_update_bits(sma1307->regmap, + SMA1307_A5_TDM1, + SMA1307_TDM_TX_MODE_MASK, + SMA1307_TDM_TX_MONO); + + switch (slot_width) { + case 16: + regmap_update_bits(sma1307->regmap, + SMA1307_A6_TDM2, + SMA1307_TDM_DL_MASK, + SMA1307_TDM_DL_16); + break; + case 32: + regmap_update_bits(sma1307->regmap, + SMA1307_A6_TDM2, + SMA1307_TDM_DL_MASK, + SMA1307_TDM_DL_32); + break; + default: + dev_err(component->dev, "%s: not support TDM %d slot_width\n", + __func__, slot_width); + return -EINVAL; + } + + switch (slots) { + case 4: + regmap_update_bits(sma1307->regmap, + SMA1307_A6_TDM2, + SMA1307_TDM_N_SLOT_MASK, + SMA1307_TDM_N_SLOT_4); + break; + case 8: + regmap_update_bits(sma1307->regmap, + SMA1307_A6_TDM2, + SMA1307_TDM_N_SLOT_MASK, + SMA1307_TDM_N_SLOT_8); + break; + default: + dev_err(component->dev, "%s: not support TDM %d slots\n", + __func__, slots); + return -EINVAL; + } + + if (sma1307->tdm_slot0_rx < slots) + regmap_update_bits(sma1307->regmap, + SMA1307_A5_TDM1, + SMA1307_TDM_SLOT0_RX_POS_MASK, + sma1307->tdm_slot0_rx << 3); + else + dev_err(component->dev, "%s: Incorrect tdm-slot0-rx %d set\n", + __func__, sma1307->tdm_slot0_rx); + + if (sma1307->tdm_slot1_rx < slots) + regmap_update_bits(sma1307->regmap, + SMA1307_A5_TDM1, + SMA1307_TDM_SLOT1_RX_POS_MASK, + sma1307->tdm_slot1_rx); + else + dev_err(component->dev, "%s: Incorrect tdm-slot1-rx %d set\n", + __func__, sma1307->tdm_slot1_rx); + + if (sma1307->tdm_slot0_tx < slots) + regmap_update_bits(sma1307->regmap, + SMA1307_A6_TDM2, + SMA1307_TDM_SLOT0_TX_POS_MASK, + sma1307->tdm_slot0_tx << 3); + else + dev_err(component->dev, "%s: Incorrect tdm-slot0-tx %d set\n", + __func__, sma1307->tdm_slot0_tx); + + if (sma1307->tdm_slot1_tx < slots) + regmap_update_bits(sma1307->regmap, + SMA1307_A6_TDM2, + SMA1307_TDM_SLOT1_TX_POS_MASK, + sma1307->tdm_slot1_tx); + else + dev_err(component->dev, "%s: Incorrect tdm-slot1-tx %d set\n", + __func__, sma1307->tdm_slot1_tx); + + return 0; +} + +static int sma1307_dai_mute_stream(struct snd_soc_dai *dai, int mute, + int stream) +{ + struct snd_soc_component *component = dai->component; + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + if (stream == SNDRV_PCM_STREAM_CAPTURE) + return 0; + if (mute) { + dev_dbg(component->dev, "%s: %s\n", __func__, "MUTE"); + regmap_update_bits(sma1307->regmap, + SMA1307_0E_MUTE_VOL_CTRL, + SMA1307_SPK_MUTE_MASK, + SMA1307_SPK_MUTE); + } else { + if (!sma1307->force_mute_status) { + dev_dbg(component->dev, "%s: %s\n", __func__, + "UNMUTE"); + regmap_update_bits(sma1307->regmap, + SMA1307_0E_MUTE_VOL_CTRL, + SMA1307_SPK_MUTE_MASK, + SMA1307_SPK_UNMUTE); + } else { + dev_dbg(sma1307->dev, "%s: FORCE MUTE!!!\n", __func__); + } + } + + return 0; +} + +static const struct snd_soc_dai_ops sma1307_dai_ops_amp = { + .hw_params = sma1307_dai_hw_params_amp, + .set_fmt = sma1307_dai_set_fmt_amp, + .set_sysclk = sma1307_dai_set_sysclk_amp, + .set_tdm_slot = sma1307_dai_set_tdm_slot, + .mute_stream = sma1307_dai_mute_stream, +}; + +#define SMA1307_RATES_PLAYBACK SNDRV_PCM_RATE_8000_96000 +#define SMA1307_RATES_CAPTURE SNDRV_PCM_RATE_8000_48000 +#define SMA1307_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver sma1307_dai[] = { + { + .name = "sma1307-amplifier", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SMA1307_RATES_PLAYBACK, + .formats = SMA1307_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SMA1307_RATES_CAPTURE, + .formats = SMA1307_FORMATS, + }, + .ops = &sma1307_dai_ops_amp, + }, +}; + +static void sma1307_check_fault_worker(struct work_struct *work) +{ + struct sma1307_priv *sma1307 = + container_of(work, struct sma1307_priv, check_fault_work.work); + unsigned int status1_val, status2_val; + char *envp[3] = { NULL, NULL, NULL }; + + if (sma1307->tsdw_cnt) + regmap_read(sma1307->regmap, + SMA1307_0A_SPK_VOL, &sma1307->cur_vol); + else + regmap_read(sma1307->regmap, + SMA1307_0A_SPK_VOL, &sma1307->init_vol); + + regmap_read(sma1307->regmap, SMA1307_FA_STATUS1, &status1_val); + regmap_read(sma1307->regmap, SMA1307_FB_STATUS2, &status2_val); + + if (~status1_val & SMA1307_OT1_OK_STATUS) { + dev_crit(sma1307->dev, + "%s: OT1(Over Temperature Level 1)\n", __func__); + envp[0] = kasprintf(GFP_KERNEL, "STATUS=OT1"); + if (sma1307->sw_ot1_prot) { + /* Volume control (Current Volume -3dB) */ + if ((sma1307->cur_vol + 6) <= 0xFA) { + sma1307->cur_vol += 6; + regmap_write(sma1307->regmap, + SMA1307_0A_SPK_VOL, + sma1307->cur_vol); + envp[1] = kasprintf(GFP_KERNEL, + "VOLUME=0x%02X", sma1307->cur_vol); + } + } + sma1307->tsdw_cnt++; + } else if (sma1307->tsdw_cnt) { + regmap_write(sma1307->regmap, + SMA1307_0A_SPK_VOL, sma1307->init_vol); + sma1307->tsdw_cnt = 0; + sma1307->cur_vol = sma1307->init_vol; + envp[0] = kasprintf(GFP_KERNEL, "STATUS=OT1_CLEAR"); + envp[1] = kasprintf(GFP_KERNEL, + "VOLUME=0x%02X", sma1307->cur_vol); + } + + if (~status1_val & SMA1307_OT2_OK_STATUS) { + dev_crit(sma1307->dev, + "%s: OT2(Over Temperature Level 2)\n", __func__); + envp[0] = kasprintf(GFP_KERNEL, "STATUS=OT2"); + } + if (status1_val & SMA1307_UVLO_STATUS) { + dev_crit(sma1307->dev, + "%s: UVLO(Under Voltage Lock Out)\n", __func__); + envp[0] = kasprintf(GFP_KERNEL, "STATUS=UVLO"); + } + if (status1_val & SMA1307_OVP_BST_STATUS) { + dev_crit(sma1307->dev, + "%s: OVP_BST(Over Voltage Protection)\n", __func__); + envp[0] = kasprintf(GFP_KERNEL, "STATUS=OVP_BST"); + } + if (status2_val & SMA1307_OCP_SPK_STATUS) { + dev_crit(sma1307->dev, + "%s: OCP_SPK(Over Current Protect SPK)\n", __func__); + envp[0] = kasprintf(GFP_KERNEL, "STATUS=OCP_SPK"); + } + if (status2_val & SMA1307_OCP_BST_STATUS) { + dev_crit(sma1307->dev, + "%s: OCP_BST(Over Current Protect Boost)\n", __func__); + envp[0] = kasprintf(GFP_KERNEL, "STATUS=OCP_BST"); + } + if (status2_val & SMA1307_CLK_MON_STATUS) { + dev_crit(sma1307->dev, + "%s: CLK_FAULT(No clock input)\n", __func__); + envp[0] = kasprintf(GFP_KERNEL, "STATUS=CLK_FAULT"); + } + + if (envp[0] != NULL) { + if (kobject_uevent_env(sma1307->kobj, KOBJ_CHANGE, envp)) + dev_err(sma1307->dev, + "%s: Error sending uevent\n", __func__); + kfree(envp[0]); + kfree(envp[1]); + } + + if (sma1307->check_fault_status) { + if (sma1307->check_fault_period > 0) + queue_delayed_work(system_freezable_wq, + &sma1307->check_fault_work, + sma1307->check_fault_period * HZ); + else + queue_delayed_work(system_freezable_wq, + &sma1307->check_fault_work, + CHECK_PERIOD_TIME * HZ); + } +} + +static void sma1307_setting_loaded(struct sma1307_priv *sma1307, const char *file) +{ + const struct firmware *fw; + int *data, size, offset, num_mode; + + request_firmware(&fw, file, sma1307->dev); + + if (!fw) { + dev_err(sma1307->dev, "%s: failed to read \"%s\"\n", + __func__, setting_file); + release_firmware(fw); + sma1307->set.status = false; + return; + } else if ((fw->size) < SMA1307_SETTING_HEADER_SIZE) { + dev_err(sma1307->dev, "%s: Invalid file\n", __func__); + release_firmware(fw); + sma1307->set.status = false; + return; + } + + data = kzalloc(fw->size, GFP_KERNEL); + size = fw->size >> 2; + memcpy(data, fw->data, fw->size); + + release_firmware(fw); + + /* HEADER */ + sma1307->set.header_size = SMA1307_SETTING_HEADER_SIZE; + sma1307->set.checksum = data[sma1307->set.header_size - 2]; + sma1307->set.num_mode = data[sma1307->set.header_size - 1]; + num_mode = sma1307->set.num_mode; + sma1307->set.header = devm_kzalloc(sma1307->dev, + sma1307->set.header_size, + GFP_KERNEL); + memcpy(sma1307->set.header, data, + sma1307->set.header_size * sizeof(int)); + + if ((sma1307->set.checksum >> 8) != SMA1307_SETTING_CHECKSUM) { + dev_err(sma1307->dev, "%s: failed by dismatch \"%s\"\n", + __func__, setting_file); + sma1307->set.status = false; + return; + } + + /* DEFAULT */ + sma1307->set.def_size = SMA1307_SETTING_DEFAULT_SIZE; + sma1307->set.def + = devm_kzalloc(sma1307->dev, + sma1307->set.def_size * sizeof(int), GFP_KERNEL); + memcpy(sma1307->set.def, + &data[sma1307->set.header_size], + sma1307->set.def_size * sizeof(int)); + + /* MODE */ + offset = sma1307->set.header_size + sma1307->set.def_size; + sma1307->set.mode_size = DIV_ROUND_CLOSEST(size - offset, num_mode + 1); + for (int i = 0; i < num_mode; i++) { + sma1307->set.mode_set[i] + = devm_kzalloc(sma1307->dev, + sma1307->set.mode_size * 2 * sizeof(int), + GFP_KERNEL); + for (int j = 0; j < sma1307->set.mode_size; j++) { + sma1307->set.mode_set[i][2 * j] + = data[offset + ((num_mode + 1) * j)]; + sma1307->set.mode_set[i][2 * j + 1] + = data[offset + ((num_mode + 1) * j + i + 1)]; + } + } + + kfree(data); + sma1307->set.status = true; + +} + +static void sma1307_reset(struct snd_soc_component *component) +{ + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + unsigned int status = 0; + + regmap_read(sma1307->regmap, SMA1307_FF_DEVICE_INDEX, &status); + + sma1307->rev_num = status & SMA1307_REV_NUM_STATUS; + dev_dbg(component->dev, "%s: SMA1307 Revision %d\n", + __func__, sma1307->rev_num); + regmap_read(sma1307->regmap, SMA1307_99_OTP_TRM2, &sma1307->otp_trm2); + regmap_read(sma1307->regmap, SMA1307_9A_OTP_TRM3, &sma1307->otp_trm3); + + if ((sma1307->otp_trm2 & SMA1307_OTP_STAT_MASK) != SMA1307_OTP_STAT_1) + dev_warn(component->dev, "%s: SMA1307 OTP Status Fail\n", + __func__); + + /* Register Initial Value Setting */ + sma1307_setting_loaded(sma1307, setting_file); + if (sma1307->set.status) + sma1307_set_binary(component); + else + sma1307_set_default(component); + + regmap_update_bits(sma1307->regmap, + SMA1307_93_INT_CTRL, + SMA1307_DIS_INT_MASK, SMA1307_HIGH_Z_INT); + regmap_write(sma1307->regmap, SMA1307_0A_SPK_VOL, sma1307->init_vol); +} + +static void sma1307_set_binary(struct snd_soc_component *component) +{ + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + int i = 0, mode = 0; + + for (i = 0; i < (sma1307->set.def_size); i++) { + if (sma1307_writeable_register(sma1307->dev, i) + && ((i < SMA1307_97_OTP_TRM0) + || (i > SMA1307_9A_OTP_TRM3))) { + regmap_write(sma1307->regmap, i, sma1307->set.def[i]); + + } + } + for (i = 0; i < (sma1307->set.mode_size); i++) { + if (sma1307_writeable_register(sma1307->dev, i) + && ((i < SMA1307_97_OTP_TRM0) + || (i > SMA1307_9A_OTP_TRM3))) { + mode = sma1307->binary_mode; + regmap_write(sma1307->regmap, + sma1307->set.mode_set[mode][2 * i], + sma1307->set.mode_set[mode][2 * i + + 1]); + } + } +} + +static void sma1307_set_default(struct snd_soc_component *component) +{ + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + int i = 0; + + for (i = 0; i < (unsigned int)ARRAY_SIZE(sma1307_reg_def); i++) + regmap_write(sma1307->regmap, + sma1307_reg_def[i].reg, + sma1307_reg_def[i].def); + + if (!strcmp(sma1307->name, DEVICE_NAME_SMA1307AQ)) + sma1307->data->init(sma1307->regmap); +} + +static int sma1307_probe(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + snd_soc_dapm_sync(dapm); + + sma1307_amp_component = component; + + snd_soc_add_component_controls(component, sma1307_binary_mode_control, + ARRAY_SIZE(sma1307_binary_mode_control)); + sma1307_reset(component); + + return 0; +} + +static const struct snd_soc_component_driver sma1307_component = { + .probe = sma1307_probe, + .controls = sma1307_snd_controls, + .num_controls = ARRAY_SIZE(sma1307_snd_controls), + .dapm_widgets = sma1307_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sma1307_dapm_widgets), + .dapm_routes = sma1307_audio_map, + .num_dapm_routes = ARRAY_SIZE(sma1307_audio_map), +}; + +static const struct regmap_config sma_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = SMA1307_FF_DEVICE_INDEX, + .readable_reg = sma1307_readable_register, + .writeable_reg = sma1307_writeable_register, + .volatile_reg = sma1307_volatile_register, + + .reg_defaults = sma1307_reg_def, + .num_reg_defaults = ARRAY_SIZE(sma1307_reg_def), +}; + +static void sma1307aq_init(struct regmap *regmap) +{ + /* Guidelines for driving 4ohm load */ + /* Brown Out Protection */ + regmap_write(regmap, SMA1307_02_BROWN_OUT_PROT1, 0x62); + regmap_write(regmap, SMA1307_03_BROWN_OUT_PROT2, 0x5D); + regmap_write(regmap, SMA1307_04_BROWN_OUT_PROT3, 0x57); + regmap_write(regmap, SMA1307_05_BROWN_OUT_PROT8, 0x54); + regmap_write(regmap, SMA1307_06_BROWN_OUT_PROT9, 0x51); + regmap_write(regmap, + SMA1307_07_BROWN_OUT_PROT10, 0x4D); + regmap_write(regmap, + SMA1307_08_BROWN_OUT_PROT11, 0x4B); + regmap_write(regmap, SMA1307_27_BROWN_OUT_PROT4, 0x3C); + regmap_write(regmap, SMA1307_28_BROWN_OUT_PROT5, 0x5B); + regmap_write(regmap, + SMA1307_29_BROWN_OUT_PROT12, 0x78); + regmap_write(regmap, + SMA1307_2A_BROWN_OUT_PROT13, 0x96); + regmap_write(regmap, + SMA1307_2B_BROWN_OUT_PROT14, 0xB4); + regmap_write(regmap, + SMA1307_2C_BROWN_OUT_PROT15, 0xD3); + /* FDPEC Gain */ + regmap_write(regmap, SMA1307_35_FDPEC_CTRL0, 0x16); + /* FLT Vdd */ + regmap_write(regmap, SMA1307_92_FDPEC_CTRL1, 0xA0); + /* Boost Max */ + regmap_write(regmap, SMA1307_AB_BOOST_CTRL4, 0x0F); +} + +static const struct sma1307_data sma1307aq_data = { + .name = DEVICE_NAME_SMA1307AQ, + .init = sma1307aq_init, +}; + +static int sma1307_i2c_probe(struct i2c_client *client) +{ + struct sma1307_priv *sma1307; + const struct sma1307_data *data; + int ret = 0; + unsigned int device_info; + + sma1307 = devm_kzalloc(&client->dev, + sizeof(*sma1307), GFP_KERNEL); + if (!sma1307) + return -ENOMEM; + + sma1307->regmap = devm_regmap_init_i2c(client, &sma_i2c_regmap); + if (IS_ERR(sma1307->regmap)) { + return dev_err_probe(&client->dev, PTR_ERR(sma1307->regmap), + "%s: failed to allocate register map\n", __func__); + } + + data = device_get_match_data(&client->dev); + if (!data) + return -ENODEV; + + sma1307->data = data; + + /* set initial value as normal AMP IC status */ + sma1307->name = client->name; + sma1307->format = SND_SOC_DAIFMT_I2S; + sma1307->sys_clk_id = SMA1307_PLL_CLKIN_BCLK; + sma1307->num_of_pll_matches = ARRAY_SIZE(sma1307_pll_matches); + + sma1307->check_fault_period = CHECK_PERIOD_TIME; + sma1307->check_fault_status = true; + sma1307->init_vol = 0x32; + sma1307->cur_vol = sma1307->init_vol; + sma1307->sw_ot1_prot = true; + + mutex_init(&sma1307->default_lock); + + INIT_DELAYED_WORK(&sma1307->check_fault_work, + sma1307_check_fault_worker); + + sma1307->dev = &client->dev; + sma1307->kobj = &client->dev.kobj; + + i2c_set_clientdata(client, sma1307); + + sma1307->pll_matches = sma1307_pll_matches; + + regmap_read(sma1307->regmap, + SMA1307_FF_DEVICE_INDEX, &device_info); + + if ((device_info & 0xF8) != SMA1307_DEVICE_ID) { + dev_err(&client->dev, + "%s: device initialization error (0x%02X)", + __func__, device_info); + return -ENODEV; + } + dev_dbg(&client->dev, "%s: chip version 0x%02X\n", + __func__, device_info); + + i2c_set_clientdata(client, sma1307); + + ret = devm_snd_soc_register_component(&client->dev, + &sma1307_component, sma1307_dai, + 1); + + if (ret) { + dev_err(&client->dev, "%s: failed to register component\n", + __func__); + + return ret; + } + + return ret; +} + +static void sma1307_i2c_remove(struct i2c_client *client) +{ + struct sma1307_priv *sma1307 = + (struct sma1307_priv *)i2c_get_clientdata(client); + + cancel_delayed_work_sync(&sma1307->check_fault_work); +} + +static const struct i2c_device_id sma1307_i2c_id[] = { + { "sma1307a", 0 }, + { "sma1307aq", 0 }, + { } +}; + +MODULE_DEVICE_TABLE(i2c, sma1307_i2c_id); + +static const struct of_device_id sma1307_of_match[] = { + { + .compatible = "irondevice,sma1307a", + }, + { + .compatible = "irondevice,sma1307aq", + .data = &sma1307aq_data //AEC-Q100 Qualificated + }, + { } +}; + +MODULE_DEVICE_TABLE(of, sma1307_of_match); + +static struct i2c_driver sma1307_i2c_driver = { + .driver = { + .name = "sma1307", + .of_match_table = sma1307_of_match, + }, + .probe = sma1307_i2c_probe, + .remove = sma1307_i2c_remove, + .id_table = sma1307_i2c_id, +}; + +module_i2c_driver(sma1307_i2c_driver); + +MODULE_DESCRIPTION("ALSA SoC SMA1307 driver"); +MODULE_AUTHOR("Gyuhwa Park, "); +MODULE_AUTHOR("KS Jo, "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sma1307.h b/sound/soc/codecs/sma1307.h new file mode 100644 index 00000000000000..44aab52a32f9b1 --- /dev/null +++ b/sound/soc/codecs/sma1307.h @@ -0,0 +1,444 @@ +/* SPDX-License-Identifier: GPL-2.0-or-later + * sma1307.h -- sma1307 ALSA SoC Audio driver + * + * Copyright 2024 Iron Device Corporation + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _SMA1307_H +#define _SMA1307_H + +#include + +enum sma1307_fault { + SMA1307_FAULT_OT1, + SMA1307_FAULT_OT2, + SMA1307_FAULT_UVLO, + SMA1307_FAULT_OVP_BST, + SMA1307_FAULT_OCP_SPK, + SMA1307_FAULT_OCP_BST, + SMA1307_FAULT_CLK +}; + +enum sma1307_mode { + SMA1307_MONO_MODE, + SMA1307_LEFT_MODE, + SMA1307_RIGHT_MODE, +}; + +enum sma1307_sdo_mode { + SMA1307_OUT_DATA_ONE_48K, + SMA1307_OUT_DATA_TWO_48K, + SMA1307_OUT_DATA_TWO_24K, + SMA1307_OUT_CLK_PLL, + SMA1307_OUT_CLK_OSC +}; + +enum sma1307_sdo_source { + SMA1307_OUT_DISABLE, + SMA1307_OUT_FORMAT_C, + SMA1307_OUT_MIXER_OUT, + SMA1307_OUT_AFTER_DSP, + SMA1307_OUT_VRMS2_AVG, + SMA1307_OUT_BATTERY, + SMA1307_OUT_TEMP, + SMA1307_OUT_AFTER_DELAY +}; + +struct sma1307_setting_file { + bool status; + char *header; + int *def; + int *mode_set[5]; + int checksum; + int num_mode; + size_t header_size; + size_t def_size; + size_t mode_size; +}; + +#define SMA1307_I2C_ADDR_00 0x1e +#define SMA1307_I2C_ADDR_01 0x3e +#define SMA1307_I2C_ADDR_10 0x5e +#define SMA1307_I2C_ADDR_11 0x7e + +#define DEVICE_NAME_SMA1307A "sma1307a" +#define DEVICE_NAME_SMA1307AQ "sma1307aq" + +#define SMA1307_EXTERNAL_CLOCK_19_2 0x00 +#define SMA1307_EXTERNAL_CLOCK_24_576 0x01 +#define SMA1307_PLL_CLKIN_MCLK 0x02 +#define SMA1307_PLL_CLKIN_BCLK 0x03 + +#define SMA1307_OFFSET_DEFAULT_MODE 0x00 +#define SMA1307_OFFSET_BURNING_MODE 0x01 + +#define SMA1307_SETTING_HEADER_SIZE 0x08 +#define SMA1307_SETTING_DEFAULT_SIZE 0xC0 + +#define SMA1307_DEFAULT_SET 0x00 +#define SMA1307_BINARY_FILE_SET 0x01 + +/* Controls Name */ +#define SMA1307_REG_CTRL_NAME "Register Byte Control" +#define SMA1307_VOL_CTRL_NAME "Speaker Volume" +#define SMA1307_FORCE_MUTE_CTRL_NAME "Force Mute Switch" +#define SMA1307_TDM_RX0_POS_NAME "TDM RX Slot0 Position" +#define SMA1307_TDM_RX1_POS_NAME "TDM RX Slot1 Position" +#define SMA1307_TDM_TX0_POS_NAME "TDM TX Slot0 Position" +#define SMA1307_TDM_TX1_POS_NAME "TDM TX Slot1 Position" +#define SMA1307_OT1_SW_PROT_CTRL_NAME "OT1 SW Protection Switch" +#define SMA1307_RESET_CTRL_NAME "Reset Switch" +#define SMA1307_CHECK_FAULT_STATUS_NAME "Check Fault Status" +#define SMA1307_CHECK_FAULT_PERIOD_NAME "Check Fault Period" + +/* DAPM Name */ +#define SMA1307_AIF_IN_NAME "AIF IN Source" +#define SMA1307_AIF_OUT0_NAME "AIF OUT0 Source" +#define SMA1307_AIF_OUT1_NAME "AIF OUT1 Source" + +/* + * SMA1307 Register Definition + */ + +/* SMA1307 Register Addresses */ +#define SMA1307_00_SYSTEM_CTRL 0x00 +#define SMA1307_01_INPUT_CTRL1 0x01 +#define SMA1307_02_BROWN_OUT_PROT1 0x02 +#define SMA1307_03_BROWN_OUT_PROT2 0x03 +#define SMA1307_04_BROWN_OUT_PROT3 0x04 +#define SMA1307_05_BROWN_OUT_PROT8 0x05 +#define SMA1307_06_BROWN_OUT_PROT9 0x06 +#define SMA1307_07_BROWN_OUT_PROT10 0x07 +#define SMA1307_08_BROWN_OUT_PROT11 0x08 +#define SMA1307_09_OUTPUT_CTRL 0x09 +#define SMA1307_0A_SPK_VOL 0x0A +#define SMA1307_0B_BST_TEST 0x0B +#define SMA1307_0C_BOOST_CTRL8 0x0C +#define SMA1307_0D_SPK_TEST 0x0D +#define SMA1307_0E_MUTE_VOL_CTRL 0x0E +#define SMA1307_0F_VBAT_TEMP_SENSING 0x0F + +#define SMA1307_10_SYSTEM_CTRL1 0x10 +#define SMA1307_11_SYSTEM_CTRL2 0x11 +#define SMA1307_12_SYSTEM_CTRL3 0x12 +#define SMA1307_13_DELAY 0x13 +#define SMA1307_14_MODULATOR 0x14 +#define SMA1307_15_BASS_SPK1 0x15 +#define SMA1307_16_BASS_SPK2 0x16 +#define SMA1307_17_BASS_SPK3 0x17 +#define SMA1307_18_BASS_SPK4 0x18 +#define SMA1307_19_BASS_SPK5 0x19 +#define SMA1307_1A_BASS_SPK6 0x1A +#define SMA1307_1B_BASS_SPK7 0x1B +#define SMA1307_1C_BROWN_OUT_PROT20 0x1C +#define SMA1307_1D_BROWN_OUT_PROT0 0x1D +#define SMA1307_1E_TONE_GENERATOR 0x1E +#define SMA1307_1F_TONE_FINE_VOLUME 0x1F + +#define SMA1307_22_COMP_HYS_SEL 0x22 +#define SMA1307_23_COMPLIM1 0x23 +#define SMA1307_24_COMPLIM2 0x24 +#define SMA1307_25_COMPLIM3 0x25 +#define SMA1307_26_COMPLIM4 0x26 +#define SMA1307_27_BROWN_OUT_PROT4 0x27 +#define SMA1307_28_BROWN_OUT_PROT5 0x28 +#define SMA1307_29_BROWN_OUT_PROT12 0x29 +#define SMA1307_2A_BROWN_OUT_PROT13 0x2A +#define SMA1307_2B_BROWN_OUT_PROT14 0x2B +#define SMA1307_2C_BROWN_OUT_PROT15 0x2C +#define SMA1307_2D_BROWN_OUT_PROT6 0x2D +#define SMA1307_2E_BROWN_OUT_PROT7 0x2E +#define SMA1307_2F_BROWN_OUT_PROT16 0x2F + +#define SMA1307_30_BROWN_OUT_PROT17 0x30 +#define SMA1307_31_BROWN_OUT_PROT18 0x31 +#define SMA1307_32_BROWN_OUT_PROT19 0x32 +#define SMA1307_34_OCP_SPK 0x34 +#define SMA1307_35_FDPEC_CTRL0 0x35 +#define SMA1307_36_PROTECTION 0x36 +#define SMA1307_37_SLOPECTRL 0x37 +#define SMA1307_38_POWER_METER 0x38 +#define SMA1307_39_PMT_NZ_VAL 0x39 +#define SMA1307_3B_TEST1 0x3B +#define SMA1307_3C_TEST2 0x3C +#define SMA1307_3D_TEST3 0x3D +#define SMA1307_3E_IDLE_MODE_CTRL 0x3E +#define SMA1307_3F_ATEST2 0x3F +#define SMA1307_8B_PLL_POST_N 0x8B +#define SMA1307_8C_PLL_N 0x8C +#define SMA1307_8D_PLL_A_SETTING 0x8D +#define SMA1307_8E_PLL_P_CP 0x8E +#define SMA1307_8F_ANALOG_TEST 0x8F + +#define SMA1307_90_CRESTLIM1 0x90 +#define SMA1307_91_CRESTLIM2 0x91 +#define SMA1307_92_FDPEC_CTRL1 0x92 +#define SMA1307_93_INT_CTRL 0x93 +#define SMA1307_94_BOOST_CTRL9 0x94 +#define SMA1307_95_BOOST_CTRL10 0x95 +#define SMA1307_96_BOOST_CTRL11 0x96 +#define SMA1307_97_OTP_TRM0 0x97 +#define SMA1307_98_OTP_TRM1 0x98 +#define SMA1307_99_OTP_TRM2 0x99 +#define SMA1307_9A_OTP_TRM3 0x9A + +#define SMA1307_A0_PAD_CTRL0 0xA0 +#define SMA1307_A1_PAD_CTRL1 0xA1 +#define SMA1307_A2_TOP_MAN1 0xA2 +#define SMA1307_A3_TOP_MAN2 0xA3 +#define SMA1307_A4_TOP_MAN3 0xA4 +#define SMA1307_A5_TDM1 0xA5 +#define SMA1307_A6_TDM2 0xA6 +#define SMA1307_A7_CLK_MON 0xA7 +#define SMA1307_A8_BOOST_CTRL1 0xA8 +#define SMA1307_A9_BOOST_CTRL2 0xA9 +#define SMA1307_AA_BOOST_CTRL3 0xAA +#define SMA1307_AB_BOOST_CTRL4 0xAB +#define SMA1307_AC_BOOST_CTRL5 0xAC +#define SMA1307_AD_BOOST_CTRL6 0xAD +#define SMA1307_AE_BOOST_CTRL7 0xAE +#define SMA1307_AF_LPF 0xAF + +#define SMA1307_B0_RMS_TC1 0xB0 +#define SMA1307_B1_RMS_TC2 0xB1 +#define SMA1307_B2_AVG_TC1 0xB2 +#define SMA1307_B3_AVG_TC2 0xB3 +#define SMA1307_B4_PRVALUE1 0xB4 +#define SMA1307_B5_PRVALUE2 0xB5 +#define SMA1307_B8_SPK_NG_CTRL1 0xB8 +#define SMA1307_B9_SPK_NG_CTRL2 0xB9 +#define SMA1307_BA_DGC1 0xBA +#define SMA1307_BB_DGC2 0xBB +#define SMA1307_BC_DGC3 0xBC +#define SMA1307_BD_MCBS_CTRL1 0xBD +#define SMA1307_BE_MCBS_CTRL2 0xBE + +/* Status Register Read Only */ +#define SMA1307_F5_READY_FOR_V_SAR 0xF5 +#define SMA1307_F7_READY_FOR_T_SAR 0xF7 +#define SMA1307_F8_STATUS_T1 0xF8 +#define SMA1307_F9_STATUS_T2 0xF9 +#define SMA1307_FA_STATUS1 0xFA +#define SMA1307_FB_STATUS2 0xFB +#define SMA1307_FC_STATUS3 0xFC +#define SMA1307_FD_STATUS4 0xFD +#define SMA1307_FE_STATUS5 0xFE +#define SMA1307_FF_DEVICE_INDEX 0xFF + +/* SMA1307 Registers Bit Fields */ +/* Power On/Off */ +#define SMA1307_POWER_MASK BIT(0) +#define SMA1307_POWER_OFF 0 +#define SMA1307_POWER_ON BIT(0) + +/* Reset */ +#define SMA1307_RESET_MASK BIT(1) +#define SMA1307_RESET_ON BIT(1) + +/* Left Polarity */ +#define SMA1307_LEFTPOL_MASK BIT(3) +#define SMA1307_LOW_FIRST_CH 0 +#define SMA1307_HIGH_FIRST_CH BIT(3) + +/* SCK Falling/Rising */ +#define SMA1307_SCK_RISING_MASK BIT(2) +#define SMA1307_SCK_FALLING_EDGE 0 +#define SMA1307_SCK_RISING_EDGE BIT(2) + +/* SPK Mute */ +#define SMA1307_SPK_MUTE_MASK BIT(0) +#define SMA1307_SPK_UNMUTE 0 +#define SMA1307_SPK_MUTE BIT(0) + +/* SPK Mode */ +#define SMA1307_SPK_MODE_MASK (BIT(2)|BIT(3)|BIT(4)) +#define SMA1307_SPK_OFF 0 +#define SMA1307_SPK_MONO BIT(2) +#define SMA1307_SPK_STEREO BIT(4) + +/* Mono Mix */ +#define SMA1307_MONOMIX_MASK BIT(0) +#define SMA1307_MONOMIX_OFF 0 +#define SMA1307_MONOMIX_ON BIT(0) + +/* LR Data Swap */ +#define SMA1307_LR_DATA_SW_MASK BIT(4) +#define SMA1307_LR_DATA_SW_NORMAL 0 +#define SMA1307_LR_DATA_SW_SWAP BIT(4) + +/* PLL On/Off */ +#define SMA1307_PLL_MASK BIT(6) +#define SMA1307_PLL_ON 0 +#define SMA1307_PLL_OFF BIT(6) + +/* Input Format */ +#define SMA1307_I2S_MODE_MASK (BIT(4)|BIT(5)|BIT(6)) +#define SMA1307_STANDARD_I2S 0 +#define SMA1307_LJ BIT(4) +#define SMA1307_RJ_16BIT BIT(6) +#define SMA1307_RJ_18BIT (BIT(4)|BIT(6)) +#define SMA1307_RJ_20BIT (BIT(5)|BIT(6)) +#define SMA1307_RJ_24BIT (BIT(4)|BIT(5)|BIT(6)) + +/* Controller / Device Setting */ +#define SMA1307_CONTROLLER_DEVICE_MASK BIT(7) +#define SMA1307_DEVICE_MODE 0 +#define SMA1307_CONTROLLER_MODE BIT(7) + +/* Port Config */ +#define SMA1307_PORT_CONFIG_MASK (BIT(6)|BIT(7)) +#define SMA1307_INPUT_PORT_ONLY 0 +#define SMA1307_OUTPUT_PORT_ENABLE BIT(7) + +/* SDO Output */ +#define SMA1307_SDO_OUTPUT_MASK BIT(3) +#define SMA1307_LOGIC_OUTPUT 0 +#define SMA1307_HIGH_Z_OUTPUT BIT(3) + +#define SMA1307_DATA_CLK_SEL_MASK (BIT(6)|BIT(7)) +#define SMA1307_SDO_DATA 0 +#define SMA1307_SDO_CLK_PLL BIT(6) +#define SMA1307_SDO_CLK_OSC (BIT(6)|BIT(7)) + +/* SDO Output2 */ +#define SMA1307_SDO_OUTPUT2_MASK BIT(0) +#define SMA1307_ONE_SDO_PER_CH 0 +#define SMA1307_TWO_SDO_PER_CH BIT(0) + +/* SDO Output3 */ +#define SMA1307_SDO_OUTPUT3_MASK BIT(2) +#define SMA1307_SDO_OUTPUT3_DIS 0 +#define SMA1307_TWO_SDO_PER_CH_24K BIT(2) + +/* SDO OUT1 Select*/ +#define SMA1307_SDO_OUT1_SEL_MASK (BIT(3)|BIT(4)|BIT(5)) +#define SMA1307_SDO1_DISABLE 0 +#define SMA1307_SDO1_FORMAT_C BIT(3) +#define SMA1307_SDO1_MONO_MIX BIT(4) +#define SMA1307_SDO1_AFTER_DSP (BIT(3)|BIT(4)) +#define SMA1307_SDO1_VRMS2_AVG BIT(5) +#define SMA1307_SDO1_VBAT_MON (BIT(3)|BIT(5)) +#define SMA1307_SDO1_TEMP_MON (BIT(4)|BIT(5)) +#define SMA1307_SDO1_AFTER_DELAY (BIT(3)|BIT(4)|BIT(5)) + +/* SDO OUT0 Select*/ +#define SMA1307_SDO_OUT0_SEL_MASK (BIT(0)|BIT(1)|BIT(2)) +#define SMA1307_SDO0_DISABLE 0 +#define SMA1307_SDO0_FORMAT_C BIT(0) +#define SMA1307_SDO0_MONO_MIX BIT(1) +#define SMA1307_SDO0_AFTER_DSP (BIT(0)|BIT(1)) +#define SMA1307_SDO0_VRMS2_AVG BIT(2) +#define SMA1307_SDO0_VBAT_MON (BIT(0)|BIT(2)) +#define SMA1307_SDO0_TEMP_MON (BIT(1)|BIT(2)) +#define SMA1307_SDO0_AFTER_DELAY (BIT(0)|BIT(1)|BIT(2)) + +/* INTERRUPT Operation */ +#define SMA1307_SEL_INT_MASK BIT(2) +#define SMA1307_INT_CLEAR_AUTO 0 +#define SMA1307_INT_CLEAR_MANUAL BIT(2) + +/* INTERRUPT CLEAR */ +#define SMA1307_CLR_INT_MASK BIT(1) +#define SMA1307_INT_READY 0 +#define SMA1307_INT_CLEAR BIT(1) + +/* INTERRUPT Disable */ +#define SMA1307_DIS_INT_MASK BIT(0) +#define SMA1307_NORMAL_INT 0 +#define SMA1307_HIGH_Z_INT BIT(0) + +/* Interface Control */ +#define SMA1307_INTERFACE_MASK (BIT(5)|BIT(6)|BIT(7)) +#define SMA1307_LJ_FORMAT BIT(5) +#define SMA1307_I2S_FORMAT (BIT(5)|BIT(6)) +#define SMA1307_TDM_FORMAT BIT(7) + +#define SMA1307_SCK_RATE_MASK (BIT(3)|BIT(4)) +#define SMA1307_SCK_64FS 0 +#define SMA1307_SCK_32FS BIT(4) + +#define SMA1307_DATA_WIDTH_MASK (BIT(1)|BIT(2)) +#define SMA1307_DATA_24BIT 0 +#define SMA1307_DATA_16BIT (BIT(1)|BIT(2)) + +#define SMA1307_TDM_TX_MODE_MASK BIT(6) +#define SMA1307_TDM_TX_MONO 0 +#define SMA1307_TDM_TX_STEREO BIT(6) + +#define SMA1307_TDM_SLOT0_RX_POS_MASK (BIT(3)|BIT(4)|BIT(5)) +#define SMA1307_TDM_SLOT0_RX_POS_0 0 +#define SMA1307_TDM_SLOT0_RX_POS_1 BIT(3) +#define SMA1307_TDM_SLOT0_RX_POS_2 BIT(4) +#define SMA1307_TDM_SLOT0_RX_POS_3 (BIT(3)|BIT(4)) +#define SMA1307_TDM_SLOT0_RX_POS_4 BIT(5) +#define SMA1307_TDM_SLOT0_RX_POS_5 (BIT(3)|BIT(5)) +#define SMA1307_TDM_SLOT0_RX_POS_6 (BIT(4)|BIT(5)) +#define SMA1307_TDM_SLOT0_RX_POS_7 (BIT(3)|BIT(4)|BIT(5)) + +#define SMA1307_TDM_SLOT1_RX_POS_MASK (BIT(0)|BIT(1)|BIT(2)) +#define SMA1307_TDM_SLOT1_RX_POS_0 0 +#define SMA1307_TDM_SLOT1_RX_POS_1 BIT(0) +#define SMA1307_TDM_SLOT1_RX_POS_2 BIT(1) +#define SMA1307_TDM_SLOT1_RX_POS_3 (BIT(0)|BIT(1)) +#define SMA1307_TDM_SLOT1_RX_POS_4 BIT(2) +#define SMA1307_TDM_SLOT1_RX_POS_5 (BIT(0)|BIT(2)) +#define SMA1307_TDM_SLOT1_RX_POS_6 (BIT(1)|BIT(2)) +#define SMA1307_TDM_SLOT1_RX_POS_7 (BIT(0)|BIT(1)|BIT(2)) + +/* TDM2 FORMAT : 0xA6 */ +#define SMA1307_TDM_DL_MASK BIT(7) +#define SMA1307_TDM_DL_16 0 +#define SMA1307_TDM_DL_32 BIT(7) + +#define SMA1307_TDM_N_SLOT_MASK BIT(6) +#define SMA1307_TDM_N_SLOT_4 0 +#define SMA1307_TDM_N_SLOT_8 BIT(6) + +#define SMA1307_TDM_SLOT0_TX_POS_MASK (BIT(3)|BIT(4)|BIT(5)) +#define SMA1307_TDM_SLOT0_TX_POS_0 0 +#define SMA1307_TDM_SLOT0_TX_POS_1 BIT(3) +#define SMA1307_TDM_SLOT0_TX_POS_2 BIT(4) +#define SMA1307_TDM_SLOT0_TX_POS_3 (BIT(3)|BIT(4)) +#define SMA1307_TDM_SLOT0_TX_POS_4 BIT(5) +#define SMA1307_TDM_SLOT0_TX_POS_5 (BIT(3)|BIT(5)) +#define SMA1307_TDM_SLOT0_TX_POS_6 (BIT(4)|BIT(5)) +#define SMA1307_TDM_SLOT0_TX_POS_7 (BIT(3)|BIT(4)|BIT(5)) + +#define SMA1307_TDM_SLOT1_TX_POS_MASK (BIT(0)|BIT(1)|BIT(2)) +#define SMA1307_TDM_SLOT1_TX_POS_0 0 +#define SMA1307_TDM_SLOT1_TX_POS_1 BIT(0) +#define SMA1307_TDM_SLOT1_TX_POS_2 BIT(1) +#define SMA1307_TDM_SLOT1_TX_POS_3 (BIT(0)|BIT(1)) +#define SMA1307_TDM_SLOT1_TX_POS_4 BIT(2) +#define SMA1307_TDM_SLOT1_TX_POS_5 (BIT(0)|BIT(2)) +#define SMA1307_TDM_SLOT1_TX_POS_6 (BIT(1)|BIT(2)) +#define SMA1307_TDM_SLOT1_TX_POS_7 (BIT(0)|BIT(1)|BIT(2)) + +/* OTP STATUS */ +#define SMA1307_OTP_STAT_MASK BIT(6) +#define SMA1307_OTP_STAT_0 0 +#define SMA1307_OTP_STAT_1 BIT(6) + +/* STATUS */ +#define SMA1307_OT1_OK_STATUS BIT(7) +#define SMA1307_OT2_OK_STATUS BIT(6) +#define SMA1307_UVLO_STATUS BIT(5) +#define SMA1307_OVP_BST_STATUS BIT(4) +#define SMA1307_POWER_FLAG BIT(3) + +#define SMA1307_SCAN_CHK BIT(7) +#define SMA1307_OCP_SPK_STATUS BIT(5) +#define SMA1307_OCP_BST_STATUS BIT(4) +#define SMA1307_BOP_STATE (BIT(1)|BIT(2)|BIT(3)) +#define SMA1307_CLK_MON_STATUS BIT(0) + +#define SMA1307_DEVICE_ID (BIT(3)|BIT(4)) +#define SMA1307_REV_NUM_STATUS (BIT(0)|BIT(1)) +#define SMA1307_REV_NUM_REV0 0 +#define SMA1307_REV_NUM_REV1 BIT(0) + +#endif From 9b915776e0e6a2d185498077e0ebdb154a2751ac Mon Sep 17 00:00:00 2001 From: Fei Shao Date: Tue, 5 Nov 2024 17:14:31 +0800 Subject: [PATCH 59/97] ASoC: dt-bindings: maxim,max98390: Reference common DAI properties MAX98390 is a smart amplifier and exposes one DAI, so '#sound-dai-cells' property is needed for describing the DAI links. Reference the dai-common.yaml schema to allow '#sound-dai-cells' to be used. This fixes dtbs_check error: '#sound-dai-cells' does not match any of the regexes: 'pinctrl-[0-9]+' Signed-off-by: Fei Shao Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20241105091513.3963102-1-fshao@chromium.org Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/maxim,max98390.yaml | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/maxim,max98390.yaml b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml index deaa6886c42fcf..d35dd8408c61cd 100644 --- a/Documentation/devicetree/bindings/sound/maxim,max98390.yaml +++ b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml @@ -9,6 +9,9 @@ title: Maxim Integrated MAX98390 Speaker Amplifier with Integrated Dynamic Speak maintainers: - Steve Lee +allOf: + - $ref: dai-common.yaml# + properties: compatible: const: maxim,max98390 @@ -32,11 +35,14 @@ properties: reset-gpios: maxItems: 1 + '#sound-dai-cells': + const: 0 + required: - compatible - reg -additionalProperties: false +unevaluatedProperties: false examples: - | From de156f3cf70e17dc6ff4c3c364bb97a6db961ffd Mon Sep 17 00:00:00 2001 From: Mingcong Bai Date: Wed, 6 Nov 2024 10:40:50 +0800 Subject: [PATCH 60/97] ASoC: amd: yc: fix internal mic on Xiaomi Book Pro 14 2022 Xiaomi Book Pro 14 2022 (MIA2210-AD) requires a quirk entry for its internal microphone to be enabled. This is likely due to similar reasons as seen previously on Redmi Book 14/15 Pro 2022 models (since they likely came with similar firmware): - commit dcff8b7ca92d ("ASoC: amd: yc: Add Xiaomi Redmi Book Pro 15 2022 into DMI table") - commit c1dd6bf61997 ("ASoC: amd: yc: Add Xiaomi Redmi Book Pro 14 2022 into DMI table") A quirk would likely be needed for Xiaomi Book Pro 15 2022 models, too. However, I do not have such device on hand so I will leave it for now. Signed-off-by: Mingcong Bai Link: https://patch.msgid.link/20241106024052.15748-1-jeffbai@aosc.io Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 438865d5e376e0..dc476bfb6da40f 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -395,6 +395,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Redmi Book Pro 15 2022"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "TIMI"), + DMI_MATCH(DMI_PRODUCT_NAME, "Xiaomi Book Pro 14 2022"), + } + }, { .driver_data = &acp6x_card, .matches = { From 94debe5eaa0adaa24a6de4a8e5f138be7381eb9e Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Wed, 6 Nov 2024 19:56:57 +0530 Subject: [PATCH 61/97] ASoC: SOF: amd: Fix for incorrect DMA ch status register offset DMA ch status register offset change in acp7.0 platform Incorrect DMA channel status register offset check lead to firmware boot failure. [ 14.432497] snd_sof_amd_acp70 0000:c4:00.5: ------------[ DSP dump start ]------------ [ 14.432533] snd_sof_amd_acp70 0000:c4:00.5: Firmware boot failure due to timeout [ 14.432549] snd_sof_amd_acp70 0000:c4:00.5: fw_state: SOF_FW_BOOT_IN_PROGRESS (3) [ 14.432610] snd_sof_amd_acp70 0000:c4:00.5: invalid header size 0x71c41000. FW oops is bogus [ 14.432626] snd_sof_amd_acp70 0000:c4:00.5: unexpected fault 0x71c40000 trace 0x71c40000 [ 14.432642] snd_sof_amd_acp70 0000:c4:00.5: ------------[ DSP dump end ]------------ [ 14.432657] snd_sof_amd_acp70 0000:c4:00.5: error: failed to boot DSP firmware -5 [ 14.432672] snd_sof_amd_acp70 0000:c4:00.5: fw_state change: 3 -> 4 [ 14.433260] dmic-codec dmic-codec: ASoC: Unregistered DAI 'dmic-hifi' [ 14.433319] snd_sof_amd_acp70 0000:c4:00.5: fw_state change: 4 -> 0 [ 14.433358] snd_sof_amd_acp70 0000:c4:00.5: error: sof_probe_work failed err: -5 Update correct register offset for DMA ch status register. Fixes: 490be7ba2a01 ("ASoC: SOF: amd: add support for acp7.0 based platform") Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20241106142658.1240929-1-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index de3001f5b9bb7c..95d4762c9d9390 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -342,11 +342,19 @@ int acp_dma_status(struct acp_dev_data *adata, unsigned char ch) { struct snd_sof_dev *sdev = adata->dev; unsigned int val; + unsigned int acp_dma_ch_sts; int ret = 0; + switch (adata->pci_rev) { + case ACP70_PCI_ID: + acp_dma_ch_sts = ACP70_DMA_CH_STS; + break; + default: + acp_dma_ch_sts = ACP_DMA_CH_STS; + } val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_DMA_CNTL_0 + ch * sizeof(u32)); if (val & ACP_DMA_CH_RUN) { - ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_DMA_CH_STS, val, !val, + ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, acp_dma_ch_sts, val, !val, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); if (ret < 0) From 9d4f9f6a7bb1afbde57d08c98f2db4ff019ee19d Mon Sep 17 00:00:00 2001 From: Claudiu Beznea Date: Wed, 6 Nov 2024 10:18:17 +0200 Subject: [PATCH 62/97] ASoC: da7213: Populate max_register to regmap_config On the Renesas RZ/G3S SMARC Carrier II board having a DA7212 codec (using da7213 driver) connected to one SSIF-2 available on the Renesas RZ/G3S SoC it has been discovered that using the runtime PM API for suspend/resume (as will be proposed in the following commits) leads to the codec not being propertly initialized after resume. This is because w/o max_register populated to regmap_config the regcache_rbtree_sync() breaks on base_reg > max condition and the regcache_sync_block() call is skipped. Fixes: ef5c2eba2412 ("ASoC: codecs: Add da7213 codec") Cc: stable@vger.kernel.org Signed-off-by: Claudiu Beznea Link: https://patch.msgid.link/20241106081826.1211088-23-claudiu.beznea.uj@bp.renesas.com Signed-off-by: Mark Brown --- sound/soc/codecs/da7213.