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[PATCH] ALSA: usb: Add Plantronics C320-M USB ctrl msg delay quirk
by John Ernberg 03 Mar '21

03 Mar '21

The microphone in the Plantronics C320-M headset will randomly fail to initialize properly, at least when using Microsoft Teams. Introducing a 20ms delay on the control messages appears to resolve the issue. Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1065 Tested-by: Andreas Kempe <kempe(a)lysator.liu.se> Signed-off-by: John Ernberg <john.ernberg(a)actia.se> --- Note: Microsoft Teams is the only application tests have been performed with and no other kinds of investigations have been performed, like USB packet logging. Originally reported as a possible PulseAudio issue, where it was suggested to try a delay quirk in the kernel. If other tests should be performed, please let me know which tests and I will try to do them. All testing was done on 5.10, patch was then applied on sound.git master for submission I did not register to the mailing lists, please keep me CC on any comments. Thanks! sound/usb/quirks.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d02dac5fcd40..a31b7db92b1e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1673,6 +1673,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) msleep(20); + /* + * Plantronics C320-M needs a delay to avoid random + * microhpone failures. + */ + if (chip->usb_id == USB_ID(0x047f, 0xc025) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + msleep(20); + /* Zoom R16/24, many Logitech(at least H650e/H570e/BCC950), * Jabra 550a, Kingston HyperX needs a tiny delay here, * otherwise requests like get/set frequency return -- 2.30.0

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[PATCH 0/4] ALSA: hda/cirrus: Add support for CS8409 HDA bridge and CS42L42 companion codec.
by Vitaly Rodionov 03 Mar '21

03 Mar '21

Dell's laptops Inspiron 3500, Inspiron 3501, Inspiron 3505 are using Cirrus Logic CS8409 HDA bridge with CS42L42 companion codec. The CS8409 is a multichannel HD audio routing controller. CS8409 includes support for four channels of digital microphone data and two bidirectional ASPs for up to 32 channels of TDM data or 4 channels of I2S data. The CS8409 is intended to be used with a remote companion codec that implements high performance analog functions in close physical proximity to the end-equipment audio port or speaker driver. The CS42L42 is a low-power audio codec with integrated MIPI SoundWire interface or I2C/I2S/TDM interfaces designed for portable applications. It provides a high-dynamic range, stereo DAC for audio playback and a mono high-dynamic-range ADC for audio capture Stefan Binding (1): ALSA: hda/cirrus: Add Headphone and Headset MIC Volume Control Vitaly Rodionov (3): ALSA: hda/cirrus: Increase AUTO_CFG_MAX_INS from 8 to 18 ALSA: hda/cirrus: Add support for CS8409 HDA bridge and CS42L42 companion codec. ALSA: hda/cirrus: Add jack detect interrupt support from CS42L42 companion codec. sound/pci/hda/hda_auto_parser.h | 2 +- sound/pci/hda/hda_local.h | 2 +- sound/pci/hda/patch_cirrus.c | 1033 +++++++++++++++++++++++++++++++ 3 files changed, 1035 insertions(+), 2 deletions(-) -- 2.25.1

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[PATCH] ASoC: SOF: Intel: broadwell: fix mutual exclusion with catpt driver
by Pierre-Louis Bossart 03 Mar '21

03 Mar '21

In v5.10, the "haswell" driver was replaced by the "catpt" driver, but the mutual exclusion with the SOF driver was not updated. This leads to errors with card names and UCM profiles not being loaded by PulseAudio. This fix should only be applied on v5.10-stable, the mutual exclusion was removed in 5.11. Reported-by: David Ward <david.ward(a)ll.mit.edu> Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=211985 Fixes: 6cbfa11d2694 ("ASoC: Intel: Select catpt and deprecate haswell") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: <stable(a)kernel.org> --- sound/soc/sof/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index de7ff2d097ab..6708a2c5a838 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -84,7 +84,7 @@ config SND_SOC_SOF_BAYTRAIL config SND_SOC_SOF_BROADWELL_SUPPORT bool "SOF support for Broadwell" - depends on SND_SOC_INTEL_HASWELL=n + depends on SND_SOC_INTEL_CATPT=n help This adds support for Sound Open Firmware for Intel(R) platforms using the Broadwell processors. -- 2.25.1

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[PATCH v2 0/5] soundwire: qcom: various improvements
by Srinivas Kandagatla 03 Mar '21

03 Mar '21

Thanks for reviewing v1 of this patchset! During testing SoundWire controller on SM8250 MTP, we found few issues like all the interrupts are not handled, all transport parameters are not read from device tree. Other major issue was register read/writes which was interrupt based was an overhead and puts lot of limitation on context it can be used from. With previous approach number of interrupts generated after enumeration are around 130: $ cat /proc/interrupts | grep soundwire 21: 130 0 0 0 0 0 0 0 GICv3 234 Edge soundwire after this patch they are just 3 interrupts $ cat /proc/interrupts | grep soundwire 21: 3 0 0 0 0 0 0 0 GICv3 234 Edge soundwire So this patchset add various improvements to the existing driver to address above issues. Tested it on SM8250 MTP with 2x WSA881x speakers, HeadPhones on WCD938x via lpass-rx-macro and Analog MICs via lpass-tx-macro. Also tested on DragonBoard DB845c with 2xWSA881x speakers. Changes since v1: - changed bgp_count to blk_group_count as suggested by Pierre - used version raw value as suggested by VKoul - updated read fifo logic and interrupt handling to do while() - renamed some of the variable to make reading easy! - removed patch that parses version Srinivas Kandagatla (5): soundwire: qcom: add support to missing transport params soundwire: qcom: set continue execution flag for ignored commands soundwire: qcom: start the clock during initialization soundwire: qcom: update register read/write routine soundwire: qcom: add support to new interrupts drivers/soundwire/qcom.c | 438 ++++++++++++++++++++++++++++++--------- 1 file changed, 335 insertions(+), 103 deletions(-) -- 2.21.0

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[PATCH] ALSA: hda/realtek: Apply dual codec quirks for MSI Godlike X570 board
by Takashi Iwai 03 Mar '21

03 Mar '21

There is another MSI board (1462:cc34) that has dual Realtek codecs, and we need to apply the existing quirk for fixing the conflicts of Master control. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211743 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 7c24d13181e1..b47504fa8dfd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2532,6 +2532,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), + SND_PCI_QUIRK(0x1462, 0xcc34, "MSI Godlike X570", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), -- 2.26.2

1 0

[PATCH] ALSA: hda - bind headset buttons to the headphone jack
by Hui Wang 03 Mar '21