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index f3ef6fb5530471..486db60bf2dd14 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -2136,6 +2136,7 @@ static const struct regmap_config da7213_regmap_config = { .reg_bits = 8, .val_bits = 8, + .max_register = DA7213_TONE_GEN_OFF_PER, .reg_defaults = da7213_reg_defaults, .num_reg_defaults = ARRAY_SIZE(da7213_reg_defaults), .volatile_reg = da7213_volatile_register, From 841256954037ad80a57b8fa17a794ae9a01b2e23 Mon Sep 17 00:00:00 2001 From: Claudiu Beznea Date: Wed, 6 Nov 2024 10:18:18 +0200 Subject: [PATCH 63/97] ASoC: da7213: Return directly the value of regcache_sync() Return directly the value of the regcache_sync() in da7213_runtime_resume(). In case of any failures this will inform the runtime resume process. Signed-off-by: Claudiu Beznea Link: https://patch.msgid.link/20241106081826.1211088-24-claudiu.beznea.uj@bp.renesas.com Signed-off-by: Mark Brown --- sound/soc/codecs/da7213.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 486db60bf2dd14..4298ca77fa30fa 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -2230,8 +2230,7 @@ static int __maybe_unused da7213_runtime_resume(struct device *dev) if (ret < 0) return ret; regcache_cache_only(da7213->regmap, false); - regcache_sync(da7213->regmap); - return 0; + return regcache_sync(da7213->regmap); } static const struct dev_pm_ops da7213_pm = { From 431e040065c814448ffcc2fac493f7dbbfb2e796 Mon Sep 17 00:00:00 2001 From: Claudiu Beznea Date: Wed, 6 Nov 2024 10:18:19 +0200 Subject: [PATCH 64/97] ASoC: da7213: Add suspend to RAM support Add suspend to RAM support. This uses the already available runtime PM support. Signed-off-by: Claudiu Beznea Link: https://patch.msgid.link/20241106081826.1211088-25-claudiu.beznea.uj@bp.renesas.com Signed-off-by: Mark Brown --- sound/soc/codecs/da7213.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 4298ca77fa30fa..01c78f8032c4d2 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -2235,6 +2235,7 @@ static int __maybe_unused da7213_runtime_resume(struct device *dev) static const struct dev_pm_ops da7213_pm = { SET_RUNTIME_PM_OPS(da7213_runtime_suspend, da7213_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) }; static const struct i2c_device_id da7213_i2c_id[] = { From 1e1a2ef95b571825ca9c0113f6bef51e9cec98b0 Mon Sep 17 00:00:00 2001 From: Hao Bui Date: Wed, 6 Nov 2024 10:18:20 +0200 Subject: [PATCH 65/97] ASoC: da7213: Avoid setting PLL when closing audio stream When audio stream is closing, audio frequency is set to 0 by ALSA but codec driver DA7213 does not handle properly in this case. This patch adds checking of 0Hz frequency to da7213_set_component_sysclk() and avoid unnecessary PLL settings. Signed-off-by: Hao Bui Signed-off-by: Claudiu Beznea Link: https://patch.msgid.link/20241106081826.1211088-26-claudiu.beznea.uj@bp.renesas.com Signed-off-by: Mark Brown --- sound/soc/codecs/da7213.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 01c78f8032c4d2..af38b2b5e1744a 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1555,6 +1555,10 @@ static int da7213_set_component_sysclk(struct snd_soc_component *component, if ((da7213->clk_src == clk_id) && (da7213->mclk_rate == freq)) return 0; + /* Maybe audio stream is closing. */ + if (freq == 0) + return 0; + if (((freq < 5000000) && (freq != 32768)) || (freq > 54000000)) { dev_err(component->dev, "Unsupported MCLK value %d\n", freq); From b3296f9095d6ad24723e5ad89c28acc317d0c3cf Mon Sep 17 00:00:00 2001 From: Hao Bui Date: Wed, 6 Nov 2024 10:18:21 +0200 Subject: [PATCH 66/97] ASoC: da7213: Extend support for the MCK in range [2, 50] MHz According to DA7212 HW manual, the codec's PLL input divider can operate with MCLK range from 2MHz to 50MHz but current driver only set the minimum supported MCLK frequency to 5MHz. That cause 11.025kHz audio which is corresponding to MCLK of 2.8224MHz (11.025kHz * 256) unable to play in case audio-simple-card is used. Signed-off-by: Hao Bui Co-developed-by: Claudiu Beznea Signed-off-by: Claudiu Beznea Link: https://patch.msgid.link/20241106081826.1211088-27-claudiu.beznea.uj@bp.renesas.com Signed-off-by: Mark Brown --- sound/soc/codecs/da7213.c | 18 +++++++++++++----- sound/soc/codecs/da7213.h | 1 + 2 files changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index af38b2b5e1744a..ca4cc954efa8e6 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include @@ -1559,7 +1560,7 @@ static int da7213_set_component_sysclk(struct snd_soc_component *component, if (freq == 0) return 0; - if (((freq < 5000000) && (freq != 32768)) || (freq > 54000000)) { + if (((freq < da7213->fin_min_rate) && (freq != 32768)) || (freq > 54000000)) { dev_err(component->dev, "Unsupported MCLK value %d\n", freq); return -EINVAL; @@ -1858,11 +1859,14 @@ static int da7213_set_bias_level(struct snd_soc_component *component, return 0; } +#define DA7213_FIN_MIN_RATE (5 * MEGA) +#define DA7212_FIN_MIN_RATE (2 * MEGA) + #if defined(CONFIG_OF) /* DT */ static const struct of_device_id da7213_of_match[] = { - { .compatible = "dlg,da7212", }, - { .compatible = "dlg,da7213", }, + { .compatible = "dlg,da7212", .data = (void *)DA7212_FIN_MIN_RATE }, + { .compatible = "dlg,da7213", .data = (void *)DA7213_FIN_MIN_RATE }, { } }; MODULE_DEVICE_TABLE(of, da7213_of_match); @@ -1870,8 +1874,8 @@ MODULE_DEVICE_TABLE(of, da7213_of_match); #ifdef CONFIG_ACPI static const struct acpi_device_id da7213_acpi_match[] = { - { "DLGS7212", 0}, - { "DLGS7213", 0}, + { "DLGS7212", DA7212_FIN_MIN_RATE }, + { "DLGS7213", DA7213_FIN_MIN_RATE }, { }, }; MODULE_DEVICE_TABLE(acpi, da7213_acpi_match); @@ -2167,6 +2171,10 @@ static int da7213_i2c_probe(struct i2c_client *i2c) if (!da7213) return -ENOMEM; + da7213->fin_min_rate = (uintptr_t)i2c_get_match_data(i2c); + if (!da7213->fin_min_rate) + return -EINVAL; + i2c_set_clientdata(i2c, da7213); /* Get required supplies */ diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 505b731c0adb93..b9ab791d6b883b 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -600,6 +600,7 @@ struct da7213_priv { struct clk *mclk; unsigned int mclk_rate; unsigned int out_rate; + unsigned int fin_min_rate; int clk_src; bool master; bool alc_calib_auto; From 7f4eb7672b1785119b29a4dff50aeef13368e813 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 7 Nov 2024 14:03:05 +0200 Subject: [PATCH 67/97] ASoC: SOF: ext_manifest: Add missing ext_manifest type for PROBE_INFO Elem type 3 is PROBE_INFO in ext_manifest, add it to the list. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Liam Girdwood Link: https://patch.msgid.link/20241107120306.30680-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/ext_manifest.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/sof/ext_manifest.h b/include/sound/sof/ext_manifest.h index fc0231d04a941c..7dfe3ccf1fe4ec 100644 --- a/include/sound/sof/ext_manifest.h +++ b/include/sound/sof/ext_manifest.h @@ -60,6 +60,7 @@ enum sof_ext_man_elem_type { SOF_EXT_MAN_ELEM_FW_VERSION = 0, SOF_EXT_MAN_ELEM_WINDOW = 1, SOF_EXT_MAN_ELEM_CC_VERSION = 2, + SOF_EXT_MAN_ELEM_PROBE_INFO = 3, SOF_EXT_MAN_ELEM_DBG_ABI = 4, SOF_EXT_MAN_ELEM_CONFIG_DATA = 5, /**< ABI3.17 */ SOF_EXT_MAN_ELEM_PLATFORM_CONFIG_DATA = 6, From 83e367c1a178045bf3675646aa9582a0c5a4e566 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 7 Nov 2024 14:03:06 +0200 Subject: [PATCH 68/97] ASoC: SOF: ipc3-loader: 'Handle' PROBE_INFO ext_manifest type when parsing Every time when the firmware is loaded we see the following printed in info level: unknown sof_ext_man header type 3 size 0x30 This is the PROBE_INFO element, which is not parsed, but it is not something that we should treat as surprise, aka unknown type. Add an empty case with a debug print to 'handle' this ext_manifest type and silence the confusing print in kernel log. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Liam Girdwood Link: https://patch.msgid.link/20241107120306.30680-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-loader.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sof/ipc3-loader.c b/sound/soc/sof/ipc3-loader.c index 35b89c2b9d4c02..7e9c76d5b2c91c 100644 --- a/sound/soc/sof/ipc3-loader.c +++ b/sound/soc/sof/ipc3-loader.c @@ -193,6 +193,9 @@ static size_t sof_ipc3_fw_parse_ext_man(struct snd_sof_dev *sdev) case SOF_EXT_MAN_ELEM_CC_VERSION: ret = ipc3_fw_ext_man_get_cc_info(sdev, elem_hdr); break; + case SOF_EXT_MAN_ELEM_PROBE_INFO: + dev_dbg(sdev->dev, "Probe info (not parsed)\n"); + break; case SOF_EXT_MAN_ELEM_DBG_ABI: ret = ipc3_fw_ext_man_get_dbg_abi_info(sdev, elem_hdr); break; From 1b1f491dac4f6ee88ec7c36c4fd27880b0a89f41 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 7 Nov 2024 14:15:32 +0200 Subject: [PATCH 69/97] ASoC: SOF: Intel: hda-stream: Always use at least two BDLE for transfers The HDA specification states that the SDnLVI (Last Valid Index) must be at least 1 (two BDLE entry). Update the debug prints as well to provide better information. While the LVI=0 worked so far without issues, it is better to align with the specification to avoid unforeseen issues with FW loading. Notes: - The LVI > 0 rules is valid and honored for audio use cases - LVI == 0 is used with software controlled (SPIB) transfers only for firmware and library loading, which is permitted by the hardware - This is not spelled out in the specification and it is better to avoid cases Signed-off-by: Peter Ujfalusi Reviewed-by: Liam Girdwood Reviewed-by: Kai Vehmanen Reviewed-by: Bard Liao Link: https://patch.msgid.link/20241107121532.3241-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-stream.c | 32 +++++++++++++++++++++++++++++--- 1 file changed, 29 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 3ac63ce67ab1ce..519bafd3b947bb 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -119,13 +119,39 @@ int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, int remain, ioc; period_bytes = hstream->period_bytes; - dev_dbg(sdev->dev, "period_bytes:0x%x\n", period_bytes); - if (!period_bytes) + dev_dbg(sdev->dev, "period_bytes: %#x, bufsize: %#x\n", period_bytes, + hstream->bufsize); + + if (!period_bytes) { + unsigned int chunk_size; + + chunk_size = snd_sgbuf_get_chunk_size(dmab, 0, hstream->bufsize); + period_bytes = hstream->bufsize; + /* + * HDA spec demands that the LVI value must be at least one + * before the DMA operation can begin. This means that there + * must be at least two BDLE present for the transfer. + * + * If the buffer is not a single continuous area then the + * hda_setup_bdle() will create multiple BDLEs for each segment. + * If the memory is a single continuous area, force it to be + * split into two 'periods', otherwise the transfer will be + * split to multiple BDLE for each chunk in hda_setup_bdle() + * + * Note: period_bytes == 0 can only happen for firmware or + * library loading. The data size is 4K aligned, which ensures + * that the second chunk's start address will be 128-byte + * aligned. + */ + if (chunk_size == hstream->bufsize) + period_bytes /= 2; + } + periods = hstream->bufsize / period_bytes; - dev_dbg(sdev->dev, "periods:%d\n", periods); + dev_dbg(sdev->dev, "periods: %d\n", periods); remain = hstream->bufsize % period_bytes; if (remain) From 1862e847bf115a3ccbf38dd035ea0118be57f2e2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 7 Nov 2024 14:14:40 +0200 Subject: [PATCH 70/97] ASoC: SOF: Intel: hda: Add support for persistent Code Loader DMA buffers It has been reported that the DMA memory allocation for firmware download can fail after extended period of uptime on systems with relatively small amount of RAM when the system memory becomes fragmented. The issue primarily happens when the system is waking up from system suspend, swap might not be available and the MM system cannot move things around to allow for successful allocation. If the IMR boot is not supported then for each DSP boot we would need to allocate the DMA buffer for firmware transfer, which can increase the chances of the issue to be hit. This patch adds support for allocating the DMA buffers once at first boot time and keep it until the system is shut down, rebooted or the drivers re-loaded and makes this as the default operation. With persistent_cl_buffer module parameter the persistent Code Loader DMA buffer can be disabled to fall back to on demand allocation. Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Liam Girdwood Reviewed-by: Kai Vehmanen Link: https://patch.msgid.link/20241107121440.1472-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 100 ++++++++++++++++++++++--------- sound/soc/sof/intel/hda.c | 6 ++ sound/soc/sof/intel/hda.h | 14 ++++- 3 files changed, 90 insertions(+), 30 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 9d8ebb7c6a1060..76a03b6b27286a 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -26,6 +26,11 @@ #include "../sof-priv.h" #include "hda.h" +static bool persistent_cl_buffer = true; +module_param(persistent_cl_buffer, bool, 0444); +MODULE_PARM_DESC(persistent_cl_buffer, "Persistent Code Loader DMA buffer " + "(default = Y, use N to force buffer re-allocation)"); + static void hda_ssp_set_cbp_cfp(struct snd_sof_dev *sdev) { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; @@ -43,9 +48,10 @@ static void hda_ssp_set_cbp_cfp(struct snd_sof_dev *sdev) } } -struct hdac_ext_stream *hda_cl_prepare(struct device *dev, unsigned int format, - unsigned int size, struct snd_dma_buffer *dmab, - int direction, bool is_iccmax) +struct hdac_ext_stream* +hda_cl_prepare(struct device *dev, unsigned int format, unsigned int size, + struct snd_dma_buffer *dmab, bool persistent_buffer, int direction, + bool is_iccmax) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); struct hdac_ext_stream *hext_stream; @@ -61,11 +67,19 @@ struct hdac_ext_stream *hda_cl_prepare(struct device *dev, unsigned int format, hstream = &hext_stream->hstream; hstream->substream = NULL; - /* allocate DMA buffer */ - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, dev, size, dmab); - if (ret < 0) { - dev_err(sdev->dev, "error: memory alloc failed: %d\n", ret); - goto out_put; + /* + * Allocate DMA buffer if it is temporary or if the buffer is intended + * to be persistent but not yet allocated. + * We cannot rely solely on !dmab->area as caller might use a struct on + * stack (when it is temporary) without clearing it to 0. + */ + if (!persistent_buffer || !dmab->area) { + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, dev, size, dmab); + if (ret < 0) { + dev_err(sdev->dev, "%s: memory alloc failed: %d\n", + __func__, ret); + goto out_put; + } } hstream->period_bytes = 0;/* initialize period_bytes */ @@ -91,6 +105,10 @@ struct hdac_ext_stream *hda_cl_prepare(struct device *dev, unsigned int format, out_free: snd_dma_free_pages(dmab); + dmab->area = NULL; + dmab->bytes = 0; + hstream->bufsize = 0; + hstream->format_val = 0; out_put: hda_dsp_stream_put(sdev, direction, hstream->stream_tag); return ERR_PTR(ret); @@ -255,7 +273,7 @@ int hda_cl_trigger(struct device *dev, struct hdac_ext_stream *hext_stream, int EXPORT_SYMBOL_NS(hda_cl_trigger, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_cl_cleanup(struct device *dev, struct snd_dma_buffer *dmab, - struct hdac_ext_stream *hext_stream) + bool persistent_buffer, struct hdac_ext_stream *hext_stream) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); struct hdac_stream *hstream = &hext_stream->hstream; @@ -279,10 +297,14 @@ int hda_cl_cleanup(struct device *dev, struct snd_dma_buffer *dmab, sd_offset + SOF_HDA_ADSP_REG_SD_BDLPU, 0); snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, sd_offset, 0); - snd_dma_free_pages(dmab); - dmab->area = NULL; - hstream->bufsize = 0; - hstream->format_val = 0; + + if (!persistent_buffer) { + snd_dma_free_pages(dmab); + dmab->area = NULL; + dmab->bytes = 0; + hstream->bufsize = 0; + hstream->format_val = 0; + } return ret; } @@ -340,8 +362,8 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev) { + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; struct hdac_ext_stream *iccmax_stream; - struct snd_dma_buffer dmab_bdl; int ret, ret1; u8 original_gb; @@ -354,7 +376,8 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev) * the data, so use a buffer of PAGE_SIZE for receiving. */ iccmax_stream = hda_cl_prepare(sdev->dev, HDA_CL_STREAM_FORMAT, PAGE_SIZE, - &dmab_bdl, SNDRV_PCM_STREAM_CAPTURE, true); + &hda->iccmax_dmab, persistent_cl_buffer, + SNDRV_PCM_STREAM_CAPTURE, true); if (IS_ERR(iccmax_stream)) { dev_err(sdev->dev, "error: dma prepare for ICCMAX stream failed\n"); return PTR_ERR(iccmax_stream); @@ -366,7 +389,8 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev) * Perform iccmax stream cleanup. This should be done even if firmware loading fails. * If the cleanup also fails, we return the initial error */ - ret1 = hda_cl_cleanup(sdev->dev, &dmab_bdl, iccmax_stream); + ret1 = hda_cl_cleanup(sdev->dev, &hda->iccmax_dmab, + persistent_cl_buffer, iccmax_stream); if (ret1 < 0) { dev_err(sdev->dev, "error: ICCMAX stream cleanup failed\n"); @@ -408,7 +432,6 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) const struct sof_intel_dsp_desc *chip_info; struct hdac_ext_stream *hext_stream; struct firmware stripped_firmware; - struct snd_dma_buffer dmab; int ret, ret1, i; if (hda->imrboot_supported && !sdev->first_boot && !hda->skip_imr_boot) { @@ -432,23 +455,31 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) return -EINVAL; } - stripped_firmware.data = sdev->basefw.fw->data + sdev->basefw.payload_offset; - stripped_firmware.size = sdev->basefw.fw->size - sdev->basefw.payload_offset; - /* init for booting wait */ init_waitqueue_head(&sdev->boot_wait); /* prepare DMA for code loader stream */ + stripped_firmware.size = sdev->basefw.fw->size - sdev->basefw.payload_offset; hext_stream = hda_cl_prepare(sdev->dev, HDA_CL_STREAM_FORMAT, stripped_firmware.size, - &dmab, SNDRV_PCM_STREAM_PLAYBACK, false); + &hda->cl_dmab, persistent_cl_buffer, + SNDRV_PCM_STREAM_PLAYBACK, false); if (IS_ERR(hext_stream)) { dev_err(sdev->dev, "error: dma prepare for fw loading failed\n"); return PTR_ERR(hext_stream); } - memcpy(dmab.area, stripped_firmware.data, - stripped_firmware.size); + /* + * Copy the payload to the DMA buffer if it is temporary or if the + * buffer is persistent but it does not have the basefw payload either + * because this is the first boot and the buffer needs to be initialized, + * or a library got loaded and it replaced the basefw. + */ + if (!persistent_cl_buffer || !hda->cl_dmab_contains_basefw) { + stripped_firmware.data = sdev->basefw.fw->data + sdev->basefw.payload_offset; + memcpy(hda->cl_dmab.area, stripped_firmware.data, stripped_firmware.size); + hda->cl_dmab_contains_basefw = true; + } /* try ROM init a few times before giving up */ for (i = 0; i < HDA_FW_BOOT_ATTEMPTS; i++) { @@ -514,7 +545,8 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) * This should be done even if firmware loading fails. * If the cleanup also fails, we return the initial error */ - ret1 = hda_cl_cleanup(sdev->dev, &dmab, hext_stream); + ret1 = hda_cl_cleanup(sdev->dev, &hda->cl_dmab, + persistent_cl_buffer, hext_stream); if (ret1 < 0) { dev_err(sdev->dev, "error: Code loader DSP cleanup failed\n"); @@ -545,7 +577,6 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream; struct firmware stripped_firmware; struct sof_ipc4_msg msg = {}; - struct snd_dma_buffer dmab; int ret, ret1; /* if IMR booting is enabled and fw context is saved for D3 state, skip the loading */ @@ -556,16 +587,28 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev, stripped_firmware.data = fw_lib->sof_fw.fw->data + fw_lib->sof_fw.payload_offset; stripped_firmware.size = fw_lib->sof_fw.fw->size - fw_lib->sof_fw.payload_offset; + /* + * force re-allocation of the cl_dmab if the preserved DMA buffer is + * smaller than what is needed for the library + */ + if (persistent_cl_buffer && stripped_firmware.size > hda->cl_dmab.bytes) { + snd_dma_free_pages(&hda->cl_dmab); + hda->cl_dmab.area = NULL; + hda->cl_dmab.bytes = 0; + } + /* prepare DMA for code loader stream */ hext_stream = hda_cl_prepare(sdev->dev, HDA_CL_STREAM_FORMAT, stripped_firmware.size, - &dmab, SNDRV_PCM_STREAM_PLAYBACK, false); + &hda->cl_dmab, persistent_cl_buffer, + SNDRV_PCM_STREAM_PLAYBACK, false); if (IS_ERR(hext_stream)) { dev_err(sdev->dev, "%s: DMA prepare failed\n", __func__); return PTR_ERR(hext_stream); } - memcpy(dmab.area, stripped_firmware.data, stripped_firmware.size); + memcpy(hda->cl_dmab.area, stripped_firmware.data, stripped_firmware.size); + hda->cl_dmab_contains_basefw = false; /* * 1st stage: SOF_IPC4_GLB_LOAD_LIBRARY_PREPARE @@ -628,7 +671,8 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev, cleanup: /* clean up even in case of error and return the first error */ - ret1 = hda_cl_cleanup(sdev->dev, &dmab, hext_stream); + ret1 = hda_cl_cleanup(sdev->dev, &hda->cl_dmab, persistent_cl_buffer, + hext_stream); if (ret1 < 0) { dev_err(sdev->dev, "%s: Code loader DSP cleanup failed\n", __func__); diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 38921c0db84eed..01b135068b1f80 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -936,6 +936,12 @@ void hda_dsp_remove(struct snd_sof_dev *sdev) /* disable DSP */ hda_dsp_ctrl_ppcap_enable(sdev, false); + /* Free the persistent DMA buffers used for base firmware download */ + if (hda->cl_dmab.area) + snd_dma_free_pages(&hda->cl_dmab); + if (hda->iccmax_dmab.area) + snd_dma_free_pages(&hda->iccmax_dmab); + skip_disable_dsp: free_irq(sdev->ipc_irq, sdev); if (sdev->msi_enabled) diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index b74a472435b5d2..22bd9c3c8216c5 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -495,6 +495,15 @@ struct sof_intel_hda_dev { int boot_iteration; + /* + * DMA buffers for base firmware download. By default the buffers are + * allocated once and kept through the lifetime of the driver. + * See module parameter: persistent_cl_buffer + */ + struct snd_dma_buffer cl_dmab; + bool cl_dmab_contains_basefw; + struct snd_dma_buffer iccmax_dmab; + struct hda_bus hbus; /* hw config */ @@ -714,11 +723,12 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream struct hdac_ext_stream *hda_cl_prepare(struct device *dev, unsigned int format, unsigned int size, struct snd_dma_buffer *dmab, - int direction, bool is_iccmax); + bool persistent_buffer, int direction, + bool is_iccmax); int hda_cl_trigger(struct device *dev, struct hdac_ext_stream *hext_stream, int cmd); int hda_cl_cleanup(struct device *dev, struct snd_dma_buffer *dmab, - struct hdac_ext_stream *hext_stream); + bool persistent_buffer, struct hdac_ext_stream *hext_stream); int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot); #define HDA_CL_STREAM_FORMAT 0x40 From 8c21e40e1e481f7fef6e570089e317068b972c45 Mon Sep 17 00:00:00 2001 From: Markus Petri Date: Thu, 7 Nov 2024 10:40:20 +0100 Subject: [PATCH 71/97] ASoC: amd: yc: Support dmic on another model of Lenovo Thinkpad E14 Gen 6 Another model of Thinkpad E14 Gen 6 (21M4) needs a quirk entry for the dmic to be detected. Signed-off-by: Markus Petri Link: https://patch.msgid.link/20241107094020.1050935-1-mp@localhost Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index dc476bfb6da40f..2436e8deb2be48 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -227,6 +227,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21M3"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21M4"), + } + }, { .driver_data = &acp6x_card, .matches = { From 63c1c87993e0e5bb11bced3d8224446a2bc62338 Mon Sep 17 00:00:00 2001 From: Luo Yifan Date: Wed, 6 Nov 2024 09:46:54 +0800 Subject: [PATCH 72/97] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate() This patch checks if div is less than or equal to zero (div <= 0). If div is zero or negative, the function returns -EINVAL, ensuring the division operation (*prate / div) is safe to perform. Signed-off-by: Luo Yifan Link: https://patch.msgid.link/20241106014654.206860-1-luoyifan@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 7bc4a96b7503f3..2570daa3e225c9 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -378,8 +378,8 @@ static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, int div; div = stm32_sai_get_clk_div(sai, *prate, rate); - if (div < 0) - return div; + if (div <= 0) + return -EINVAL; mclk->freq = *prate / div; From 23569c8b314925bdb70dd1a7b63cfe6100868315 Mon Sep 17 00:00:00 2001 From: Luo Yifan Date: Thu, 7 Nov 2024 09:59:36 +0800 Subject: [PATCH 73/97] ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div() This patch checks if div is less than or equal to zero (div <= 0). If div is zero or negative, the function returns -EINVAL, ensuring the division operation is safe to perform. Signed-off-by: Luo Yifan Reviewed-by: Olivier Moysan Link: https://patch.msgid.link/20241107015936.211902-1-luoyifan@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 2570daa3e225c9..5828f9dd866e4d 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -317,7 +317,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, int div; div = DIV_ROUND_CLOSEST(input_rate, output_rate); - if (div > SAI_XCR1_MCKDIV_MAX(version)) { + if (div > SAI_XCR1_MCKDIV_MAX(version) || div <= 0) { dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); return -EINVAL; } From 7a117225b15b2d1dc021ab9fc36687c1e61ad2b1 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Thu, 7 Nov 2024 15:49:57 +0200 Subject: [PATCH 74/97] ASoC: SOF: Intel: hda: handle only paused streams in hda_dai_suspend() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit hda_dai_suspend() was added to handle paused stream during system suspend. But as a side effect, it also ends up cleaning up the DMA data for those streams that were prepared but not triggered before a system suspend. Since these streams will not receive the prepare callback after resuming, we need to preserve the DMA data during suspend. So, add the check to handle only those streams that are in the paused state to avoid losing the DMA data for all other streams. Link: https://github.com/thesofproject/linux/issues/5080 Signed-off-by: Ranjani Sridharan Reviewed-by: Fred Oh Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://patch.msgid.link/20241107134957.25160-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 1c823f9eea5700..f1a8491bba9ebf 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -617,6 +617,10 @@ static int hda_dai_suspend(struct hdac_bus *bus) sdai = swidget->private; ops = sdai->platform_private; + if (rtd->dpcm[hext_stream->link_substream->stream].state != + SND_SOC_DPCM_STATE_PAUSED) + continue; + /* for consistency with TRIGGER_SUSPEND */ if (ops->post_trigger) { ret = ops->post_trigger(sdev, cpu_dai, From fa24fdc8ae9e6f6fe7a4f7676a4d8c14433a86c0 Mon Sep 17 00:00:00 2001 From: Tang Bin Date: Thu, 7 Nov 2024 15:54:40 +0800 Subject: [PATCH 75/97] ASoC: ux500: Remove redundant casts In the function ux500_msp_drv_probe, the 'int' type cast in front of the PTR_ERR() macro is redundant, thus remove it. Signed-off-by: Tang Bin Link: https://patch.msgid.link/20241107075440.2770-1-tangbin@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_dai.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index a2dd739fdf2df3..7798957c6504d5 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -733,7 +733,7 @@ static int ux500_msp_drv_probe(struct platform_device *pdev) drvdata->reg_vape = devm_regulator_get(&pdev->dev, "v-ape"); if (IS_ERR(drvdata->reg_vape)) { - ret = (int)PTR_ERR(drvdata->reg_vape); + ret = PTR_ERR(drvdata->reg_vape); dev_err(&pdev->dev, "%s: ERROR: Failed to get Vape supply (%d)!\n", __func__, ret); @@ -743,7 +743,7 @@ static int ux500_msp_drv_probe(struct platform_device *pdev) drvdata->pclk = devm_clk_get(&pdev->dev, "apb_pclk"); if (IS_ERR(drvdata->pclk)) { - ret = (int)PTR_ERR(drvdata->pclk); + ret = PTR_ERR(drvdata->pclk); dev_err(&pdev->dev, "%s: ERROR: devm_clk_get of pclk failed (%d)!\n", __func__, ret); @@ -752,7 +752,7 @@ static int ux500_msp_drv_probe(struct platform_device *pdev) drvdata->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(drvdata->clk)) { - ret = (int)PTR_ERR(drvdata->clk); + ret = PTR_ERR(drvdata->clk); dev_err(&pdev->dev, "%s: ERROR: devm_clk_get failed (%d)!\n", __func__, ret); From 48b86532c10128cf50c854a90c2d5b1410f4012d Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Thu, 7 Nov 2024 15:28:40 +0200 Subject: [PATCH 76/97] ASoC: SOF: sof-client-probes-ipc4: Set param_size extension bits Write the size of the optional payload of SOF_IPC4_MOD_INIT_INSTANCE message to extension param_size-bits. The previous IPC4 version does not set these bits that should indicate the size of the optional payload (struct sof_ipc4_probe_cfg). The old firmware side component code works well without these bits, but when the probes are converted to use the generic module API, this does not work anymore. Fixes: f5623593060f ("ASoC: SOF: IPC4: probes: Implement IPC4 ops for probes client device") Signed-off-by: Jyri Sarha Reviewed-by: Ranjani Sridharan Reviewed-by: Liam Girdwood Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://patch.msgid.link/20241107132840.17386-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-client-probes-ipc4.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/sof-client-probes-ipc4.c b/sound/soc/sof/sof-client-probes-ipc4.c index 796eac0a2e74f7..603aed222480ff 100644 --- a/sound/soc/sof/sof-client-probes-ipc4.c +++ b/sound/soc/sof/sof-client-probes-ipc4.c @@ -125,6 +125,7 @@ static int ipc4_probes_init(struct sof_client_dev *cdev, u32 stream_tag, msg.primary |= SOF_IPC4_MSG_TARGET(SOF_IPC4_MODULE_MSG); msg.extension = SOF_IPC4_MOD_EXT_DST_MOD_INSTANCE(INVALID_PIPELINE_ID); msg.extension |= SOF_IPC4_MOD_EXT_CORE_ID(0); + msg.extension |= SOF_IPC4_MOD_EXT_PARAM_SIZE(sizeof(cfg) / sizeof(uint32_t)); msg.data_size = sizeof(cfg); msg.data_ptr = &cfg; From 8509bb1f11a1fed710271631c2e06fd66452f510 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Thu, 7 Nov 2024 16:51:41 +0100 Subject: [PATCH 77/97] ASoC: dt-bindings: add stm32mp25 support for sai Add STM32MP25 support for STM32 SAI peripheral, through "st,stm32mp25-sai" compatible. Signed-off-by: Olivier Moysan Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20241107155143.1340523-2-olivier.moysan@foss.st.com Signed-off-by: Mark Brown --- .../bindings/sound/st,stm32-sai.yaml | 26 ++++++++++++++++++- 1 file changed, 25 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml index 68f97b462598bc..4a7129d0b15747 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -20,6 +20,7 @@ properties: enum: - st,stm32f4-sai - st,stm32h7-sai + - st,stm32mp25-sai reg: items: @@ -43,9 +44,11 @@ properties: const: 1 clocks: + minItems: 1 maxItems: 3 clock-names: + minItems: 1 maxItems: 3 access-controllers: @@ -156,7 +159,13 @@ allOf: items: - const: x8k - const: x11k - else: + + - if: + properties: + compatible: + contains: + const: st,stm32mph7-sai + then: properties: clocks: items: @@ -170,6 +179,21 @@ allOf: - const: x8k - const: x11k + - if: + properties: + compatible: + contains: + const: st,stm32mp25-sai + then: + properties: + clocks: + items: + - description: pclk feeds the peripheral bus interface. + + clock-names: + items: + - const: pclk + additionalProperties: false examples: From 2cfe1ff22555717bf63526f9e6ea096dde13cc59 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Thu, 7 Nov 2024 16:51:42 +0100 Subject: [PATCH 78/97] ASoC: stm32: sai: add stm32mp25 support Add STM32MP25 support for STM32 SAI. On STM32MP25 the SAI driver does not manage SAI kernel clock rate by chosing its parent clock, dependending on audio stream rate. The driver requests a rate change on SAI kernel clock instead. This rate change is performed with the following guidelines: - Chose highest rate multiple of the audio stream (Try to get clock accuracy within 1000 ppm) - Ensure clock rate compatibility between SAI sub-blocks A&B and between instances sharing the same flexgen. Use clk_rate_exclusive API to fulfill this requirement. The STM32 SAI peripheral does not support the