03 Mar '21

With the HDA driver, if the headset buttons are supported, an audio Jack will be created for them. This audio Jack is a bit confusing to users since it can't report headphone/mic insertion events but it claims to support these events. And in addition, the driver already builds a headphone Jack and a mic Jack, and most of those buttons are used for headphone playback, so do some change to bind those buttons to the headphone Jack. After this change, the key events are generated from NID 0x55, and are reported to the input layer via headphone jack (NID 0x21). If there is no headphone Jack, then build an audio jack to support those buttons same as previously. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_jack.c | 43 ++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_jack.h | 5 ++++ sound/pci/hda/patch_realtek.c | 23 ++++++++++++++++--- 3 files changed, 67 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index b8b568046592..640b72b598e8 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -388,6 +388,39 @@ int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, } EXPORT_SYMBOL_GPL(snd_hda_jack_set_gating_jack); +/** + * snd_hda_jack_bind_keymap - bind keys generated from one NID to another jack. + * @codec: the HDA codec + * @key_nid: key event is generated by this pin NID + * @keymap: map of key type and key code + * @jack_nid: key reports to the jack of this pin NID + * + * This function is used in the case of key is generated from one NID while is + * reported to the jack of another NID. + */ +int snd_hda_jack_bind_keymap(struct hda_codec *codec, hda_nid_t key_nid, + const struct hda_jack_keymap *keymap, + hda_nid_t jack_nid) +{ + const struct hda_jack_keymap *map; + struct hda_jack_tbl *key_gen = snd_hda_jack_tbl_get(codec, key_nid); + struct hda_jack_tbl *report_to = snd_hda_jack_tbl_get(codec, jack_nid); + + WARN_ON(codec->dp_mst); + + if (!key_gen || !report_to || !report_to->jack) + return -EINVAL; + + key_gen->key_report_jack = jack_nid; + + if (keymap) + for (map = keymap; map->type; map++) + snd_jack_set_key(report_to->jack, map->type, map->key); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hda_jack_bind_keymap); + /** * snd_hda_jack_report_sync - sync the states of all jacks and report if changed * @codec: the HDA codec @@ -651,7 +684,15 @@ void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res) } if (!event) return; - event->jack_dirty = 1; + + if (event->key_report_jack) { + struct hda_jack_tbl *report_to = + snd_hda_jack_tbl_get_mst(codec, event->key_report_jack, + event->dev_id); + if (report_to) + report_to->jack_dirty = 1; + } else + event->jack_dirty = 1; call_jack_callback(codec, res, event); snd_hda_jack_report_sync(codec); diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 8ceaf0ef5df1..339746dd9578 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -40,6 +40,7 @@ struct hda_jack_tbl { unsigned int block_report:1; /* in a transitional state - do not report to userspace */ hda_nid_t gating_jack; /* valid when gating jack plugged */ hda_nid_t gated_jack; /* gated is dependent on this jack */ + hda_nid_t key_report_jack; /* key reports to this jack */ int type; int button_state; struct snd_jack *jack; @@ -99,6 +100,10 @@ snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid); +int snd_hda_jack_bind_keymap(struct hda_codec *codec, hda_nid_t key_nid, + const struct hda_jack_keymap *keymap, + hda_nid_t jack_nid); + u32 snd_hda_jack_pin_sense(struct hda_codec *codec, hda_nid_t nid, int dev_id); /* the jack state returned from snd_hda_jack_detect_state() */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 52907506e16e..f381474f9403 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3102,6 +3102,16 @@ static void alc_headset_btn_callback(struct hda_codec *codec, if (jack->unsol_res & (7 << 10)) report |= SND_JACK_BTN_3; + if (jack->jack->key_report_jack) { + struct hda_jack_tbl *report_to = + snd_hda_jack_tbl_get(codec, jack->jack->key_report_jack); + + if (report_to) { + report_to->button_state = report; + return; + } + } + jack->jack->button_state = report; } @@ -3163,16 +3173,23 @@ static void alc_fixup_headset_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin; switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: spec->has_hs_key = 1; snd_hda_jack_detect_enable_callback(codec, 0x55, alc_headset_btn_callback); - snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, - SND_JACK_HEADSET, alc_headset_btn_keymap); break; - case HDA_FIXUP_ACT_INIT: + case HDA_FIXUP_ACT_BUILD: + hp_pin = alc_get_hp_pin(spec); + if (!hp_pin || snd_hda_jack_bind_keymap(codec, 0x55, + alc_headset_btn_keymap, + hp_pin)) + snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", + false, SND_JACK_HEADSET, + alc_headset_btn_keymap); + alc_enable_headset_jack_key(codec); break; } -- 2.25.1

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[PATCH] ASoC: soc-core: Prevent warning if no DMI table is present
by Jon Hunter 03 Mar '21

03 Mar '21

Many systems do not provide a DMI table and on these systems a warning, such as the following, is printed on boot ... WARNING KERN tegra-audio-graph-card sound: ASoC: no DMI vendor name! If DMI support is enabled in the kernel, there is no simple way to detect if a DMI table is table or not. Note that the variable 'dmi_available' is not exported and so cannot be used by kernel modules. It could be possible to have every ASoC sound card driver set the long name to avoid the above message, but it might be intentional for the long name, that we fall back to using the sound card name. Therefore, make this a debug print by default to avoid the warning. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f6d4e99b590c..f1189e7c1fcc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1576,7 +1576,7 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) /* make up dmi long name as: vendor-product-version-board */ vendor = dmi_get_system_info(DMI_BOARD_VENDOR); if (!vendor || !is_dmi_valid(vendor)) { - dev_warn(card->dev, "ASoC: no DMI vendor name!\n"); + dev_dbg(card->dev, "ASoC: no DMI vendor name!\n"); return 0; } -- 2.25.1

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[PATCH v5 1/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Mar '21

03 Mar '21

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. CPU0 CPU1 dpcm_be_dai_trigger(): dpcm_be_disconnect(): for_each_dpcm_be(fe, stream, dpcm) { spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); struct snd_soc_pcm_runtime *be = dpcm->be; <-- Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_be_dai_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- include/sound/soc-dpcm.h | 5 ++++ sound/soc/soc-pcm.c | 59 +++++++++++++++++++++++++++++++++------- 2 files changed, 54 insertions(+), 10 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 0f6c50b17bba..599cd6054bc3 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -103,6 +103,11 @@ struct snd_soc_dpcm_runtime { int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */ }; +struct snd_soc_dpcm_rtd_list { + int num_rtds; + struct snd_soc_pcm_runtime *rtds[0]; +}; + #define for_each_dpcm_fe(be, stream, _dpcm) \ list_for_each_entry(_dpcm, &(be)->dpcm[stream].fe_clients, list_fe) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 14d85ca1e435..ccda0f77e827 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2028,15 +2028,51 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_be_list_create(struct snd_soc_pcm_runtime *fe, int stream, + struct snd_soc_dpcm_rtd_list **be_list) +{ + struct snd_soc_dpcm *dpcm; + unsigned long flags; + int size, i, ret = 0; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + + size = 0; + for_each_dpcm_be(fe, stream, dpcm) + size++; + + *be_list = kzalloc(struct_size(*be_list, rtds, size), GFP_ATOMIC); + if (*be_list) { + i = 0; + for_each_dpcm_be(fe, stream, dpcm) + (*be_list)->rtds[i++] = dpcm->be; + + if (i != size) + dev_err(fe->dev, "ASoC: abnormal change in be clients: %d -> %d\n", + size, i); + + (*be_list)->num_rtds = i; + } else { + ret = -ENOMEM; + } + + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} + int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { - struct snd_soc_dpcm *dpcm; - int ret = 0; + struct snd_soc_dpcm_rtd_list *be_list; + int i, ret = 0; - for_each_dpcm_be(fe, stream, dpcm) { + ret = dpcm_be_list_create(fe, stream, &be_list); + if (ret < 0) + return ret; - struct snd_soc_pcm_runtime *be = dpcm->be; + for (i = 0; i < be_list->num_rtds; i++) { + struct snd_soc_pcm_runtime *be = be_list->rtds[i]; struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); @@ -2056,7 +2092,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2066,7 +2102,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2076,7 +2112,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2090,7 +2126,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2103,7 +2139,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2116,13 +2152,16 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret < 0) + break; } + kfree(be_list); return ret; } EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger); -- 2.21.0

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[PATCH 0/3] ALSA: digigram: fix kernel-doc warnings
by Pierre-Louis Bossart 03 Mar '21

03 Mar '21

make W=1 reports these warnings with v5.12-rc1. Pierre-Louis Bossart (3): ALSA: vx: fix kernel-doc warning ALSA: pci: mixart: fix kernel-doc warning ALSA: pci: vx222: fix kernel-doc warning sound/drivers/vx/vx_core.c | 2 +- sound/pci/mixart/mixart_hwdep.c | 3 ++- sound/pci/vx222/vx222_ops.c | 10 +++++----- 3 files changed, 8 insertions(+), 7 deletions(-) -- 2.25.1