DMA burst mode on STM32MP25. Add a field in compatible structure to manage DMA burst support capability. Signed-off-by: Olivier Moysan Link: https://patch.msgid.link/20241107155143.1340523-3-olivier.moysan@foss.st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai.c | 58 +++++++++++--- sound/soc/stm/stm32_sai.h | 6 ++ sound/soc/stm/stm32_sai_sub.c | 144 ++++++++++++++++++++++++++++++++-- 3 files changed, 191 insertions(+), 17 deletions(-) diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index b45ee7e24f224d..bc8180fc8462db 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -19,26 +19,42 @@ #include "stm32_sai.h" +static int stm32_sai_get_parent_clk(struct stm32_sai_data *sai); + static const struct stm32_sai_conf stm32_sai_conf_f4 = { .version = STM_SAI_STM32F4, .fifo_size = 8, .has_spdif_pdm = false, + .get_sai_ck_parent = stm32_sai_get_parent_clk, }; /* - * Default settings for stm32 H7 socs and next. + * Default settings for STM32H7x socs and STM32MP1x. * These default settings will be overridden if the soc provides * support of hardware configuration registers. + * - STM32H7: rely on default settings + * - STM32MP1: retrieve settings from registers */ static const struct stm32_sai_conf stm32_sai_conf_h7 = { .version = STM_SAI_STM32H7, .fifo_size = 8, .has_spdif_pdm = true, + .get_sai_ck_parent = stm32_sai_get_parent_clk, +}; + +/* + * STM32MP2x: + * - do not use SAI parent clock source selection + * - do not use DMA burst mode + */ +static const struct stm32_sai_conf stm32_sai_conf_mp25 = { + .no_dma_burst = true, }; static const struct of_device_id stm32_sai_ids[] = { { .compatible = "st,stm32f4-sai", .data = (void *)&stm32_sai_conf_f4 }, { .compatible = "st,stm32h7-sai", .data = (void *)&stm32_sai_conf_h7 }, + { .compatible = "st,stm32mp25-sai", .data = (void *)&stm32_sai_conf_mp25 }, {} }; @@ -148,6 +164,29 @@ static int stm32_sai_set_sync(struct stm32_sai_data *sai_client, return ret; } +static int stm32_sai_get_parent_clk(struct stm32_sai_data *sai) +{ + struct device *dev = &sai->pdev->dev; + + sai->clk_x8k = devm_clk_get(dev, "x8k"); + if (IS_ERR(sai->clk_x8k)) { + if (PTR_ERR(sai->clk_x8k) != -EPROBE_DEFER) + dev_err(dev, "missing x8k parent clock: %ld\n", + PTR_ERR(sai->clk_x8k)); + return PTR_ERR(sai->clk_x8k); + } + + sai->clk_x11k = devm_clk_get(dev, "x11k"); + if (IS_ERR(sai->clk_x11k)) { + if (PTR_ERR(sai->clk_x11k) != -EPROBE_DEFER) + dev_err(dev, "missing x11k parent clock: %ld\n", + PTR_ERR(sai->clk_x11k)); + return PTR_ERR(sai->clk_x11k); + } + + return 0; +} + static int stm32_sai_probe(struct platform_device *pdev) { struct stm32_sai_data *sai; @@ -160,6 +199,8 @@ static int stm32_sai_probe(struct platform_device *pdev) if (!sai) return -ENOMEM; + sai->pdev = pdev; + sai->base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(sai->base)) return PTR_ERR(sai->base); @@ -178,15 +219,11 @@ static int stm32_sai_probe(struct platform_device *pdev) "missing bus clock pclk\n"); } - sai->clk_x8k = devm_clk_get(&pdev->dev, "x8k"); - if (IS_ERR(sai->clk_x8k)) - return dev_err_probe(&pdev->dev, PTR_ERR(sai->clk_x8k), - "missing x8k parent clock\n"); - - sai->clk_x11k = devm_clk_get(&pdev->dev, "x11k"); - if (IS_ERR(sai->clk_x11k)) - return dev_err_probe(&pdev->dev, PTR_ERR(sai->clk_x11k), - "missing x11k parent clock\n"); + if (sai->conf.get_sai_ck_parent) { + ret = sai->conf.get_sai_ck_parent(sai); + if (ret) + return ret; + } /* init irqs */ sai->irq = platform_get_irq(pdev, 0); @@ -227,7 +264,6 @@ static int stm32_sai_probe(struct platform_device *pdev) } clk_disable_unprepare(sai->pclk); - sai->pdev = pdev; sai->set_sync = &stm32_sai_set_sync; platform_set_drvdata(pdev, sai); diff --git a/sound/soc/stm/stm32_sai.h b/sound/soc/stm/stm32_sai.h index 33e4bff8c2f575..07b71133db2a64 100644 --- a/sound/soc/stm/stm32_sai.h +++ b/sound/soc/stm/stm32_sai.h @@ -264,16 +264,22 @@ enum stm32_sai_syncout { STM_SAI_SYNC_OUT_B, }; +struct stm32_sai_data; + /** * struct stm32_sai_conf - SAI configuration + * @get_sai_ck_parent: get parent clock of SAI kernel clock * @version: SAI version * @fifo_size: SAI fifo size as words number * @has_spdif_pdm: SAI S/PDIF and PDM features support flag + * @no_dma_burst: Support only DMA single transfers if set */ struct stm32_sai_conf { + int (*get_sai_ck_parent)(struct stm32_sai_data *sai); u32 version; u32 fifo_size; bool has_spdif_pdm; + bool no_dma_burst; }; /** diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index a772bc8ea7be18..27b7fdef7cb04b 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -60,6 +60,9 @@ #define SAI_MCLK_NAME_LEN 32 #define SAI_RATE_11K 11025 +#define SAI_MAX_SAMPLE_RATE_8K 192000 +#define SAI_MAX_SAMPLE_RATE_11K 176400 +#define SAI_CK_RATE_TOLERANCE 1000 /* ppm */ /** * struct stm32_sai_sub_data - private data of SAI sub block (block A or B) @@ -80,6 +83,7 @@ * @dir: SAI block direction (playback or capture). set at init * @master: SAI block mode flag. (true=master, false=slave) set at init * @spdif: SAI S/PDIF iec60958 mode flag. set at init + * @sai_ck_used: flag set while exclusivity on SAI kernel clock is active * @fmt: SAI block format. relevant only for custom protocols. set at init * @sync: SAI block synchronization mode. (none, internal or external) * @synco: SAI block ext sync source (provider setting). (none, sub-block A/B) @@ -93,6 +97,8 @@ * @iec958: iec958 data * @ctrl_lock: control lock * @irq_lock: prevent race condition with IRQ + * @set_sai_ck_rate: set SAI kernel clock rate + * @put_sai_ck_rate: put SAI kernel clock rate */ struct stm32_sai_sub_data { struct platform_device *pdev; @@ -112,6 +118,7 @@ struct stm32_sai_sub_data { int dir; bool master; bool spdif; + bool sai_ck_used; int fmt; int sync; int synco; @@ -125,6 +132,8 @@ struct stm32_sai_sub_data { struct snd_aes_iec958 iec958; struct mutex ctrl_lock; /* protect resources accessed by controls */ spinlock_t irq_lock; /* used to prevent race condition with IRQ */ + int (*set_sai_ck_rate)(struct stm32_sai_sub_data *sai, unsigned int rate); + void (*put_sai_ck_rate)(struct stm32_sai_sub_data *sai); }; enum stm32_sai_fifo_th { @@ -351,8 +360,26 @@ static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai, return ret; } -static int stm32_sai_set_parent_clock(struct stm32_sai_sub_data *sai, - unsigned int rate) +static bool stm32_sai_rate_accurate(unsigned int max_rate, unsigned int rate) +{ + u64 delta, dividend; + int ratio; + + ratio = DIV_ROUND_CLOSEST(max_rate, rate); + if (!ratio) + return false; + + dividend = mul_u32_u32(1000000, abs(max_rate - (ratio * rate))); + delta = div_u64(dividend, max_rate); + + if (delta <= SAI_CK_RATE_TOLERANCE) + return true; + + return false; +} + +static int stm32_sai_set_parent_clk(struct stm32_sai_sub_data *sai, + unsigned int rate) { struct platform_device *pdev = sai->pdev; struct clk *parent_clk = sai->pdata->clk_x8k; @@ -370,6 +397,92 @@ static int stm32_sai_set_parent_clock(struct stm32_sai_sub_data *sai, return ret; } +static void stm32_sai_put_parent_rate(struct stm32_sai_sub_data *sai) +{ + if (sai->sai_ck_used) { + sai->sai_ck_used = false; + clk_rate_exclusive_put(sai->sai_ck); + } +} + +static int stm32_sai_set_parent_rate(struct stm32_sai_sub_data *sai, + unsigned int rate) +{ + struct platform_device *pdev = sai->pdev; + unsigned int sai_ck_rate, sai_ck_max_rate, sai_curr_rate, sai_new_rate; + int div, ret; + + /* + * Set maximum expected kernel clock frequency + * - mclk on or spdif: + * f_sai_ck = MCKDIV * mclk-fs * fs + * Here typical 256 ratio is assumed for mclk-fs + * - mclk off: + * f_sai_ck = MCKDIV * FRL * fs + * Where FRL=[8..256], MCKDIV=[1..n] (n depends on SAI version) + * Set constraint MCKDIV * FRL <= 256, to ensure MCKDIV is in available range + * f_sai_ck = sai_ck_max_rate * pow_of_two(FRL) / 256 + */ + if (!(rate % SAI_RATE_11K)) + sai_ck_max_rate = SAI_MAX_SAMPLE_RATE_11K * 256; + else + sai_ck_max_rate = SAI_MAX_SAMPLE_RATE_8K * 256; + + if (!sai->sai_mclk && !STM_SAI_PROTOCOL_IS_SPDIF(sai)) + sai_ck_max_rate /= DIV_ROUND_CLOSEST(256, roundup_pow_of_two(sai->fs_length)); + + /* + * Request exclusivity, as the clock is shared by SAI sub-blocks and by + * some SAI instances. This allows to ensure that the rate cannot be + * changed while one or more SAIs are using the clock. + */ + clk_rate_exclusive_get(sai->sai_ck); + sai->sai_ck_used = true; + + /* + * Check current kernel clock rate. If it gives the expected accuracy + * return immediately. + */ + sai_curr_rate = clk_get_rate(sai->sai_ck); + if (stm32_sai_rate_accurate(sai_ck_max_rate, sai_curr_rate)) + return 0; + + /* + * Otherwise try to set the maximum rate and check the new actual rate. + * If the new rate does not give the expected accuracy, try to set + * lower rates for the kernel clock. + */ + sai_ck_rate = sai_ck_max_rate; + div = 1; + do { + /* Check new rate accuracy. Return if ok */ + sai_new_rate = clk_round_rate(sai->sai_ck, sai_ck_rate); + if (stm32_sai_rate_accurate(sai_ck_rate, sai_new_rate)) { + ret = clk_set_rate(sai->sai_ck, sai_ck_rate); + if (ret) { + dev_err(&pdev->dev, "Error %d setting sai_ck rate. %s", + ret, ret == -EBUSY ? + "Active stream rates may be in conflict\n" : "\n"); + goto err; + } + + return 0; + } + + /* Try a lower frequency */ + div++; + sai_ck_rate = sai_ck_max_rate / div; + } while (sai_ck_rate > rate); + + /* No accurate rate found */ + dev_err(&pdev->dev, "Failed to find an accurate rate"); + +err: + stm32_sai_put_parent_rate(sai); + + return -EINVAL; +} + static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, unsigned long *prate) { @@ -565,11 +678,15 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai, clk_rate_exclusive_put(sai->sai_mclk); sai->mclk_rate = 0; } + + if (sai->put_sai_ck_rate) + sai->put_sai_ck_rate(sai); + return 0; } - /* If master clock is used, set parent clock now */ - ret = stm32_sai_set_parent_clock(sai, freq); + /* If master clock is used, configure SAI kernel clock now */ + ret = sai->set_sai_ck_rate(sai, freq); if (ret) return ret; @@ -993,7 +1110,7 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, int ret; if (!sai->sai_mclk) { - ret = stm32_sai_set_parent_clock(sai, rate); + ret = sai->set_sai_ck_rate(sai, rate); if (ret) return ret; } @@ -1154,6 +1271,14 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(sai->sai_ck); + /* + * Release kernel clock if following conditions are fulfilled + * - Master clock is not used. Kernel clock won't be released trough sysclk + * - Put handler is defined. Involve that clock is managed exclusively + */ + if (!sai->sai_mclk && sai->put_sai_ck_rate) + sai->put_sai_ck_rate(sai); + spin_lock_irqsave(&sai->irq_lock, flags); sai->substream = NULL; spin_unlock_irqrestore(&sai->irq_lock, flags); @@ -1188,7 +1313,7 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) * constraints). */ sai->dma_params.maxburst = 4; - if (sai->pdata->conf.fifo_size < 8) + if (sai->pdata->conf.fifo_size < 8 || sai->pdata->conf.no_dma_burst) sai->dma_params.maxburst = 1; /* Buswidth will be set by framework at runtime */ sai->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_UNDEFINED; @@ -1526,6 +1651,13 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return -EINVAL; } + if (sai->pdata->conf.get_sai_ck_parent) { + sai->set_sai_ck_rate = stm32_sai_set_parent_clk; + } else { + sai->set_sai_ck_rate = stm32_sai_set_parent_rate; + sai->put_sai_ck_rate = stm32_sai_put_parent_rate; + } + ret = stm32_sai_sub_parse_of(pdev, sai); if (ret) return ret; From c69b7edc10d2fff4bcb3dd464ee26cbf22818fec Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Thu, 7 Nov 2024 15:47:11 +0100 Subject: [PATCH 79/97] ASoC: dt-bindings: add stm32mp25 support for i2s Add STM32MP25 support for STM32 I2S peripheral, through "st,stm32mp25-i2s" compatible. Signed-off-by: Olivier Moysan Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20241107144712.1305638-2-olivier.moysan@foss.st.com Signed-off-by: Mark Brown --- .../bindings/sound/st,stm32-i2s.yaml | 36 +++++++++++++++++-- 1 file changed, 33 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml index 8978f6bd63e59e..b4f44f9c7c7d0f 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml @@ -13,13 +13,11 @@ description: The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. Only some SPI instances support I2S. -allOf: - - $ref: dai-common.yaml# - properties: compatible: enum: - st,stm32h7-i2s + - st,stm32mp25-i2s "#sound-dai-cells": const: 0 @@ -33,6 +31,7 @@ properties: - description: clock feeding the internal clock generator. - description: I2S parent clock for sampling rates multiple of 8kHz. - description: I2S parent clock for sampling rates multiple of 11.025kHz. + minItems: 2 clock-names: items: @@ -40,6 +39,7 @@ properties: - const: i2sclk - const: x8k - const: x11k + minItems: 2 interrupts: maxItems: 1 @@ -79,6 +79,36 @@ required: - dmas - dma-names +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + const: st,stm32h7-i2s + + then: + properties: + clocks: + minItems: 4 + + clock-names: + minItems: 4 + + - if: + properties: + compatible: + contains: + const: st,stm32mp25-i2s + + then: + properties: + clocks: + maxItems: 2 + + clock-names: + maxItems: 2 + unevaluatedProperties: false examples: From 20bf873dcc860507965077ab73bfd4335314b6e2 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Thu, 7 Nov 2024 15:47:12 +0100 Subject: [PATCH 80/97] ASoC: stm32: i2s: add stm32mp25 support Add STM32MP25 support for STM32 I2S. On STM32MP25 the I2S driver does not manage I2S kernel clock rate by choosing its parent clock, depending on audio stream rate. The driver requests a rate change on I2S kernel clock instead. It tries to set the higher possible rate, which is a multiple of the audio stream rate and which gives an accuracy of at least 1000 ppm. Signed-off-by: Olivier Moysan Link: https://patch.msgid.link/20241107144712.1305638-3-olivier.moysan@foss.st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_i2s.c | 211 ++++++++++++++++++++++++++++++++++---- 1 file changed, 189 insertions(+), 22 deletions(-) diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index faa00103ee7f17..19dc61008a75f3 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -200,10 +200,13 @@ enum i2s_datlen { #define STM32_I2S_NAME_LEN 32 #define STM32_I2S_RATE_11K 11025 +#define STM32_I2S_MAX_SAMPLE_RATE_8K 192000 +#define STM32_I2S_MAX_SAMPLE_RATE_11K 176400 +#define STM32_I2S_CLK_RATE_TOLERANCE 1000 /* ppm */ /** * struct stm32_i2s_data - private data of I2S - * @regmap_conf: I2S register map configuration pointer + * @conf: I2S configuration pointer * @regmap: I2S register map pointer * @pdev: device data pointer * @dai_drv: DAI driver pointer @@ -224,11 +227,14 @@ enum i2s_datlen { * @divider: prescaler division ratio * @div: prescaler div field * @odd: prescaler odd field + * @i2s_clk_flg: flag set while exclusivity on I2S kernel clock is active * @refcount: keep count of opened streams on I2S * @ms_flg: master mode flag. + * @set_i2s_clk_rate: set I2S kernel clock rate + * @put_i2s_clk_rate: put I2S kernel clock rate */ struct stm32_i2s_data { - const struct regmap_config *regmap_conf; + const struct stm32_i2s_conf *conf; struct regmap *regmap; struct platform_device *pdev; struct snd_soc_dai_driver *dai_drv; @@ -249,8 +255,21 @@ struct stm32_i2s_data { unsigned int divider; unsigned int div; bool odd; + bool i2s_clk_flg; int refcount; int ms_flg; + int (*set_i2s_clk_rate)(struct stm32_i2s_data *i2s, unsigned int rate); + void (*put_i2s_clk_rate)(struct stm32_i2s_data *i2s); +}; + +/** + * struct stm32_i2s_conf - I2S configuration + * @regmap_conf: regmap configuration pointer + * @get_i2s_clk_parent: get parent clock of I2S kernel clock + */ +struct stm32_i2s_conf { + const struct regmap_config *regmap_conf; + int (*get_i2s_clk_parent)(struct stm32_i2s_data *i2s); }; struct stm32_i2smclk_data { @@ -261,6 +280,8 @@ struct stm32_i2smclk_data { #define to_mclk_data(_hw) container_of(_hw, struct stm32_i2smclk_data, hw) +static int stm32_i2s_get_parent_clk(struct stm32_i2s_data *i2s); + static int stm32_i2s_calc_clk_div(struct stm32_i2s_data *i2s, unsigned long input_rate, unsigned long output_rate) @@ -312,6 +333,33 @@ static int stm32_i2s_set_clk_div(struct stm32_i2s_data *i2s) cgfr_mask, cgfr); } +static bool stm32_i2s_rate_accurate(struct stm32_i2s_data *i2s, + unsigned int max_rate, unsigned int rate) +{ + struct platform_device *pdev = i2s->pdev; + u64 delta, dividend; + int ratio; + + if (!rate) { + dev_err(&pdev->dev, "Unexpected null rate\n"); + return false; + } + + ratio = DIV_ROUND_CLOSEST(max_rate, rate); + if (!ratio) + return false; + + dividend = mul_u32_u32(1000000, abs(max_rate - (ratio * rate))); + delta = div_u64(dividend, max_rate); + + if (delta <= STM32_I2S_CLK_RATE_TOLERANCE) + return true; + + dev_dbg(&pdev->dev, "Rate [%u] not accurate\n", rate); + + return false; +} + static int stm32_i2s_set_parent_clock(struct stm32_i2s_data *i2s, unsigned int rate) { @@ -332,6 +380,87 @@ static int stm32_i2s_set_parent_clock(struct stm32_i2s_data *i2s, return ret; } +static void stm32_i2s_put_parent_rate(struct stm32_i2s_data *i2s) +{ + if (i2s->i2s_clk_flg) { + i2s->i2s_clk_flg = false; + clk_rate_exclusive_put(i2s->i2sclk); + } +} + +static int stm32_i2s_set_parent_rate(struct stm32_i2s_data *i2s, + unsigned int rate) +{ + struct platform_device *pdev = i2s->pdev; + unsigned int i2s_clk_rate, i2s_clk_max_rate, i2s_curr_rate, i2s_new_rate; + int ret, div; + + /* + * Set maximum expected kernel clock frequency + * - mclk on: + * f_i2s_ck = MCKDIV * mclk-fs * fs + * Here typical 256 ratio is assumed for mclk-fs + * - mclk off: + * f_i2s_ck = MCKDIV * FRL * fs + * Where FRL=[16,32], MCKDIV=[1..256] + * f_i2s_ck = i2s_clk_max_rate * 32 / 256 + */ + if (!