2 4

[PATCH] ALSA: hda/realtek: Add quirk for Intel NUC 10
by Werner Sembach 03 Mar '21

03 Mar '21

ALSA: hda/realtek: Add quirk for Intel NUC 10 This adds a new SND_PCI_QUIRK(...) and applies it to the Intel NUC 10 devices. This fixes the issue of the devices not having audio input and output on the headset jack because the kernel does not recognize when something is plugged in. The new quirk was inspired by the quirk for the Intel NUC 8 devices, but it turned out that the NUC 10 uses another pin. This information was acquired by black box testing likely pins. Co-developed-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> --- Resend of this patch with git send-email, because last patch got tabs replaced with spaces. >From d281364b8ca6c054a0e5ce20caa599bf7d08160d Mon Sep 17 00:00:00 2001 From: Werner Sembach <wse(a)tuxedocomputers.com> Date: Fri, 26 Feb 2021 13:54:30 +0100 Subject: [PATCH] Fix Intel NUC10 no output and input on headset jack --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290645516313..c14d624dbaf1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6362,6 +6362,7 @@ enum { ALC269_FIXUP_LEMOTE_A1802, ALC269_FIXUP_LEMOTE_A190X, ALC256_FIXUP_INTEL_NUC8_RUGGED, + ALC256_FIXUP_INTEL_NUC10, ALC255_FIXUP_XIAOMI_HEADSET_MIC, ALC274_FIXUP_HP_MIC, ALC274_FIXUP_HP_HEADSET_MIC, @@ -7744,6 +7745,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC256_FIXUP_INTEL_NUC10] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, [ALC255_FIXUP_XIAOMI_HEADSET_MIC] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -8183,6 +8193,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), + SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), #if 0 /* Below is a quirk table taken from the old code. -- 2.25.1

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[PATCH v5 0/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Mar '21

03 Mar '21

On Wed, 2021-02-17 16:29, Takashi Iwai wrote: >On Wed, 17 Feb 2021 05:31:49 +0100, >Gyeongtaek Lee wrote: >> >> If stop by underrun and DPCM BE disconnection is run simultaneously, >> data abort can be occurred by the sequence below. >> >> CPU0 CPU1 >> dpcm_be_dai_trigger(): dpcm_be_disconnect(): >> >> for_each_dpcm_be(fe, stream, dpcm) { >> >> spin_lock_irqsave(&fe->card->dpcm_lock, flags); >> list_del(&dpcm->list_be); >> list_del(&dpcm->list_fe); >> spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); >> kfree(dpcm); >> >> struct snd_soc_pcm_runtime *be = dpcm->be; <-- Accessing freed memory >> >> To prevent this situation, dpcm_lock should be acquired during >> iteration of dpcm list in dpcm_be_dai_trigger(). > >I don't think we can apply spin lock there blindly. There is >non-atomic PCM that must not take a spin lock there, too. Hi, Takashi I fixed the patch to hold dpcm_lock during accessing dpcm related data only, not on the trigger callback of driver. Thank you. Gyeongtaek Lee > > >thanks, > >Takashi > Gyeongtaek Lee (1): ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger() include/sound/soc-dpcm.h | 5 ++++ sound/soc/soc-pcm.c | 59 +++++++++++++++++++++++++++++++++------- 2 files changed, 54 insertions(+), 10 deletions(-) base-commit: fe07bfda2fb9cdef8a4d4008a409bb02f35f1bd8 -- 2.21.0

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Re: [PATCH] ASoC: wm8960: Remove bitclk relax condition
by S.j. Wang 03 Mar '21

03 Mar '21

> > > > Using a higher bitclk then expected doesn't always work. > > Here is an example: > > > > aplay -Dhw:0,0 -d 5 -r 48000 -f S24_LE -c 2 audio48k24b2c.wav > > > > In this case, the required bitclk is 48000 * 24 * 2 = 2304000 but the > > closest bitclk that can be derived is 3072000. Since the clock is > > faster than expected, it will start to send bytes from the next > > channel so the sound will be corrupted. > > > > Fixes: 82bab88910ee ("ASoC: codec: wm8960: Relax bit clock computation > > when using PLL") > > Fixes: 3c01b9ee2ab9 ("ASoC: codec: wm8960: Relax bit clock > > computation") > > Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > I think this is probably going to need a much more involved fix. > The problem is that there are systems that depend on this behaviour, so you > can't just flat out revert it. And to be fair the I2S specification says that bit > clock can run at a higher rate than required for the data, so the behaviour is > correct as well. > > Probably the best solution here is to add additional contraints from the > machine driver on which rates/bit depths/channels are supported. > Just double check the issue, the real reason is in below: The call sequence in wm8960_configure_clocking is ret = wm8960_configure_sysclk(); if (ret >= 0) goto configure_clock; .... ret = wm8960_configure_pll(); configure_clock: ... wm8960_configure_sysclk is called before wm8960_configure_pll, as there is bitclk relax on both functions, so wm8960_configure_sysclk always return success, then wm8960_configure_pll() never be called. So bitclk relax condition should be removed in wm8960_configure_sysclk, That wm8960_configure_pll can be called, and there is also bitclk relax function in wm8960_configure_pll. I will update and resend this patch. Best regards Wang shengjiu

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Fiio M3K: Failed to find a working profile
by GitHub issues - edited 03 Mar '21

03 Mar '21

alsa-project/alsa-lib issue #123 was edited from x1unix: Hello, I have a [Fiio M3K](https://fiio.com/m3k) player which I often use as an external USB DAC. It worked well with Pipewire, but I had some other issues with Pipewire so I had to rollback to PulseAudio. Unfortunately, PulseAudio and module-alsa-card cannot setup this device, so it doesn't work. Here is a small portion of PulseAudio log (full log is available at link below): ``` E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="2" name="usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" card_name="alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 (alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00) failed to load module. D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC2 is accessible: yes D: [pulseaudio] module-udev-detect.c: /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 is busy: no ``` I'm using ArchLinux with Linux 5.11.2 and PulseAudio 14.2. **Logs and debug info** USB device in `lsusb` output: ``` Bus 005 Device 022: ID 2972:0044 FiiO Electronics Technology FiiO M3K ``` aplay, pulseaudio debug log and other info I've posted as gist: https://gist.github.com/x1unix/dae3fb810ac499cc63362a805220968f Issue URL : https://github.com/alsa-project/alsa-lib/issues/123 Repository URL: https://github.com/alsa-project/alsa-lib

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Fiio M3K: Failed to find a working profile
by GitHub issues - edited 03 Mar '21

03 Mar '21

alsa-project/alsa-lib issue #123 was edited from x1unix: Hello, I have a [Fiio M3K](https://fiio.com/m3k) player which I often use as an external USB DAC. It worked well with Pipewire, but I had some other issues with Pipewire so I had to rollback to PulseAudio. Unfortunately, PulseAudio and module-alsa-card cannot setup this device, so it doesn't work. Here is a small portion of PulseAudio log (full log is available at link below): ``` E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="2" name="usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" card_name="alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 (alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00) failed to load module. D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC2 is accessible: yes D: [pulseaudio] module-udev-detect.c: /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 is busy: no ``` I'm using ArchLinux with Linux 5.11.2 and PulseAudio 14.2. **Logs and debug info** aplay, pulseaudio debug log and other info I've posted as gist: https://gist.github.com/x1unix/dae3fb810ac499cc63362a805220968f Issue URL : https://github.com/alsa-project/alsa-lib/issues/123 Repository URL: https://github.com/alsa-project/alsa-lib

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Fiio M3K: Failed to find a working profile
by GitHub issues - edited 03 Mar '21

03 Mar '21

alsa-project/alsa-lib issue #123 was edited from x1unix: Hello, I have a [Fiio M3K](https://fiio.com/m3k) player which I often use as an external USB DAC. It worked well with Pipewire, but I had some other issues with Pipewire so I had to rollback to PulseAudio. Unfortunately, PulseAudio and module-alsa-card cannot setup this device, so it doesn't work. Here is a small portion of PulseAudio log (full log is available at link below): ``` E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="2" name="usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" card_name="alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 (alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00) failed to load module. D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC2 is accessible: yes D: [pulseaudio] module-udev-detect.c: /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 is busy: no ``` I'm using ArchLinux and Linux 5.11.2 and PulseAudio 14.2. **Logs and debug info** aplay, pulseaudio debug log and other info I've posted as gist: https://gist.github.com/x1unix/dae3fb810ac499cc63362a805220968f Issue URL : https://github.com/alsa-project/alsa-lib/issues/123 Repository URL: https://github.com/alsa-project/alsa-lib

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Fiio M3K: Failed to find a working profile
by GitHub issues - opened 03 Mar '21

03 Mar '21

alsa-project/alsa-lib issue #123 was opened from x1unix: Hello, I have a [Fiio M3K](https://fiio.com/m3k) player which I often use as an external USB DAC. It worked well with Pipewire, but I had some other issues with Pipewire so I had to rollback to PulseAudio. Unfortunately, PulseAudio and module-alsa-card cannot setup this device, so it doesn't work. Here is a small portion of PulseAudio log (full log is available at link below): ``` E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="2" name="usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" card_name="alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 (alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00) failed to load module. D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC2 is accessible: yes D: [pulseaudio] module-udev-detect.c: /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 is busy: no ``` I'm using ArchLinux and Linux 5.11.2. **Logs and debug info** aplay, pulseaudio debug log and other info I've posted as gist: https://gist.github.com/x1unix/dae3fb810ac499cc63362a805220968f Issue URL : https://github.com/alsa-project/alsa-lib/issues/123 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 00/15] Report jack and button detection + Capture Support
by Lucas Tanure 03 Mar '21

03 Mar '21

Hi All, Here is a patch series for reporting to user space jack and button events and add the support for Capture. With some cleanups and fixes along the way. Regards, Lucas Tanure Lucas Tanure (12): ASoC: cs42l42: Don't enable/disable regulator at Bias Level ASoC: cs42l42: Always wait at least 3ms after reset ASoC: cs42l42: Remove power if the driver is being removed ASoC: cs42l42: Disable regulators if probe fails ASoC: cs42l42: Fix Bitclock polarity inversion ASoC: cs42l42: Provide finer control on playback path ASoC: cs42l42: Set clock source for both ways of stream ASoC: cs42l42: Fix channel width support ASoC: cs42l42: Add Capture Support ASoC: cs42l42: Report jack and button detection ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called ASoC: cs42l42: Fix mixer volume control Richard Fitzgerald (3): ASoC: cs42l42: Wait at least 150us after writing SCLK_PRESENT ASoC: cs42l42: Only start PLL if it is needed ASoC: cs42l42: Wait for PLL to lock before switching to it sound/soc/codecs/cs42l42.c | 497 +++++++++++++++++++++---------------- sound/soc/codecs/cs42l42.h | 42 +++- 2 files changed, 317 insertions(+), 222 deletions(-) -- 2.30.1

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[PATCH] ALSA: hda/realtek: Add quirk for Intel NUC 10
by Werner Sembach 02 Mar '21

02 Mar '21

ALSA: hda/realtek: Add quirk for Intel NUC 10 This adds a new SND_PCI_QUIRK(...) and applies it to the Intel NUC 10 devices. This fixes the issue of the devices not having audio input and output on the headset jack because the kernel does not recognize when something is plugged in. The new quirk was inspired by the quirk for the Intel NUC 8 devices, but it turned out that the NUC 10 uses another pin. This information was acquired by black box testing likely pins. Co-developed-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> --- Forgot to add the "From"-line >From d281364b8ca6c054a0e5ce20caa599bf7d08160d Mon Sep 17 00:00:00 2001 From: Werner Sembach <wse(a)tuxedocomputers.com> Date: Fri, 26 Feb 2021 13:54:30 +0100 Subject: [PATCH] Fix Intel NUC10 no output and input on headset jack --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290645516313..c14d624dbaf1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6362,6 +6362,7 @@ enum { ALC269_FIXUP_LEMOTE_A1802, ALC269_FIXUP_LEMOTE_A190X, ALC256_FIXUP_INTEL_NUC8_RUGGED, + ALC256_FIXUP_INTEL_NUC10, ALC255_FIXUP_XIAOMI_HEADSET_MIC, ALC274_FIXUP_HP_MIC, ALC274_FIXUP_HP_HEADSET_MIC, @@ -7744,6 +7745,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC256_FIXUP_INTEL_NUC10] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, [ALC255_FIXUP_XIAOMI_HEADSET_MIC] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -8183,6 +8193,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), + SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), #if 0 /* Below is a quirk table taken from the old code. -- 2.25.1

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[PATCH] ALSA: hda/realtek: Add quirk for Clevo NH55RZQ
by Werner Sembach 02 Mar '21

02 Mar '21

From: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> ALSA: hda/realtek: Add quirk for Clevo NH55RZQ This applies a SND_PCI_QUIRK(...) to the Clevo NH55RZQ barebone. This fixes the issue of the device not recognizing a pluged in microphone. The device has both, a microphone only jack, and a speaker + microphone combo jack. The combo jack already works. The microphone-only jack does not recognize when a device is pluged in without this patch. Signed-off-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Co-developed-by: Werner Sembach <wse(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> --- This is a resend of the patch because I missed that the editor I used to write the commit message replaced tabs with spaces. >From 2835edd753fd19c72a644dccb7e941cfc0ecdf8e Mon Sep 17 00:00:00 2001 From: Werner Sembach <wse(a)tuxedocomputers.com> Date: Fri, 26 Feb 2021 13:50:15 +0100 Subject: [PATCH] Fix device detection on microphone jack of Clevo NH55RZQ --- 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 290645516313..8014e80d72c3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8089,6 +8089,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8551, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[5|7][0-9]RZ[Q]", ALC269_FIXUP_DMIC), SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.25.1

2 2

[PATCH] ALSA: hda/realtek: Add quirk for Clevo NH55RZQ
by Werner Sembach 02 Mar '21

02 Mar '21

From: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> ALSA: hda/realtek: Add quirk for Clevo NH55RZQ This applies a SND_PCI_QUIRK(...) to the Clevo NH55RZQ barebone. This fixes the issue of the device not recognizing a pluged in microphone. The device has both, a microphone only jack, and a speaker + microphone combo jack. The combo jack already works. The microphone-only jack does not recognize when a device is pluged in without this patch. Signed-off-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Co-developed-by: Werner Sembach <wse(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> --- Third time's the charm, now using git send-email, I'm really sorry for the spam. >From 2835edd753fd19c72a644dccb7e941cfc0ecdf8e Mon Sep 17 00:00:00 2001 From: Werner Sembach <wse(a)tuxedocomputers.com> Date: Fri, 26 Feb 2021 13:50:15 +0100 Subject: [PATCH] Fix device detection on microphone jack of Clevo NH55RZQ --- 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 290645516313..8014e80d72c3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8089,6 +8089,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8551, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[5|7][0-9]RZ[Q]", ALC269_FIXUP_DMIC), SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.25.1

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[PATCH v3] ALSA: hda/hdmi: let new platforms assign the pcm slot dynamically
by Hui Wang 02 Mar '21