(rate % STM32_I2S_RATE_11K)) + i2s_clk_max_rate = STM32_I2S_MAX_SAMPLE_RATE_11K * 256; + else + i2s_clk_max_rate = STM32_I2S_MAX_SAMPLE_RATE_8K * 256; + + if (!i2s->i2smclk) + i2s_clk_max_rate /= 8; + + /* Request exclusivity, as the clock may be shared by I2S instances */ + clk_rate_exclusive_get(i2s->i2sclk); + i2s->i2s_clk_flg = true; + + /* + * Check current kernel clock rate. If it gives the expected accuracy + * return immediately. + */ + i2s_curr_rate = clk_get_rate(i2s->i2sclk); + if (stm32_i2s_rate_accurate(i2s, i2s_clk_max_rate, i2s_curr_rate)) + return 0; + + /* + * Otherwise try to set the maximum rate and check the new actual rate. + * If the new rate does not give the expected accuracy, try to set + * lower rates for the kernel clock. + */ + i2s_clk_rate = i2s_clk_max_rate; + div = 1; + do { + /* Check new rate accuracy. Return if ok */ + i2s_new_rate = clk_round_rate(i2s->i2sclk, i2s_clk_rate); + if (stm32_i2s_rate_accurate(i2s, i2s_clk_rate, i2s_new_rate)) { + ret = clk_set_rate(i2s->i2sclk, i2s_clk_rate); + if (ret) { + dev_err(&pdev->dev, "Error %d setting i2s_clk_rate rate. %s", + ret, ret == -EBUSY ? + "Active stream rates may be in conflict\n" : "\n"); + goto err; + } + + return 0; + } + + /* Try a lower frequency */ + div++; + i2s_clk_rate = i2s_clk_max_rate / div; + } while (i2s_clk_rate > rate); + + /* no accurate rate found */ + dev_err(&pdev->dev, "Failed to find an accurate rate"); + +err: + stm32_i2s_put_parent_rate(i2s); + + return -EINVAL; +} + static long stm32_i2smclk_round_rate(struct clk_hw *hw, unsigned long rate, unsigned long *prate) { @@ -635,12 +764,16 @@ static int stm32_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, clk_rate_exclusive_put(i2s->i2smclk); i2s->mclk_rate = 0; } + + if (i2s->put_i2s_clk_rate) + i2s->put_i2s_clk_rate(i2s); + return regmap_update_bits(i2s->regmap, STM32_I2S_CGFR_REG, I2S_CGFR_MCKOE, 0); } /* If master clock is used, set parent clock now */ - ret = stm32_i2s_set_parent_clock(i2s, freq); + ret = i2s->set_i2s_clk_rate(i2s, freq); if (ret) return ret; ret = clk_set_rate_exclusive(i2s->i2smclk, freq); @@ -667,10 +800,11 @@ static int stm32_i2s_configure_clock(struct snd_soc_dai *cpu_dai, u32 cgfr; int ret; - if (!(rate % 11025)) - clk_set_parent(i2s->i2sclk, i2s->x11kclk); - else - clk_set_parent(i2s->i2sclk, i2s->x8kclk); + if (!i2s->mclk_rate) { + ret = i2s->set_i2s_clk_rate(i2s, rate); + if (ret) + return ret; + } i2s_clock_rate = clk_get_rate(i2s->i2sclk); /* @@ -915,6 +1049,14 @@ static void stm32_i2s_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(i2s->i2sclk); + /* + * Release kernel clock if following conditions are fulfilled + * - Master clock is not used. Kernel clock won't be released trough sysclk + * - Put handler is defined. Involve that clock is managed exclusively + */ + if (!i2s->i2smclk && i2s->put_i2s_clk_rate) + i2s->put_i2s_clk_rate(i2s); + spin_lock_irqsave(&i2s->irq_lock, flags); i2s->substream = NULL; spin_unlock_irqrestore(&i2s->irq_lock, flags); @@ -1012,14 +1154,36 @@ static int stm32_i2s_dais_init(struct platform_device *pdev, return 0; } +static const struct stm32_i2s_conf stm32_i2s_conf_h7 = { + .regmap_conf = &stm32_h7_i2s_regmap_conf, + .get_i2s_clk_parent = stm32_i2s_get_parent_clk, +}; + +static const struct stm32_i2s_conf stm32_i2s_conf_mp25 = { + .regmap_conf = &stm32_h7_i2s_regmap_conf +}; + static const struct of_device_id stm32_i2s_ids[] = { - { - .compatible = "st,stm32h7-i2s", - .data = &stm32_h7_i2s_regmap_conf - }, + { .compatible = "st,stm32h7-i2s", .data = &stm32_i2s_conf_h7 }, + { .compatible = "st,stm32mp25-i2s", .data = &stm32_i2s_conf_mp25 }, {}, }; +static int stm32_i2s_get_parent_clk(struct stm32_i2s_data *i2s) +{ + struct device *dev = &i2s->pdev->dev; + + i2s->x8kclk = devm_clk_get(dev, "x8k"); + if (IS_ERR(i2s->x8kclk)) + return dev_err_probe(dev, PTR_ERR(i2s->x8kclk), "Cannot get x8k parent clock\n"); + + i2s->x11kclk = devm_clk_get(dev, "x11k"); + if (IS_ERR(i2s->x11kclk)) + return dev_err_probe(dev, PTR_ERR(i2s->x11kclk), "Cannot get x11k parent clock\n"); + + return 0; +} + static int stm32_i2s_parse_dt(struct platform_device *pdev, struct stm32_i2s_data *i2s) { @@ -1031,8 +1195,8 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev, if (!np) return -ENODEV; - i2s->regmap_conf = device_get_match_data(&pdev->dev); - if (!i2s->regmap_conf) + i2s->conf = device_get_match_data(&pdev->dev); + if (!i2s->conf) return -EINVAL; i2s->base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); @@ -1052,15 +1216,18 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev, return dev_err_probe(&pdev->dev, PTR_ERR(i2s->i2sclk), "Could not get i2sclk\n"); - i2s->x8kclk = devm_clk_get(&pdev->dev, "x8k"); - if (IS_ERR(i2s->x8kclk)) - return dev_err_probe(&pdev->dev, PTR_ERR(i2s->x8kclk), - "Could not get x8k parent clock\n"); + if (i2s->conf->get_i2s_clk_parent) { + i2s->set_i2s_clk_rate = stm32_i2s_set_parent_clock; + } else { + i2s->set_i2s_clk_rate = stm32_i2s_set_parent_rate; + i2s->put_i2s_clk_rate = stm32_i2s_put_parent_rate; + } - i2s->x11kclk = devm_clk_get(&pdev->dev, "x11k"); - if (IS_ERR(i2s->x11kclk)) - return dev_err_probe(&pdev->dev, PTR_ERR(i2s->x11kclk), - "Could not get x11k parent clock\n"); + if (i2s->conf->get_i2s_clk_parent) { + ret = i2s->conf->get_i2s_clk_parent(i2s); + if (ret) + return ret; + } /* Register mclk provider if requested */ if (of_property_present(np, "#clock-cells")) { @@ -1126,7 +1293,7 @@ static int stm32_i2s_probe(struct platform_device *pdev) return ret; i2s->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "pclk", - i2s->base, i2s->regmap_conf); + i2s->base, i2s->conf->regmap_conf); if (IS_ERR(i2s->regmap)) return dev_err_probe(&pdev->dev, PTR_ERR(i2s->regmap), "Regmap init error\n"); From f3c889745cd3500bcbce6f6c8cb7e343f067ac18 Mon Sep 17 00:00:00 2001 From: Tang Bin Date: Fri, 25 Oct 2024 17:09:38 +0800 Subject: [PATCH 81/97] ASoC: mediatek: mt8183: Remove unnecessary variable assignments In the function mt8183_dai_i2s_register, the variable 'ret' is redundant, thus remove it. Signed-off-by: Tang Bin Link: https://patch.msgid.link/20241025090938.3480-1-tangbin@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-dai-i2s.c b/sound/soc/mediatek/mt8183/mt8183-dai-i2s.c index 65e46ebe7be6c6..5cf5592336d33f 100644 --- a/sound/soc/mediatek/mt8183/mt8183-dai-i2s.c +++ b/sound/soc/mediatek/mt8183/mt8183-dai-i2s.c @@ -1036,7 +1036,6 @@ static int mt8183_dai_i2s_set_priv(struct mtk_base_afe *afe) int mt8183_dai_i2s_register(struct mtk_base_afe *afe) { struct mtk_base_afe_dai *dai; - int ret; dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); if (!dai) @@ -1055,9 +1054,5 @@ int mt8183_dai_i2s_register(struct mtk_base_afe *afe) dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_i2s_routes); /* set all dai i2s private data */ - ret = mt8183_dai_i2s_set_priv(afe); - if (ret) - return ret; - - return 0; + return mt8183_dai_i2s_set_priv(afe); } From 08aa540a196a672b8597fb611e2dc25e42986bd9 Mon Sep 17 00:00:00 2001 From: Tang Bin Date: Fri, 8 Nov 2024 09:16:17 +0800 Subject: [PATCH 82/97] ASoC: sma1307: Fix invalid logical judgement In the function sma1307_dai_hw_params_amp, the variable 'ret' has not been assigned a value, so the logical judgement is invalid, thus fix it. Signed-off-by: Tang Bin Link: https://patch.msgid.link/20241108011617.2284-1-tangbin@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/codecs/sma1307.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/sma1307.c b/sound/soc/codecs/sma1307.c index 985a247b331072..81638768ac1254 100644 --- a/sound/soc/codecs/sma1307.c +++ b/sound/soc/codecs/sma1307.c @@ -1191,7 +1191,6 @@ static int sma1307_dai_hw_params_amp(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); unsigned int bclk = 0; - int ret = 0; if (sma1307->format == SND_SOC_DAIFMT_DSP_A) bclk = params_rate(params) * sma1307->frame_size; @@ -1336,8 +1335,6 @@ static int sma1307_dai_hw_params_amp(struct snd_pcm_substream *substream, params_format(params)); return -EINVAL; } - if (ret < 0) - return -EINVAL; return 0; } From c5bbc47f8e094f4489934a244ee1c14947a5075e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 7 Nov 2024 12:02:03 +0200 Subject: [PATCH 83/97] ASoC: Intel: Kconfig: Only select SND_SOC_SDCA if ACPI is enabled The SDCA module have hard dependency to compile on ACPI, it can only be selected if ACPI is also enabled. The SDCA header provides inline prototypes for other modules to compile in case SND_SOC_SDCA is not enabled. Fixes: 845cb1ddf1fc ("ASoC: Intel: Kconfig: select SND_SOC_SDCA by SND_SOC_ACPI_INTEL_SDCA_QUIRKS") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202411070806.UhCRpZok-lkp@intel.com/ Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Link: https://patch.msgid.link/20241107100204.24952-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 46b45f390ae9ea..5021297728564b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -78,7 +78,7 @@ config SND_SOC_ACPI_INTEL_MATCH config SND_SOC_ACPI_INTEL_SDCA_QUIRKS tristate - select SND_SOC_SDCA + select SND_SOC_SDCA if ACPI endif ## SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL From 4f1636e7b0384c43856e1442755f48d7c690c2a9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 7 Nov 2024 12:02:04 +0200 Subject: [PATCH 84/97] ASoC: Intel: Kconfig: Revert make SND_SOC_ACPI_INTEL_MATCH depend on ACPI The acpi-intel-match should be compiled even if ACPI is not selected. The ACPI is not compile time requirement, but the exported symbols are needed for other modules. Fixes: b6bd3f3b6357 ("ASoC: Intel: Kconfig: make SND_SOC_ACPI_INTEL_MATCH depend on ACPI") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202411070806.UhCRpZok-lkp@intel.com/ Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Link: https://patch.msgid.link/20241107100204.24952-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 5021297728564b..2db494b0e3cff0 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -70,8 +70,7 @@ if SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL config SND_SOC_ACPI_INTEL_MATCH tristate - depends on ACPI - select SND_SOC_ACPI + select SND_SOC_ACPI if ACPI select SND_SOC_ACPI_INTEL_SDCA_QUIRKS # this option controls the compilation of ACPI matching tables and # helpers and is not meant to be selected by the user. From eab936aa8500ea09fb3faad25af817221b8a5cb2 Mon Sep 17 00:00:00 2001 From: Luo Yifan Date: Fri, 8 Nov 2024 11:27:02 +0800 Subject: [PATCH 85/97] ALSA: ump: remove unnecessary check on blk The unsigned expression 'blk' will never be negative, so remove the unnecessary check. Signed-off-by: Luo Yifan Link: https://patch.msgid.link/20241108032702.217168-1-luoyifan@cmss.chinamobile.com Signed-off-by: Takashi Iwai --- sound/core/ump.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/ump.c b/sound/core/ump.c index 7d59a0a9b037ad..ab4932dc499f4a 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -366,7 +366,7 @@ int snd_ump_block_new(struct snd_ump_endpoint *ump, unsigned int blk, { struct snd_ump_block *fb, *p; - if (blk < 0 || blk >= SNDRV_UMP_MAX_BLOCKS) + if (blk >= SNDRV_UMP_MAX_BLOCKS) return -EINVAL; if (snd_ump_get_block(ump, blk)) From d1f4390dd28ba110f232615dc4610ac1bb2f39f2 Mon Sep 17 00:00:00 2001 From: Bartosz Golaszewski Date: Fri, 8 Nov 2024 16:07:37 +0200 Subject: [PATCH 86/97] regmap: provide regmap_assign_bits() Add another bits helper to regmap API: this one sets given bits if value is true and clears them if it's false. Suggested-by: Andy Shevchenko Signed-off-by: Bartosz Golaszewski Signed-off-by: Tomi Valkeinen Link: https://patch.msgid.link/20241108-assign-bits-v1-1-382790562d99@ideasonboard.com Signed-off-by: Mark Brown --- include/linux/regmap.h | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/include/linux/regmap.h b/include/linux/regmap.h index da07584284ab36..0d5b74f9bd3211 100644 --- a/include/linux/regmap.h +++ b/include/linux/regmap.h @@ -1335,6 +1335,15 @@ static inline int regmap_clear_bits(struct regmap *map, return regmap_update_bits_base(map, reg, bits, 0, NULL, false, false); } +static inline int regmap_assign_bits(struct regmap *map, unsigned int reg, + unsigned int bits, bool value) +{ + if (value) + return regmap_set_bits(map, reg, bits); + else + return regmap_clear_bits(map, reg, bits); +} + int regmap_test_bits(struct regmap *map, unsigned int reg, unsigned int bits); /** @@ -1803,6 +1812,13 @@ static inline int regmap_clear_bits(struct regmap *map, return -EINVAL; } +static inline int regmap_assign_bits(struct regmap *map, unsigned int reg, + unsigned int bits, bool value) +{ + WARN_ONCE(1, "regmap API is disabled"); + return -EINVAL; +} + static inline int regmap_test_bits(struct regmap *map, unsigned int reg, unsigned int bits) { From ed7bca5b2b891caedf2ed3ffc427eba23559da95 Mon Sep 17 00:00:00 2001 From: Alexey Klimov Date: Fri, 18 Oct 2024 03:54:47 +0100 Subject: [PATCH 87/97] ASoC: qcom: sm8250: add handling of secondary MI2S clock Add handling of clock related to secondary MI2S_RX in sm8250_snd_startup(). Cc: Srinivas Kandagatla Reviewed-by: Dmitry Baryshkov Signed-off-by: Alexey Klimov Link: https://patch.msgid.link/20241018025452.1362293-2-alexey.klimov@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/sm8250.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index 91e9bba192c081..45e0c33fc3f376 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -63,6 +63,14 @@ static int sm8250_snd_startup(struct snd_pcm_substream *substream) snd_soc_dai_set_fmt(cpu_dai, fmt); snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); break; + case SECONDARY_MI2S_RX: + codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_I2S; + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT, + MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_fmt(cpu_dai, fmt); + snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); + break; case TERTIARY_MI2S_RX: codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_I2S; snd_soc_dai_set_sysclk(cpu_dai, From f8da001ae7af0abd9f6250c02c01a1121074ca60 Mon Sep 17 00:00:00 2001 From: John Watts Date: Fri, 8 Nov 2024 12:37:15 +1100 Subject: [PATCH 88/97] ASoC: audio-graph-card2: Purge absent supplies for device tree nodes The audio graph card doesn't mark its subnodes such as multi {}, dpcm {} and c2c {} as not requiring any suppliers. This causes a hang as Linux waits for these phantom suppliers to show up on boot. Make it clear these nodes have no suppliers. Example error message: [ 15.208558] platform 2034000.i2s: deferred probe pending: platform: wait for supplier /sound/multi [ 15.208584] platform sound: deferred probe pending: asoc-audio-graph-card2: parse error Signed-off-by: John Watts Acked-by: Kuninori Morimoto Link: https://patch.msgid.link/20241108-graph_dt_fix-v1-1-173e2f9603d6@jookia.org Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card2.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 4ad3d1b0714f6a..93eee40cec760c 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -270,16 +270,19 @@ static enum graph_type __graph_get_type(struct device_node *lnk) if (of_node_name_eq(np, GRAPH_NODENAME_MULTI)) { ret = GRAPH_MULTI; + fw_devlink_purge_absent_suppliers(&np->fwnode); goto out_put; } if (of_node_name_eq(np, GRAPH_NODENAME_DPCM)) { ret = GRAPH_DPCM; + fw_devlink_purge_absent_suppliers(&np->fwnode); goto out_put; } if (of_node_name_eq(np, GRAPH_NODENAME_C2C)) { ret = GRAPH_C2C; + fw_devlink_purge_absent_suppliers(&np->fwnode); goto out_put; } From d859923faeca740ae9235e2b9328999836e681b9 Mon Sep 17 00:00:00 2001 From: Deep Harsora Date: Mon, 11 Nov 2024 15:06:18 +0800 Subject: [PATCH 89/97] ASoC: intel: sof_sdw: add quirk for Dell SKU MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This patch adds a quirk to include the codec amplifier function for this Dell SKU. Note: In this SKU '0CF1', the RT722 codec amplifier is excluded, and an external amplifier is used instead. Signed-off-by: Deep Harsora Reviewed-by: Liam Girdwood Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Signed-off-by: Bard Liao Link: https://patch.msgid.link/20241111070618.5414-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 35d707d3ae9c71..4a0ab50d1e50dc 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -590,6 +590,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(SOC_SDW_CODEC_SPKR), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CF1") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, { .callback = sof_sdw_quirk_cb, .matches = { From e90dbd3839f554bef35786c4bec8276455691b20 Mon Sep 17 00:00:00 2001 From: anish kumar Date: Sat, 9 Nov 2024 11:22:31 -0800 Subject: [PATCH 90/97] ASoC: machine: update documentation 1. Added clocking details. 