02 Mar '21

If the platform set the dyn_pcm_assign to true, it will call hdmi_find_pcm_slot() to find a pcm slot when hdmi/dp monitor is connected and need to create a pcm. So far only intel_hsw_common_init() and patch_nvhdmi() set the dyn_pcm_assign to true, here we let tgl platforms assign the pcm slot dynamically first, if the driver runs for a period of time and there is no regression reported, we could set no_fixed_assgin to true in the intel_hsw_common_init(), and then set it to true in the patch_nvhdmi(). This change comes from the discussion between Takashi and Kai Vehmanen. Please refer to: https://github.com/alsa-project/alsa-lib/pull/118 Suggested-and-reviewed-by: Takashi Iwai <tiwai(a)suse.de> Suggested-and-reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e405be7929e3..e6d0843ee9df 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -157,6 +157,7 @@ struct hdmi_spec { bool dyn_pin_out; bool dyn_pcm_assign; + bool dyn_pcm_no_legacy; bool intel_hsw_fixup; /* apply Intel platform-specific fixups */ /* * Non-generic VIA/NVIDIA specific @@ -1345,6 +1346,12 @@ static int hdmi_find_pcm_slot(struct hdmi_spec *spec, { int i; + /* on the new machines, try to assign the pcm slot dynamically, + * not use the preferred fixed map (legacy way) anymore. + */ + if (spec->dyn_pcm_no_legacy) + goto last_try; + /* * generic_hdmi_build_pcms() may allocate extra PCMs on some * platforms (with maximum of 'num_nids + dev_num - 1') @@ -1374,6 +1381,7 @@ static int hdmi_find_pcm_slot(struct hdmi_spec *spec, return i; } + last_try: /* the last try; check the empty slots in pins */ for (i = 0; i < spec->num_nids; i++) { if (!test_bit(i, &spec->pcm_bitmap)) @@ -2987,8 +2995,16 @@ static int patch_i915_tgl_hdmi(struct hda_codec *codec) * the index indicate the port number. */ static const int map[] = {0x4, 0x6, 0x8, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf}; + int ret; - return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); + ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); + if (!ret) { + struct hdmi_spec *spec = codec->spec; + + spec->dyn_pcm_no_legacy = true; + } + + return ret; } /* Intel Baytrail and Braswell; with eld notifier */ -- 2.25.1

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[PATCH] ALSA: hda/realtek: Add quirk for Clevo NH55RZQ
by Werner Sembach 02 Mar '21

02 Mar '21

From: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> ALSA: hda/realtek: Add quirk for Clevo NH55RZQ This applies a SND_PCI_QUIRK(...) to the Clevo NH55RZQ barebone. This fixes the issue of the device not recognizing a pluged in microphone. The device has both, a microphone only jack, and a speaker + microphone combo jack. The combo jack already works. The microphone-only jack does not recognize when a device is pluged in without this patch. Signed-off-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Co-developed-by: Werner Sembach <wse(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> --- Hi, this is my first ever submitted kernel patch, feel free to criticise me if I made an error or missed a best practise bullet point. Also: I'm unsure if this would fit the requirements for stable(a)vger.kernel.org and/or trivial(a)kernel.org, but I think not (correct me if I'm wrong). Kind regards Werner Sembach >From 2835edd753fd19c72a644dccb7e941cfc0ecdf8e Mon Sep 17 00:00:00 2001 From: Werner Sembach <wse(a)tuxedocomputers.com> Date: Fri, 26 Feb 2021 13:50:15 +0100 Subject: [PATCH] Fix device detection on microphone jack of Clevo NH55RZQ --- 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 290645516313..8014e80d72c3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8089,6 +8089,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8551, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[5|7][0-9]RZ[Q]", ALC269_FIXUP_DMIC), SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.25.1

2 2

[PATCH] ASoC: simple-card: Add dummy dai support simple sound card
by Shengjiu Wang 02 Mar '21

02 Mar '21

If sound card doesn't need specific codec device, just dummy codec is enough, then we can link the dummy component directly. In this case, user needs to specify below setting in devicetree. Previously the sound-dai is a node of codec, now we check if it is zero before parsing the node, zero means dummy component is specified. simple-audio-card,codec { sound-dai = <0>; }; Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/generic/simple-card.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index c262281bc64d..8b0cabd1bad1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -34,10 +34,19 @@ static int asoc_simple_parse_dai(struct device_node *node, { struct of_phandle_args args; int ret; + u32 val; if (!node) return 0; + ret = of_property_read_u32(node, DAI, &val); + if (val == 0) { + dlc->of_node = NULL; + dlc->dai_name = "snd-soc-dummy-dai"; + dlc->name = "snd-soc-dummy"; + return 0; + } + /* * Get node via "sound-dai = <&phandle port>" * it will be used as xxx_of_node on soc_bind_dai_link() -- 2.27.0

3 3

[RFC PATCH 0/5] Flexible sysclk/pll configuration
by Sameer Pujar 02 Mar '21

02 Mar '21

This series attempts to add flexible system clock (sysclk) and pll configuration for the DAI controller from simple card or audio graph card. Sysclk configuration ==================== Motivation: ----------- The problem this tries to address is, some Codecs may have multiple clocks and multiple sources for its sysclk. For example sysclk may depend on MCLK provided by SoC or it may used an internal pll. The simple card or audio graph card driver is generic and can be used on multiple platforms. So if a platform using this driver needs a generic configuration, then the Codec specific stuff cannot be hardcoded in the driver. Hence it would be better if the info is driven from DT, especially from a Codec DAI or endpoint subnode. Solution: --------- "system-clock-id" and "system-clock-source" DT properties can be exposed for a flexible sysclk configuration. These properties go into a Codec DAI or endpoint subnode. Please note that other sysclk related properties like "system-clock-frequency" and "system-clock-direction-out" are already present. Currently snd_soc_dai_set_sysclk() does not have an argument for source. This series adds this additional field and updates all the required drivers. This is needed because ideally sysclk "clk_id" and "source" fields are different. Some drivers are making use of "clk_id" field to program source information. **May be this needs to be corrected going ahead** Pll configuration ================= Motivation: ----------- Similar to sysclk configuration, pll configuration also needs little more flexibility. If a Codec has internal pll and it can be sourced from multiple clocks a specific configuration may be required for a given platform. Some of the cases are: - MCLK (supplied by SoC) --> Codec pll -> Codec sysclk - SoC I2S bit clock (BCLK) --> Codec pll -> Codec sysclk Solution: --------- Approach is similar to what is done for extending sysclk flexibility. Following DT properties are added to address the same. "pll-id" "pll-source" "pll-input-reference" "pll-output-reference" "pll-input-frequency" "pll-output-frequency" All these are optional properties. Simple card or audio graph card drivers can use above info and call snd_soc_dai_set_pll() for necessary configuration. Sameer Pujar (5): ASoC: soc-component: Fix return value of snd_soc_component_set_pll() ASoC: soc-dai: Add sysclk source to snd_soc_dai_set_sysclk() ASoC: audio-graph-card: Add bindings for sysclk and pll ASoC: simple-card-utils: Parse sysclk id and source ASoC: simple-card-utils: Support pll configuration .../bindings/sound/audio-graph-port.yaml | 71 +++++++++ include/sound/simple_card_utils.h | 35 +++++ include/sound/soc-dai.h | 4 +- sound/soc/amd/acp-da7219-max98357a.c | 2 +- sound/soc/amd/acp-rt5645.c | 2 +- sound/soc/amd/acp3x-rt5682-max9836.c | 4 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 2 +- sound/soc/atmel/mikroe-proto.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-i2s.c | 2 +- sound/soc/cirrus/snappercl15.c | 4 +- sound/soc/codecs/88pm860x-codec.c | 2 +- sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adau17x1.c | 2 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/ak4613.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/alc5623.c | 2 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/arizona.c | 2 +- sound/soc/codecs/cpcap.c | 4 +- sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/cs35l34.c | 2 +- sound/soc/codecs/cs35l35.c | 2 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs4265.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/cs42l42.c | 2 +- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/cs42l56.c | 2 +- sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/cs42xx8.c | 2 +- sound/soc/codecs/cs43130.c | 2 +- sound/soc/codecs/cs53l30.c | 2 +- sound/soc/codecs/cx2072x.c | 2 +- sound/soc/codecs/da7210.c | 2 +- sound/soc/codecs/da7218.c | 2 +- sound/soc/codecs/da7219.c | 2 +- sound/soc/codecs/da732x.c | 2 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/es7134.c | 2 +- sound/soc/codecs/es7241.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/es8328.c | 2 +- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/madera.c | 3 +- sound/soc/codecs/max98088.c | 3 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max98095.c | 2 +- sound/soc/codecs/max98390.c | 2 +- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/max98925.c | 2 +- sound/soc/codecs/max98927.c | 2 +- sound/soc/codecs/mc13783.c | 6 +- sound/soc/codecs/ml26124.c | 2 +- sound/soc/codecs/nau8810.c | 2 +- sound/soc/codecs/nau8822.c | 2 +- sound/soc/codecs/pcm186x.c | 2 +- sound/soc/codecs/pcm3060.c | 2 +- sound/soc/codecs/pcm3168a.c | 2 +- sound/soc/codecs/rt1011.c | 2 +- sound/soc/codecs/rt1305.c | 2 +- sound/soc/codecs/rt274.c | 2 +- sound/soc/codecs/rt286.c | 2 +- sound/soc/codecs/rt298.c | 2 +- sound/soc/codecs/rt5514.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5660.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/sta32x.c | 2 +- sound/soc/codecs/sta350.c | 2 +- sound/soc/codecs/sti-sas.c | 2 +- sound/soc/codecs/tas2552.c | 2 +- sound/soc/codecs/tas5086.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/tscs454.c | 2 +- sound/soc/codecs/twl4030.c | 4 +- sound/soc/codecs/twl6040.c | 2 +- sound/soc/codecs/uda1334.c | 2 +- sound/soc/codecs/uda134x.c | 2 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8524.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8776.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/fsl/eukrea-tlv320.c | 4 +- sound/soc/fsl/fsl-asoc-card.c | 8 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_micfil.c | 2 +- sound/soc/fsl/fsl_sai.c | 8 +- sound/soc/fsl/imx-audmix.c | 2 +- sound/soc/fsl/imx-hdmi.c | 2 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- sound/soc/generic/simple-card-utils.c | 173 +++++++++++++++++++-- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 4 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 4 +- sound/soc/intel/boards/bytcr_rt5651.c | 4 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 4 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 2 +- sound/soc/intel/boards/sof_realtek_common.c | 2 +- sound/soc/intel/boards/sof_rt5682.c | 4 +- sound/soc/intel/boards/sof_sdw_rt1308.c | 2 +- sound/soc/intel/boards/sof_wm8804.c | 2 +- sound/soc/jz4740/jz4740-i2s.c | 2 +- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 4 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 4 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 8 +- sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 2 +- sound/soc/mediatek/mt8183/mt8183-dai-tdm.c | 2 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 6 +- sound/soc/mediatek/mt8192/mt8192-dai-i2s.c | 2 +- sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 2 +- .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 8 +- sound/soc/meson/aiu-encoder-i2s.c | 2 +- sound/soc/meson/axg-tdm-interface.c | 2 +- sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/mxs/mxs-saif.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 4 +- sound/soc/pxa/brownstone.c | 4 +- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/imote2.c | 4 +- sound/soc/pxa/magician.c | 4 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/z2.c | 4 +- sound/soc/pxa/zylonite.c | 2 +- sound/soc/qcom/apq8096.c | 2 +- sound/soc/qcom/lpass-cpu.c | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- sound/soc/qcom/sc7180.c | 6 +- sound/soc/qcom/sdm845.c | 22 +-- sound/soc/qcom/sm8250.c | 2 +- sound/soc/qcom/storm.c | 2 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 4 +- sound/soc/rockchip/rk3399_gru_sound.c | 14 +- sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_max98090.c | 4 +- sound/soc/rockchip/rockchip_rt5645.c | 4 +- sound/soc/samsung/aries_wm8994.c | 6 +- sound/soc/samsung/arndale.c | 8 +- sound/soc/samsung/bells.c | 6 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 6 +- sound/soc/samsung/jive_wm8750.c | 2 +- sound/soc/samsung/littlemill.c | 14 +- sound/soc/samsung/midas_wm1811.c | 10 +- sound/soc/samsung/neo1973_wm8753.c | 4 +- sound/soc/samsung/odroid.c | 2 +- sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c24xx-i2s.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 4 +- sound/soc/samsung/s3c24xx_uda134x.c | 4 +- sound/soc/samsung/smartq_wm8987.c | 6 +- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 4 +- sound/soc/samsung/snow.c | 2 +- sound/soc/samsung/spdif.c | 2 +- sound/soc/samsung/speyside.c | 8 +- sound/soc/samsung/tm2_wm5110.c | 6 +- sound/soc/samsung/tobermory.c | 6 +- sound/soc/sh/migor.c | 4 +- sound/soc/sh/siu_dai.c | 2 +- sound/soc/sh/ssi.c | 2 +- sound/soc/soc-component.c | 2 +- sound/soc/soc-dai.c | 9 +- sound/soc/sti/uniperif_player.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/stm/stm32_i2s.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/sunxi/sun4i-i2s.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 2 +- sound/soc/tegra/trimslice.c | 2 +- sound/soc/ti/davinci-evm.c | 4 +- sound/soc/ti/davinci-mcasp.c | 2 +- sound/soc/ti/j721e-evm.c | 8 +- sound/soc/ti/n810.c | 2 +- sound/soc/ti/omap-abe-twl6040.c | 6 +- sound/soc/ti/omap-dmic.c | 2 +- sound/soc/ti/omap-mcbsp.c | 2 +- sound/soc/ti/omap3pandora.c | 4 +- sound/soc/ti/rx51.c | 2 +- sound/soc/uniphier/aio-cpu.c | 4 +- sound/soc/ux500/ux500_msp_dai.c | 2 +- 271 files changed, 639 insertions(+), 389 deletions(-) -- 2.7.4