2. Updated ways to register the dai's 3. Bit more detail about card registration details. Signed-off-by: anish kumar Reviewed-by: Bagas Sanjaya Link: https://patch.msgid.link/20241109192231.11623-1-yesanishhere@gmail.com Signed-off-by: Mark Brown --- Documentation/sound/soc/machine.rst | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/Documentation/sound/soc/machine.rst b/Documentation/sound/soc/machine.rst index 515c9444deafc6..9db132bc00709e 100644 --- a/Documentation/sound/soc/machine.rst +++ b/Documentation/sound/soc/machine.rst @@ -71,6 +71,18 @@ struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. .ops = &corgi_ops, }; +In the above struct, dai’s are registered using names but you can pass +either dai name or device tree node but not both. Also, names used here +for cpu/codec/platform dais should be globally unique. + +Additionaly below example macro can be used to register cpu, codec and +platform dai:: + + SND_SOC_DAILINK_DEFS(wm2200_cpu_dsp, + DAILINK_COMP_ARRAY(COMP_CPU("samsung-i2s.0")), + DAILINK_COMP_ARRAY(COMP_CODEC("spi0.0", "wm0010-sdi1")), + DAILINK_COMP_ARRAY(COMP_PLATFORM("samsung-i2s.0"))); + struct snd_soc_card then sets up the machine with its DAIs. e.g. :: @@ -81,6 +93,10 @@ struct snd_soc_card then sets up the machine with its DAIs. e.g. .num_links = 1, }; +Following this, ``devm_snd_soc_register_card`` can be used to register +the sound card. During the registration, the individual components +such as the codec, CPU, and platform are probed. If all these components +are successfully probed, the sound card gets registered. Machine Power Map ----------------- @@ -95,3 +111,13 @@ Machine Controls ---------------- Machine specific audio mixer controls can be added in the DAI init function. + + +Clocking Controls +----------------- + +As previously noted, clock configuration is handled within the machine driver. +For details on the clock APIs that the machine driver can utilize for +setup, please refer to Documentation/sound/soc/clocking.rst. However, the +callback needs to be registered by the CPU/Codec/Platform drivers to configure +the clocks that is needed for the corresponding device operation. From 725570f96321f3e0ae1c6a1f80482d2907538d07 Mon Sep 17 00:00:00 2001 From: Marek Vasut Date: Sat, 9 Nov 2024 00:53:47 +0100 Subject: [PATCH 91/97] ASoC: max98088: Remove duplicate DACs This codec only has one set of left and right DACs, remove the duplicate DACs with duplicated bits controlling them as the userspace can set those controls to mismatched value. This most likely does break userspace ABI, but there seem to be no in-kernel users. Signed-off-by: Marek Vasut Link: https://patch.msgid.link/20241108235453.196289-1-marex@denx.de Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 80 ++++++++++++------------------------- 1 file changed, 26 insertions(+), 54 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 8b0645c6346207..adb17a458ca3ec 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -515,10 +515,8 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = { /* Left speaker mixer switch */ static const struct snd_kcontrol_new max98088_left_speaker_mixer_controls[] = { - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("Left DAC Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0), SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 5, 1, 0), SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 6, 1, 0), SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 1, 1, 0), @@ -529,10 +527,8 @@ static const struct snd_kcontrol_new max98088_left_speaker_mixer_controls[] = { /* Right speaker mixer switch */ static const struct snd_kcontrol_new max98088_right_speaker_mixer_controls[] = { - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 0, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC Switch", M98088_REG_2C_MIX_SPK_RIGHT, 7, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98088_REG_2C_MIX_SPK_RIGHT, 0, 1, 0), SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 5, 1, 0), SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 6, 1, 0), SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 1, 1, 0), @@ -543,10 +539,8 @@ static const struct snd_kcontrol_new max98088_right_speaker_mixer_controls[] = { /* Left headphone mixer switch */ static const struct snd_kcontrol_new max98088_left_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("Left DAC Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0), SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_25_MIX_HP_LEFT, 5, 1, 0), SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_25_MIX_HP_LEFT, 6, 1, 0), SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_25_MIX_HP_LEFT, 1, 1, 0), @@ -557,10 +551,8 @@ static const struct snd_kcontrol_new max98088_left_hp_mixer_controls[] = { /* Right headphone mixer switch */ static const struct snd_kcontrol_new max98088_right_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_26_MIX_HP_RIGHT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_26_MIX_HP_RIGHT, 0, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_26_MIX_HP_RIGHT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_26_MIX_HP_RIGHT, 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC Switch", M98088_REG_26_MIX_HP_RIGHT, 7, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98088_REG_26_MIX_HP_RIGHT, 0, 1, 0), SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_26_MIX_HP_RIGHT, 5, 1, 0), SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_26_MIX_HP_RIGHT, 6, 1, 0), SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_26_MIX_HP_RIGHT, 1, 1, 0), @@ -571,10 +563,8 @@ static const struct snd_kcontrol_new max98088_right_hp_mixer_controls[] = { /* Left earpiece/receiver mixer switch */ static const struct snd_kcontrol_new max98088_left_rec_mixer_controls[] = { - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("Left DAC Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0), SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_28_MIX_REC_LEFT, 5, 1, 0), SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_28_MIX_REC_LEFT, 6, 1, 0), SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_28_MIX_REC_LEFT, 1, 1, 0), @@ -585,10 +575,8 @@ static const struct snd_kcontrol_new max98088_left_rec_mixer_controls[] = { /* Right earpiece/receiver mixer switch */ static const struct snd_kcontrol_new max98088_right_rec_mixer_controls[] = { - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_29_MIX_REC_RIGHT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_29_MIX_REC_RIGHT, 0, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_29_MIX_REC_RIGHT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_29_MIX_REC_RIGHT, 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC Switch", M98088_REG_29_MIX_REC_RIGHT, 7, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98088_REG_29_MIX_REC_RIGHT, 0, 1, 0), SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_29_MIX_REC_RIGHT, 5, 1, 0), SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_29_MIX_REC_RIGHT, 6, 1, 0), SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_29_MIX_REC_RIGHT, 1, 1, 0), @@ -717,13 +705,9 @@ static const struct snd_soc_dapm_widget max98088_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", M98088_REG_4C_PWR_EN_IN, 1, 0), SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", M98088_REG_4C_PWR_EN_IN, 0, 0), - SND_SOC_DAPM_DAC("DACL1", "HiFi Playback", + SND_SOC_DAPM_DAC("DACL", "HiFi Playback", M98088_REG_4D_PWR_EN_OUT, 1, 0), - SND_SOC_DAPM_DAC("DACR1", "HiFi Playback", - M98088_REG_4D_PWR_EN_OUT, 0, 0), - SND_SOC_DAPM_DAC("DACL2", "Aux Playback", - M98088_REG_4D_PWR_EN_OUT, 1, 0), - SND_SOC_DAPM_DAC("DACR2", "Aux Playback", + SND_SOC_DAPM_DAC("DACR", "HiFi Playback", M98088_REG_4D_PWR_EN_OUT, 0, 0), SND_SOC_DAPM_PGA("HP Left Out", M98088_REG_4D_PWR_EN_OUT, @@ -819,10 +803,8 @@ static const struct snd_soc_dapm_widget max98088_dapm_widgets[] = { static const struct snd_soc_dapm_route max98088_audio_map[] = { /* Left headphone output mixer */ - {"Left HP Mixer", "Left DAC1 Switch", "DACL1"}, - {"Left HP Mixer", "Left DAC2 Switch", "DACL2"}, - {"Left HP Mixer", "Right DAC1 Switch", "DACR1"}, - {"Left HP Mixer", "Right DAC2 Switch", "DACR2"}, + {"Left HP Mixer", "Left DAC Switch", "DACL"}, + {"Left HP Mixer", "Right DAC Switch", "DACR"}, {"Left HP Mixer", "MIC1 Switch", "MIC1 Input"}, {"Left HP Mixer", "MIC2 Switch", "MIC2 Input"}, {"Left HP Mixer", "INA1 Switch", "INA1 Input"}, @@ -831,10 +813,8 @@ static const struct snd_soc_dapm_route max98088_audio_map[] = { {"Left HP Mixer", "INB2 Switch", "INB2 Input"}, /* Right headphone output mixer */ - {"Right HP Mixer", "Left DAC1 Switch", "DACL1"}, - {"Right HP Mixer", "Left DAC2 Switch", "DACL2" }, - {"Right HP Mixer", "Right DAC1 Switch", "DACR1"}, - {"Right HP Mixer", "Right DAC2 Switch", "DACR2"}, + {"Right HP Mixer", "Left DAC Switch", "DACL"}, + {"Right HP Mixer", "Right DAC Switch", "DACR"}, {"Right HP Mixer", "MIC1 Switch", "MIC1 Input"}, {"Right HP Mixer", "MIC2 Switch", "MIC2 Input"}, {"Right HP Mixer", "INA1 Switch", "INA1 Input"}, @@ -843,10 +823,8 @@ static const struct snd_soc_dapm_route max98088_audio_map[] = { {"Right HP Mixer", "INB2 Switch", "INB2 Input"}, /* Left speaker output mixer */ - {"Left SPK Mixer", "Left DAC1 Switch", "DACL1"}, - {"Left SPK Mixer", "Left DAC2 Switch", "DACL2"}, - {"Left SPK Mixer", "Right DAC1 Switch", "DACR1"}, - {"Left SPK Mixer", "Right DAC2 Switch", "DACR2"}, + {"Left SPK Mixer", "Left DAC Switch", "DACL"}, + {"Left SPK Mixer", "Right DAC Switch", "DACR"}, {"Left SPK Mixer", "MIC1 Switch", "MIC1 Input"}, {"Left SPK Mixer", "MIC2 Switch", "MIC2 Input"}, {"Left SPK Mixer", "INA1 Switch", "INA1 Input"}, @@ -855,10 +833,8 @@ static const struct snd_soc_dapm_route max98088_audio_map[] = { {"Left SPK Mixer", "INB2 Switch", "INB2 Input"}, /* Right speaker output mixer */ - {"Right SPK Mixer", "Left DAC1 Switch", "DACL1"}, - {"Right SPK Mixer", "Left DAC2 Switch", "DACL2"}, - {"Right SPK Mixer", "Right DAC1 Switch", "DACR1"}, - {"Right SPK Mixer", "Right DAC2 Switch", "DACR2"}, + {"Right SPK Mixer", "Left DAC Switch", "DACL"}, + {"Right SPK Mixer", "Right DAC Switch", "DACR"}, {"Right SPK Mixer", "MIC1 Switch", "MIC1 Input"}, {"Right SPK Mixer", "MIC2 Switch", "MIC2 Input"}, {"Right SPK Mixer", "INA1 Switch", "INA1 Input"}, @@ -867,10 +843,8 @@ static const struct snd_soc_dapm_route max98088_audio_map[] = { {"Right SPK Mixer", "INB2 Switch", "INB2 Input"}, /* Earpiece/Receiver output mixer */ - {"Left REC Mixer", "Left DAC1 Switch", "DACL1"}, - {"Left REC Mixer", "Left DAC2 Switch", "DACL2"}, - {"Left REC Mixer", "Right DAC1 Switch", "DACR1"}, - {"Left REC Mixer", "Right DAC2 Switch", "DACR2"}, + {"Left REC Mixer", "Left DAC Switch", "DACL"}, + {"Left REC Mixer", "Right DAC Switch", "DACR"}, {"Left REC Mixer", "MIC1 Switch", "MIC1 Input"}, {"Left REC Mixer", "MIC2 Switch", "MIC2 Input"}, {"Left REC Mixer", "INA1 Switch", "INA1 Input"}, @@ -879,10 +853,8 @@ static const struct snd_soc_dapm_route max98088_audio_map[] = { {"Left REC Mixer", "INB2 Switch", "INB2 Input"}, /* Earpiece/Receiver output mixer */ - {"Right REC Mixer", "Left DAC1 Switch", "DACL1"}, - {"Right REC Mixer", "Left DAC2 Switch", "DACL2"}, - {"Right REC Mixer", "Right DAC1 Switch", "DACR1"}, - {"Right REC Mixer", "Right DAC2 Switch", "DACR2"}, + {"Right REC Mixer", "Left DAC Switch", "DACL"}, + {"Right REC Mixer", "Right DAC Switch", "DACR"}, {"Right REC Mixer", "MIC1 Switch", "MIC1 Input"}, {"Right REC Mixer", "MIC2 Switch", "MIC2 Input"}, {"Right REC Mixer", "INA1 Switch", "INA1 Input"}, From 1bd775da9ba919b87b2313a78d5957afc1a62dde Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 6 Nov 2024 00:10:44 +0000 Subject: [PATCH 92/97] ASoC: add symmetric_ prefix for dai->rate/channels/sample_bits snd_soc_dai has rate/channels/sample_bits parameter, but it is only valid if symmetry is being enforced by symmetric_xxx flag on driver. It is very difficult to know about it from current naming, and easy to misunderstand it. add symmetric_ prefix for it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87zfmd8bnf.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 6 ++--- sound/soc/mediatek/mt8188/mt8188-dai-pcm.c | 2 +- sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 2 +- sound/soc/mediatek/mt8365/mt8365-dai-dmic.c | 6 ++--- sound/soc/mediatek/mt8365/mt8365-dai-pcm.c | 2 +- sound/soc/soc-compress.c | 4 +-- sound/soc/soc-pcm.c | 29 +++++++++++---------- 7 files changed, 26 insertions(+), 25 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index b275201b02f608..aab57c19f62b23 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -449,9 +449,9 @@ struct snd_soc_dai { struct snd_soc_dai_stream stream[SNDRV_PCM_STREAM_LAST + 1]; /* Symmetry data - only valid if symmetry is being enforced */ - unsigned int rate; - unsigned int channels; - unsigned int sample_bits; + unsigned int symmetric_rate; + unsigned int symmetric_channels; + unsigned int symmetric_sample_bits; /* parent platform/codec */ struct snd_soc_component *component; diff --git a/sound/soc/mediatek/mt8188/mt8188-dai-pcm.c b/sound/soc/mediatek/mt8188/mt8188-dai-pcm.c index 5bc854a8f3df34..8ca7cc75e21dc8 100644 --- a/sound/soc/mediatek/mt8188/mt8188-dai-pcm.c +++ b/sound/soc/mediatek/mt8188/mt8188-dai-pcm.c @@ -128,7 +128,7 @@ static int mtk_dai_pcm_configure(struct snd_pcm_substream *substream, unsigned int lrck_inv; unsigned int bck_inv; unsigned int fmt; - unsigned int bit_width = dai->sample_bits; + unsigned int bit_width = dai->symmetric_sample_bits; unsigned int val = 0; unsigned int mask = 0; int fs = 0; diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c b/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c index 6d6d79300d5121..cdc16057d50e2f 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c @@ -127,7 +127,7 @@ static int mtk_dai_pcm_configure(struct snd_pcm_substream *substream, unsigned int lrck_inv; unsigned int bck_inv; unsigned int fmt; - unsigned int bit_width = dai->sample_bits; + unsigned int bit_width = dai->symmetric_sample_bits; unsigned int val = 0; unsigned int mask = 0; int fs = 0; diff --git a/sound/soc/mediatek/mt8365/mt8365-dai-dmic.c b/sound/soc/mediatek/mt8365/mt8365-dai-dmic.c index f9945c2a2cd13c..0bac143b48bfb3 100644 --- a/sound/soc/mediatek/mt8365/mt8365-dai-dmic.c +++ b/sound/soc/mediatek/mt8365/mt8365-dai-dmic.c @@ -118,13 +118,13 @@ static int mt8365_dai_configure_dmic(struct mtk_base_afe *afe, unsigned int clk_phase_sel_ch1 = dmic_data->clk_phase_sel_ch1; unsigned int clk_phase_sel_ch2 = dmic_data->clk_phase_sel_ch2; unsigned int val = 0; - unsigned int rate = dai->rate; - int reg = get_chan_reg(dai->channels); + unsigned int rate = dai->symmetric_rate; + int reg = get_chan_reg(dai->symmetric_channels); if (reg < 0) return -EINVAL; - dmic_data->dmic_channel = dai->channels; + dmic_data->dmic_channel = dai->symmetric_channels; val |= DMIC_TOP_CON_SDM3_LEVEL_MODE; diff --git a/sound/soc/mediatek/mt8365/mt8365-dai-pcm.c b/sound/soc/mediatek/mt8365/mt8365-dai-pcm.c index f85ec07249c3b7..3373b88da28eaa 100644 --- a/sound/soc/mediatek/mt8365/mt8365-dai-pcm.c +++ b/sound/soc/mediatek/mt8365/mt8365-dai-pcm.c @@ -44,7 +44,7 @@ static int mt8365_dai_configure_pcm1(struct snd_pcm_substream *substream, bool lrck_inv = pcm_priv->lrck_inv; bool bck_inv = pcm_priv->bck_inv; unsigned int fmt = pcm_priv->format; - unsigned int bit_width = dai->sample_bits; + unsigned int bit_width = dai->symmetric_sample_bits; unsigned int val = 0; if (!slave_mode) { diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3c514703fa33d6..563dc0767c177a 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -69,10 +69,10 @@ static int soc_compr_clean(struct snd_compr_stream *cstream, int rollback) snd_soc_dai_digital_mute(codec_dai, 1, stream); if (!snd_soc_dai_active(cpu_dai)) - cpu_dai->rate = 0; + cpu_dai->symmetric_rate = 0; if (!snd_soc_dai_active(codec_dai)) - codec_dai->rate = 0; + codec_dai->symmetric_rate = 0; snd_soc_link_compr_shutdown(cstream, rollback); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index fb7f25fd8ec5bd..1150455619aa49 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -447,13 +447,13 @@ static void soc_pcm_set_dai_params(struct snd_soc_dai *dai, struct snd_pcm_hw_params *params) { if (params) { - dai->rate = params_rate(params); - dai->channels = params_channels(params); - dai->sample_bits = snd_pcm_format_physical_width(params_format(params)); + dai->symmetric_rate = params_rate(params); + dai->symmetric_channels = params_channels(params); + dai->symmetric_sample_bits = snd_pcm_format_physical_width(params_format(params)); } else { - dai->rate = 0; - dai->channels = 0; - dai->sample_bits = 0; + dai->symmetric_rate = 0; + dai->symmetric_channels = 0; + dai->symmetric_sample_bits = 0; } } @@ -467,14 +467,14 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, return 0; #define __soc_pcm_apply_symmetry(name, NAME) \ - if (soc_dai->name && (soc_dai->driver->symmetric_##name || \ - rtd->dai_link->symmetric_##name)) { \ + if (soc_dai->symmetric_##name && \ + (soc_dai->driver->symmetric_##name || rtd->dai_link->symmetric_##name)) { \ dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %s to %d\n",\ - #name, soc_dai->name); \ + #name, soc_dai->symmetric_##name); \ \ ret = snd_pcm_hw_constraint_single(substream->runtime, \ SNDRV_PCM_HW_PARAM_##NAME,\ - soc_dai->name); \ + soc_dai->symmetric_##name); \ if (ret < 0) { \ dev_err(soc_dai->dev, \ "ASoC: Unable to apply %s constraint: %d\n",\ @@ -510,9 +510,11 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, if (symmetry) \ for_each_rtd_cpu_dais(rtd, i, cpu_dai) \ if (!snd_soc_dai_is_dummy(cpu_dai) && \ - cpu_dai->xxx && cpu_dai->xxx != d.xxx) { \ + cpu_dai->symmetric_##xxx && \ + cpu_dai->symmetric_##xxx != d.symmetric_##xxx) { \ dev_err(rtd->dev, "ASoC: unmatched %s symmetry: %s:%d - %s:%d\n", \ - #xxx, cpu_dai->name, cpu_dai->xxx, d.name, d.xxx); \ + #xxx, cpu_dai->name, cpu_dai->symmetric_##xxx, \ + d.name, d.symmetric_##xxx); \ return -EINVAL; \ } @@ -783,8 +785,7 @@ static int soc_pcm_clean(struct snd_soc_pcm_runtime *rtd, /* Make sure DAI parameters cleared if the DAI becomes inactive */ for_each_rtd_dais(rtd, i, dai) { - if (snd_soc_dai_active(dai) == 0 && - (dai->rate || dai->channels || dai->sample_bits)) + if (snd_soc_dai_active(dai) == 0) soc_pcm_set_dai_params(dai, NULL); } } From 98d34ddd43de0b040bd3ff74b511be7074b0b310 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Tue, 5 Nov 2024 14:59:41 +0100 Subject: [PATCH 93/97] ASoC: dt-bindings: stm32: add missing port property Add missing port property in STM32 SPDIFRX binding. This will prevent potential warning: Unevaluated properties are not allowed ('port' was unexpected) Signed-off-by: Olivier Moysan Link: https://patch.msgid.link/20241105135942.526624-1-olivier.moysan@foss.st.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml index 3dedc81ec12f67..56c5738ea4c532 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml +++ b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml @@ -50,6 +50,10 @@ properties: resets: maxItems: 1 + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + access-controllers: minItems: 1 maxItems: 2 From 9a59718a5340aa0240a442115eb499de2ed18ee4 Mon Sep 17 00:00:00 2001 From: Suraj Sonawane Date: Thu, 7 Nov 2024 12:06:09 +0530 Subject: [PATCH 94/97] ASoc: SOF: ipc4-pcm: fix uninit-value in sof_ipc4_pcm_dai_link_fixup_rate Fix an issue detected by the Smatch tool: sound/soc/sof/ipc4-pcm.c: sof_ipc4_pcm_dai_link_fixup_rate() error: uninitialized symbol 'be_rate'. The warning highlights a case where `be_rate` could remain uninitialized if `num_input_formats` is zero, which would cause undefined behavior when setting `rate->min` and `rate->max` based on `be_rate`. To address this issue, a `WARN_ON_ONCE(!num_input_formats)` check was added to ensure `num_input_formats` is greater than zero. If this condition fails, the function returns `-EINVAL`, preventing the use of an uninitialized `be_rate`. This change improves the robustness of the function by catching an invalid state early and providing better feedback during development. Signed-off-by: Suraj Sonawane Link: https://patch.msgid.link/20241107063609.11627-1-surajsonawane0215@gmail.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 4df2be3d39eba0..18fff2df76f97e 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -603,6 +603,9 @@ static int sof_ipc4_pcm_dai_link_fixup_rate(struct snd_sof_dev *sdev, unsigned int be_rate; int i; + if (WARN_ON_ONCE(!num_input_formats)) + return -EINVAL; + /* * Copier does not change sampling rate, so we * need to only consider the input pin information. From db8f6b5e95b246b7e152424b69fa983f5cde8904 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 12 Nov 2024 12:22:21 +0800 Subject: [PATCH 95/97] Revert "regcache: Store values more directly in maple trees" This reverts commit 49a85851b14cf6630013d1b9bae2ac2345c9430b. The commit was reverted in upstream. Signed-off-by: Bard Liao --- drivers/base/regmap/regcache-maple.c | 320 ++++++++++++++++----------- 1 file changed, 187 insertions(+), 133 deletions(-) diff --git a/drivers/base/regmap/regcache-maple.c b/drivers/base/regmap/regcache-maple.c index a6a894aaf41bef..8d27d3653ea3e7 100644 --- a/drivers/base/regmap/regcache-maple.c +++ b/drivers/base/regmap/regcache-maple.c @@ -13,53 +13,26 @@ #include "internal.h" -#if IS_ENABLED(CONFIG_64BIT) -/* - * On 64 bit systems uintptr_t will be 64 bit but unsigned long 32 bit - * so we can store the register values directly in the maple tree. We - * need to always set an out of bounds bit due to maple tree's - * handling of NULL. - */ -#define REGCACHE_MAPLE_NONZERO (1UL << 32) - -static unsigned long regcache_maple_entry_to_value(void *entry) -{ - return (uintptr_t)entry & ~REGCACHE_MAPLE_NONZERO; -} - -static int regcache_maple_write(struct regmap *map, unsigned int reg, - unsigned int val) +static int regcache_maple_read(struct regmap *map, + unsigned int reg, unsigned int *value) { struct maple_tree *mt = map->cache; MA_STATE(mas, mt, reg, reg); - uintptr_t entry = val | REGCACHE_MAPLE_NONZERO; - int ret; - - /* - * This is safe because the regmap lock means the Maple lock - * is redundant, but we need to take it due to lockdep asserts - * in the maple tree code. - */ - mas_lock(&mas); + unsigned long *entry; - mas_set_range(&mas, reg, reg); - ret = mas_store_gfp(&mas, (void *)entry, map->alloc_flags); + rcu_read_lock(); - mas_unlock(&mas); + entry = mas_walk(&mas); + if (!entry) { + rcu_read_unlock(); + return -ENOENT; + } - return ret; -} + *value = entry[reg - mas.index]; -#else + rcu_read_unlock(); -/* - * On 32 bit systems we can't distingush between NULL and a valid 0 - * value in a 32 bit register so kmalloc() extra storage for the - * values. - */ -static unsigned long regcache_maple_entry_to_value(unsigned long *entry) -{ - return *entry; + return 0; } static int regcache_maple_write(struct regmap *map, unsigned int reg, @@ -67,24 +40,49 @@ static int regcache_maple_write(struct regmap *map, unsigned int reg, { struct maple_tree *mt = map->cache; MA_STATE(mas, mt, reg, reg); - unsigned long *entry; + unsigned long *entry, *upper, *lower; + unsigned long index, last; + size_t lower_sz, upper_sz; int ret; rcu_read_lock(); entry = mas_walk(&mas); if (entry) { - *entry = val; + entry[reg - mas.index] = val; rcu_read_unlock(); return 0; } + /* Any adjacent entries to extend/merge? */ + mas_set_range(&mas, reg - 1, reg + 1); + index = reg; + last = reg; + + lower = mas_find(&mas, reg - 1); + if (lower) { + index = mas.index; + lower_sz = (mas.last - mas.index + 1) * sizeof(unsigned long); + } + + upper = mas_find(&mas, reg + 1); + if (upper) { + last = mas.last; + upper_sz = (mas.last - mas.index + 1) * sizeof(unsigned long); + } + rcu_read_unlock(); - entry = kmalloc(sizeof(unsigned long), map->alloc_flags); + entry = kmalloc((last - index + 1) * sizeof(unsigned long), + map->alloc_flags); if (!entry) return -ENOMEM; - *entry = val; + + if (lower) + memcpy(entry, lower, lower_sz); + entry[reg - index] = val; + if (upper) + memcpy(&entry[reg - index + 1], upper, upper_sz); /* * This is safe because the regmap lock means the Maple lock @@ -93,75 +91,97 @@ static int regcache_maple_write(struct regmap *map, unsigned int reg, */ mas_lock(&mas); - mas_set_range(&mas, reg, reg); + mas_set_range(&mas, index, last); ret = mas_store_gfp(&mas, entry, map->alloc_flags); mas_unlock(&mas); - if (ret != 0) - kfree(entry); + if (ret == 0) { + kfree(lower); + kfree(upper); + } return ret; } -#endif - -static int regcache_maple_read(struct regmap *map, - unsigned int reg, unsigned int *value) -{ - struct maple_tree *mt = map->cache; - MA_STATE(mas, mt, reg, reg); - void *entry; - - rcu_read_lock(); - - entry = mas_walk(&mas); - if (!entry) { - rcu_read_unlock(); - return -ENOENT; - } - - *value = regcache_maple_entry_to_value(entry); - - rcu_read_unlock(); - - return 0; -} static int regcache_maple_drop(struct regmap *map, unsigned int min, unsigned int max) { struct maple_tree *mt = map->cache; MA_STATE(mas, mt, min, max); - unsigned long *entry; + unsigned long *entry, *lower, *upper; + /* initialized to work around false-positive -Wuninitialized warning */ + unsigned long lower_index = 0, lower_last = 0; + unsigned long upper_index, upper_last; int ret = 0; + lower = NULL; + upper = NULL; + mas_lock(&mas); mas_for_each(&mas, entry, max) { - if (WARN_ON_ONCE(mas.index != mas.last)) { - ret = -EFAULT; - goto out; - } - - if (mas.index < min || mas.last > max) - continue; - /* * This is safe because the regmap lock means the * Maple lock is redundant, but we need to take it due * to lockdep asserts in the maple tree code. */ - if (!IS_ENABLED(CONFIG_64BIT)) { - mas_unlock(&mas); - kfree(entry); - mas_lock(&mas); + mas_unlock(&mas); + + /* Do we need to save any of this entry? */ + if (mas.index < min) { + lower_index = mas.index; + lower_last = min -1; + + lower = kmemdup_array(entry, + min - mas.index, sizeof(*lower), + map->alloc_flags); + if (!lower) { + ret = -ENOMEM; + goto out_unlocked; + } + } + + if (mas.last > max) { + upper_index = max + 1; + upper_last = mas.last; + + upper = kmemdup_array(&entry[max - mas.index + 1], + mas.last - max, sizeof(*upper), + map->alloc_flags); + if (!upper) { + ret = -ENOMEM; + goto out_unlocked; + } } + kfree(entry); + mas_lock(&mas); mas_erase(&mas); + + /* Insert new nodes with the saved data */ + if (lower) { + mas_set_range(&mas, lower_index, lower_last); + ret = mas_store_gfp(&mas, lower, map->alloc_flags); + if (ret != 0) + goto out; + lower = NULL; + } + + if (upper) { + mas_set_range(&mas, upper_index, upper_last); + ret = mas_store_gfp(&mas, upper, map->alloc_flags); + if (ret != 0) + goto out; + upper = NULL; + } } out: mas_unlock(&mas); +out_unlocked: + kfree(lower); + kfree(upper); return ret; } @@ -171,7 +191,6 @@ static int regcache_maple_sync_block(struct regmap *map, unsigned long *entry, unsigned int min, unsigned int max) { void *buf; - unsigned int v; unsigned long r; size_t val_bytes = map->format.val_bytes; int ret = 0; @@ -192,25 +211,19 @@ static int regcache_maple_sync_block(struct regmap *map, unsigned long *entry, } /* Render the data for a raw write */ - for (r = min; r < max + 1; r++) { - ret = regcache_maple_read(map, r, &v); - if (ret != 0) { - kfree(buf); - goto out; - } - regcache_set_val(map, buf, r - min, v); + for (r = min; r < max; r++) { + regcache_set_val(map, buf, r - min, + entry[r - mas->index]); } - ret = _regmap_raw_write(map, min, buf, (max - min + 1) * val_bytes, + ret = _regmap_raw_write(map, min, buf, (max - min) * val_bytes, false); kfree(buf); } else { - for (r = min; r < max + 1; r++) { - ret = regcache_maple_read(map, r, &v); - if (ret != 0) - goto out; - ret = _regmap_write(map, r, v); + for (r = min; r < max; r++) { + ret = _regmap_write(map, r, + entry[r - mas->index]); if (ret != 0) goto out; } @@ -228,7 +241,9 @@ static int regcache_maple_sync(struct regmap *map, unsigned int min, struct maple_tree *mt = map->cache; unsigned long *entry; MA_STATE(mas, mt, min, max); - unsigned int v, last, sync_start; + unsigned long lmin = min; + unsigned long lmax = max; + unsigned int r, v, sync_start; int ret = 0; bool sync_needed = false; @@ -237,39 +252,34 @@ static int regcache_maple_sync(struct regmap *map, unsigned int min, rcu_read_lock(); mas_for_each(&mas, entry, max) { - /* Flush if we hit a gap in the cache */ - if (sync_needed && mas.index != last + 1) { + for (r = max(mas.index, lmin); r <= min(mas.last, lmax); r++) { + v = entry[r - mas.index]; + + if (regcache_reg_needs_sync(map, r, v)) { + if (!sync_needed) { + sync_start = r; + sync_needed = true; + } + continue; + } + + if (!sync_needed) + continue; + ret = regcache_maple_sync_block(map, entry, &mas, - sync_start, last); + sync_start, r); if (ret != 0) goto out; sync_needed = false; } - v = regcache_maple_entry_to_value(entry); - - if (regcache_reg_needs_sync(map, mas.index, v)) { - if (!sync_needed) { - sync_start = mas.index; - sync_needed = true; - } - last = mas.index; - continue; + if (sync_needed) { + ret = regcache_maple_sync_block(map, entry, &mas, + sync_start, r); + if (ret != 0) + goto out; + sync_needed = false; } - - if (!sync_needed) - continue; - - ret = regcache_maple_sync_block(map, entry, &mas, - sync_start, last); - if (ret != 0) - goto out; - sync_needed = false; - } - - if (sync_needed) { - ret = regcache_maple_sync_block(map, entry, &mas, - sync_start, last); } out: @@ -291,10 +301,8 @@ static int regcache_maple_exit(struct regmap *map) return 0; mas_lock(&mas); - if (!IS_ENABLED(CONFIG_64BIT)) { - mas_for_each(&mas, entry, UINT_MAX) - kfree(entry); - } + mas_for_each(&mas, entry, UINT_MAX) + kfree(entry); __mt_destroy(mt); mas_unlock(&mas); @@ -304,11 +312,41 @@ static int regcache_maple_exit(struct regmap *map) return 0; } +static int regcache_maple_insert_block(struct regmap *map, int first, + int last) +{ + struct maple_tree *mt = map->cache; + MA_STATE(mas, mt, first, last); + unsigned long *entry; + int i, ret; + + entry = kcalloc(last - first + 1, sizeof(unsigned long), map->alloc_flags); + if (!entry) + return -ENOMEM; + + for (i = 0; i < last - first + 1; i++) + entry[i] = map->reg_defaults[first + i].def; + + mas_lock(&mas); + + mas_set_range(&mas, map->reg_defaults[first].reg, + map->reg_defaults[last].reg); + ret = mas_store_gfp(&mas, entry, map->alloc_flags); + + mas_unlock(&mas); + + if (ret) + kfree(entry); + + return ret; +} + static int regcache_maple_init(struct regmap *map) { struct maple_tree *mt; int i; int ret; + int range_start; mt = kmalloc(sizeof(*mt), map->alloc_flags); if (!mt) @@ -317,14 +355,30 @@ static int regcache_maple_init(struct regmap *map) mt_init(mt); - for (i = 0; i < map->num_reg_defaults; i++) { - ret = regcache_maple_write(map, - map->reg_defaults[i].reg, - map->reg_defaults[i].def); - if (ret != 0) - goto err; + if (!map->num_reg_defaults) + return 0; + + range_start = 0; + + /* Scan for ranges of contiguous registers */ + for (i = 1; i < map->num_reg_defaults; i++) { + if (map->reg_defaults[i].reg != + map->reg_defaults[i - 1].reg + 1) { + ret = regcache_maple_insert_block(map, range_start, + i - 1); + if (ret != 0) + goto err; + + range_start = i; + } } + /* Add the last block */ + ret = regcache_maple_insert_block(map, range_start, + map->num_reg_defaults - 1); + if (ret != 0) + goto err; + return 0; err: From 127d3623665ee60dfb0e45f51ba7f2c4eb2b6b7e Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 12 Nov 2024 12:46:40 +0800 Subject: [PATCH 96/97] Revert "ASoC: Intel: Kconfig: select SND_SOC_SDCA by SND_SOC_ACPI_INTEL_SDCA_QUIRKS" This reverts commit fddade38dc6dc60a4f7911e0a688241ec61180e3. Will use upstream version to avoid merge conflict. Signed-off-by: Bard Liao --- sound/soc/intel/Kconfig | 2 +- sound/soc/sdca/Kconfig | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 46b45f390ae9ea..f644e94234280c 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -78,7 +78,7 @@ config SND_SOC_ACPI_INTEL_MATCH config SND_SOC_ACPI_INTEL_SDCA_QUIRKS tristate - select SND_SOC_SDCA + imply SND_SOC_SDCA endif ## SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL diff --git a/sound/soc/sdca/Kconfig b/sound/soc/sdca/Kconfig index ee20b9914aa1fa..07f6822fa614f3 100644 --- a/sound/soc/sdca/Kconfig +++ b/sound/soc/sdca/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_SOC_SDCA - tristate + tristate "ASoC SDCA library" depends on ACPI help This option enables support for the MIPI SoundWire Device From 503963d60d62a914fd8bb83261d19ee0073d08a5 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 12 Nov 2024 12:48:12 +0800 Subject: [PATCH 97/97] Revert "ASoC: Intel: Kconfig: make SND_SOC_ACPI_INTEL_MATCH depend on ACPI" This reverts commit b33e4da42171a96e061a620bbf04de5d95fa9402. Will use the upstream version to avoid merge conflict. Signed-off-by: Bard Liao --- sound/soc/intel/Kconfig | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f644e94234280c..14461dee3e52ce 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -70,8 +70,7 @@ if SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL config SND_SOC_ACPI_INTEL_MATCH tristate - depends on ACPI - select SND_SOC_ACPI + select SND_SOC_ACPI if ACPI select SND_SOC_ACPI_INTEL_SDCA_QUIRKS # this option controls the compilation of ACPI matching tables and # helpers and is not meant to be selected by the user.