3 9

[PATCH v6 0/9] ALSA: add virtio sound driver
by Anton Yakovlev 02 Mar '21

02 Mar '21

This series implements a driver part of the virtio sound device specification v8 [1]. The driver supports PCM playback and capture substreams, jack and channel map controls. A message-based transport is used to write/read PCM frames to/from a device. As a device part was used OpenSynergy proprietary implementation. v6 changes: - Code readability improvements suggested by Takashi. - Added additional comments about PCM message lifetime. - Temporarily removed DEVICE_NEEDS_RESET device status handling. Some kind of assistance from the virtio bus/level is required. In turn, this requires a separate discussion of the possibility of implementing such handling for virtio devices whose configuration may change (which may require re-probing, like in case of the sound device). [1] https://lists.oasis-open.org/archives/virtio-dev/202003/msg00185.html Anton Yakovlev (9): uapi: virtio_ids: add a sound device type ID from OASIS spec ALSA: virtio: add virtio sound driver ALSA: virtio: handling control messages ALSA: virtio: build PCM devices and substream hardware descriptors ALSA: virtio: handling control and I/O messages for the PCM device ALSA: virtio: PCM substream operators ALSA: virtio: introduce jack support ALSA: virtio: introduce PCM channel map support ALSA: virtio: introduce device suspend/resume support MAINTAINERS | 9 + include/uapi/linux/virtio_ids.h | 1 + include/uapi/linux/virtio_snd.h | 334 +++++++++++++++++++++ sound/Kconfig | 2 + sound/Makefile | 3 +- sound/virtio/Kconfig | 10 + sound/virtio/Makefile | 13 + sound/virtio/virtio_card.c | 410 ++++++++++++++++++++++++++ sound/virtio/virtio_card.h | 111 +++++++ sound/virtio/virtio_chmap.c | 219 ++++++++++++++ sound/virtio/virtio_ctl_msg.c | 310 ++++++++++++++++++++ sound/virtio/virtio_ctl_msg.h | 78 +++++ sound/virtio/virtio_jack.c | 233 +++++++++++++++ sound/virtio/virtio_pcm.c | 498 ++++++++++++++++++++++++++++++++ sound/virtio/virtio_pcm.h | 118 ++++++++ sound/virtio/virtio_pcm_msg.c | 417 ++++++++++++++++++++++++++ sound/virtio/virtio_pcm_ops.c | 478 ++++++++++++++++++++++++++++++ 17 files changed, 3243 insertions(+), 1 deletion(-) create mode 100644 include/uapi/linux/virtio_snd.h create mode 100644 sound/virtio/Kconfig create mode 100644 sound/virtio/Makefile create mode 100644 sound/virtio/virtio_card.c create mode 100644 sound/virtio/virtio_card.h create mode 100644 sound/virtio/virtio_chmap.c create mode 100644 sound/virtio/virtio_ctl_msg.c create mode 100644 sound/virtio/virtio_ctl_msg.h create mode 100644 sound/virtio/virtio_jack.c create mode 100644 sound/virtio/virtio_pcm.c create mode 100644 sound/virtio/virtio_pcm.h create mode 100644 sound/virtio/virtio_pcm_msg.c create mode 100644 sound/virtio/virtio_pcm_ops.c -- 2.30.1

2 31

[PATCH] ALSA: hda: ignore invalid NHLT table
by Mark Pearson 02 Mar '21

02 Mar '21

On some Lenovo systems if the microphone is disabled in the BIOS only the NHLT table header is created, with no data. This means the endpoints field is not correctly set to zero - leading to an unintialised variable and hence invalid descriptors are parsed leading to page faults. The Lenovo firmware team is addressing this, but adding a check preventing invalid tables being parsed is worthwhile. Tested on a Lenovo T14. Tested-by: Philipp Leskovitz <philipp.leskovitz(a)secunet.com> Reported-by: Philipp Leskovitz <philipp.leskovitz(a)secunet.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> --- sound/hda/intel-nhlt.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 059aaf04f..0889f2cc5 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -37,6 +37,11 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) if (!nhlt) return 0; + if (nhlt->header.length <= sizeof(struct acpi_table_header)) { + dev_warn(dev, "Invalid DMIC description table\n"); + return 0; + } + epnt = (struct nhlt_endpoint *)nhlt->desc; for (j = 0; j < nhlt->endpoint_count; j++) { -- 2.29.2

1 0

[PATCH] ASoC: codec: Omit superfluous error message in jz4760_codec_probe()
by Tang Bin 02 Mar '21

02 Mar '21

The function devm_platform_ioremap_resource has already contained error message, so remove the redundant dev_err here. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/codecs/jz4760.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/jz4760.c b/sound/soc/codecs/jz4760.c index e8f28ccc145a..c2d8a107f159 100644 --- a/sound/soc/codecs/jz4760.c +++ b/sound/soc/codecs/jz4760.c @@ -843,7 +843,6 @@ static int jz4760_codec_probe(struct platform_device *pdev) codec->base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(codec->base)) { ret = PTR_ERR(codec->base); - dev_err(dev, "Failed to ioremap mmio memory: %d\n", ret); return ret; } -- 2.18.2

4 3

[PATCH] ASoC: wm8960: Remove bitclk relax condition
by Shengjiu Wang 02 Mar '21

02 Mar '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> Using a higher bitclk then expected doesn't always work. Here is an example: aplay -Dhw:0,0 -d 5 -r 48000 -f S24_LE -c 2 audio48k24b2c.wav In this case, the required bitclk is 48000 * 24 * 2 = 2304000 but the closest bitclk that can be derived is 3072000. Since the clock is faster than expected, it will start to send bytes from the next channel so the sound will be corrupted. Fixes: 82bab88910ee ("ASoC: codec: wm8960: Relax bit clock computation when using PLL") Fixes: 3c01b9ee2ab9 ("ASoC: codec: wm8960: Relax bit clock computation") Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8960.c | 29 +++-------------------------- 1 file changed, 3 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index df351519a3a6..368bec1dfbdb 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -608,10 +608,6 @@ static const int bclk_divs[] = { * - lrclk = sysclk / dac_divs * - 10 * bclk = sysclk / bclk_divs * - * If we cannot find an exact match for (sysclk, lrclk, bclk) - * triplet, we relax the bclk such that bclk is chosen as the - * closest available frequency greater than expected bclk. - * * @wm8960: codec private data * @mclk: MCLK used to derive sysclk * @sysclk_idx: sysclk_divs index for found sysclk @@ -629,7 +625,7 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, { int sysclk, bclk, lrclk; int i, j, k; - int diff, closest = mclk; + int diff; /* marker for no match */ *bclk_idx = -1; @@ -653,12 +649,6 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, *bclk_idx = k; break; } - if (diff > 0 && closest > diff) { - *sysclk_idx = i; - *dac_idx = j; - *bclk_idx = k; - closest = diff; - } } if (k != ARRAY_SIZE(bclk_divs)) break; @@ -676,10 +666,6 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, * - freq_out = sysclk * sysclk_divs * - 10 * sysclk = bclk * bclk_divs * - * If we cannot find an exact match for (sysclk, lrclk, bclk) - * triplet, we relax the bclk such that bclk is chosen as the - * closest available frequency greater than expected bclk. - * * @component: component structure * @freq_in: input frequency used to derive freq out via PLL * @sysclk_idx: sysclk_divs index for found sysclk @@ -697,14 +683,12 @@ int wm8960_configure_pll(struct snd_soc_component *component, int freq_in, { struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component); int sysclk, bclk, lrclk, freq_out; - int diff, closest, best_freq_out; + int diff; int i, j, k; bclk = wm8960->bclk; lrclk = wm8960->lrclk; - closest = freq_in; - best_freq_out = -EINVAL; *sysclk_idx = *dac_idx = *bclk_idx = -1; for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { @@ -725,18 +709,11 @@ int wm8960_configure_pll(struct snd_soc_component *component, int freq_in, *bclk_idx = k; return freq_out; } - if (diff > 0 && closest > diff) { - *sysclk_idx = i; - *dac_idx = j; - *bclk_idx = k; - closest = diff; - best_freq_out = freq_out; - } } } } - return best_freq_out; + return -EINVAL; } static int wm8960_configure_clocking(struct snd_soc_component *component) { -- 2.27.0

2 1

[PATCH] ALSA: hda/realtek: Add quirk for Intel NUC 10
by Werner Sembach 02 Mar '21

02 Mar '21

From: Werner Sembach <wse(a)tuxedocomputers.com> ALSA: hda/realtek: Add quirk for Intel NUC 10 This adds a new SND_PCI_QUIRK(...) and applies it to the Intel NUC 10 devices. This fixes the issue of the devices not having audio input and output on the headset jack because the kernel does not recognize when something is plugged in. The new quirk was inspired by the quirk for the Intel NUC 8 devices, but it turned out that the NUC 10 uses another pin. This information was acquired by black box testing likely pins. Co-developed-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> --- Hi, this is my second ever submitted kernel patch with the first one send just some minutes ago, feel free to criticise me if I made an error or missed a best practice bullet point. Also: I'm unsure if this would fit the requirements for stable(a)vger.kernel.org and/or trivial(a)kernel.org, but I think not (correct me if I'm wrong). Kind regards Werner Sembach >From d281364b8ca6c054a0e5ce20caa599bf7d08160d Mon Sep 17 00:00:00 2001 From: Werner Sembach <wse(a)tuxedocomputers.com> Date: Fri, 26 Feb 2021 13:54:30 +0100 Subject: [PATCH] Fix Intel NUC10 no output and input on headset jack --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290645516313..c14d624dbaf1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6362,6 +6362,7 @@ enum { ALC269_FIXUP_LEMOTE_A1802, ALC269_FIXUP_LEMOTE_A190X, ALC256_FIXUP_INTEL_NUC8_RUGGED, + ALC256_FIXUP_INTEL_NUC10, ALC255_FIXUP_XIAOMI_HEADSET_MIC, ALC274_FIXUP_HP_MIC, ALC274_FIXUP_HP_HEADSET_MIC, @@ -7744,6 +7745,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC256_FIXUP_INTEL_NUC10] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, [ALC255_FIXUP_XIAOMI_HEADSET_MIC] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -8183,6 +8193,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), + SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), #if 0 /* Below is a quirk table taken from the old code. -- 2.25.1

1 0

[PATCH] ALSA: hda: intel-nhlt: verify config type
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

Multiple bug reports report issues with the SOF and SST drivers when dealing with single microphone cases. We currently read the DMIC array information unconditionally but we don't check that the configuration type is actually a mic array. When the DMIC link does not rely on a mic array configuration, the recommendation is to check the format information to infer the maximum number of channels, and map this to the number of microphones. This leaves a potential for a mismatch between actual microphones available in hardware and what the ACPI table contains, but we have no other source of information. Note that single microphone configurations can alternatively be handled with a 'mic array' configuration along with a 'vendor-defined' geometry. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201251 BugLink: https://github.com/thesofproject/linux/issues/2725 Fixes: 7a33ea70e1868 ('ALSA: hda: intel-nhlt: handle NHLT VENDOR_DEFINED DMIC geometry') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/intel-nhlt.h | 5 ++++ sound/hda/intel-nhlt.c | 54 +++++++++++++++++++++++++++++++------- 2 files changed, 50 insertions(+), 9 deletions(-) diff --git a/include/sound/intel-nhlt.h b/include/sound/intel-nhlt.h index 743c2f442280..d0574805865f 100644 --- a/include/sound/intel-nhlt.h +++ b/include/sound/intel-nhlt.h @@ -112,6 +112,11 @@ struct nhlt_vendor_dmic_array_config { /* TODO add vendor mic config */ } __packed; +enum { + NHLT_CONFIG_TYPE_GENERIC = 0, + NHLT_CONFIG_TYPE_MIC_ARRAY = 1 +}; + enum { NHLT_MIC_ARRAY_2CH_SMALL = 0xa, NHLT_MIC_ARRAY_2CH_BIG = 0xb, diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 059aaf04f536..d053beccfaec 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -31,18 +31,44 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) struct nhlt_endpoint *epnt; struct nhlt_dmic_array_config *cfg; struct nhlt_vendor_dmic_array_config *cfg_vendor; + struct nhlt_fmt *fmt_configs; unsigned int dmic_geo = 0; - u8 j; + u16 max_ch = 0; + u8 i, j; if (!nhlt) return 0; - epnt = (struct nhlt_endpoint *)nhlt->desc; + for (j = 0, epnt = nhlt->desc; j < nhlt->endpoint_count; j++, + epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length)) { - for (j = 0; j < nhlt->endpoint_count; j++) { - if (epnt->linktype == NHLT_LINK_DMIC) { - cfg = (struct nhlt_dmic_array_config *) - (epnt->config.caps); + if (epnt->linktype != NHLT_LINK_DMIC) + continue; + + cfg = (struct nhlt_dmic_array_config *)(epnt->config.caps); + fmt_configs = (struct nhlt_fmt *)(epnt->config.caps + epnt->config.size); + + /* find max number of channels based on format_configuration */ + if (fmt_configs->fmt_count) { + dev_dbg(dev, "%s: found %d format definitions\n", + __func__, fmt_configs->fmt_count); + + for (i = 0; i < fmt_configs->fmt_count; i++) { + struct wav_fmt_ext *fmt_ext; + + fmt_ext = &fmt_configs->fmt_config[i].fmt_ext; + + if (fmt_ext->fmt.channels > max_ch) + max_ch = fmt_ext->fmt.channels; + } + dev_dbg(dev, "%s: max channels found %d\n", __func__, max_ch); + } else { + dev_dbg(dev, "%s: No format information found\n", __func__); + } + + if (cfg->device_config.config_type != NHLT_CONFIG_TYPE_MIC_ARRAY) { + dmic_geo = max_ch; + } else { switch (cfg->array_type) { case NHLT_MIC_ARRAY_2CH_SMALL: case NHLT_MIC_ARRAY_2CH_BIG: @@ -59,13 +85,23 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) dmic_geo = cfg_vendor->nb_mics; break; default: - dev_warn(dev, "undefined DMIC array_type 0x%0x\n", - cfg->array_type); + dev_warn(dev, "%s: undefined DMIC array_type 0x%0x\n", + __func__, cfg->array_type); + } + + if (dmic_geo > 0) { + dev_dbg(dev, "%s: Array with %d dmics\n", __func__, dmic_geo); + } + if (max_ch > dmic_geo) { + dev_dbg(dev, "%s: max channels %d exceed dmic number %d\n", + __func__, max_ch, dmic_geo); } } - epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); } + dev_dbg(dev, "%s: dmic number %d max_ch %d\n", + __func__, dmic_geo, max_ch); + return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); -- 2.25.1

2 1

[PATCH v2] ALSA: hda: fix kernel-doc warnings
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

v5.12-rc1 flags new warnings with make W=1, fix missing or broken function descriptors. sound/pci/hda/hda_codec.c:3492: warning: expecting prototype for snd_hda_input_mux_info_info(). Prototype was for snd_hda_input_mux_info() instead sound/pci/hda/hda_codec.c:3521: warning: expecting prototype for snd_hda_input_mux_info_put(). Prototype was for snd_hda_input_mux_put() instead sound/pci/hda/hda_codec.c:3958: warning: expecting prototype for _snd_hda_pin_ctl(). Prototype was for _snd_hda_set_pin_ctl() instead sound/pci/hda/hda_jack.c:223: warning: expecting prototype for snd_hda_set_dirty_all(). Prototype was for snd_hda_jack_set_dirty_all() instead sound/pci/hda/hda_jack.c:309: warning: expecting prototype for snd_hda_jack_detect_enable_mst(). Prototype was for snd_hda_jack_detect_enable_callback_mst() instead sound/pci/hda/hda_generic.c:3933: warning: expecting prototype for snd_dha_gen_add_mute_led_cdev(). Prototype was for snd_hda_gen_add_mute_led_cdev() instead sound/pci/hda/hda_generic.c:4093: warning: expecting prototype for snd_dha_gen_add_micmute_led_cdev(). Prototype was for snd_hda_gen_add_micmute_led_cdev() instead sound/pci/hda/patch_ca0132.c:2357: warning: expecting prototype for Prepare and send the SCP message to DSP(). Prototype was for dspio_scp() instead sound/pci/hda/patch_ca0132.c:2883: warning: expecting prototype for Allocate router ports(). Prototype was for dsp_allocate_router_ports() instead sound/pci/hda/patch_ca0132.c:3202: warning: expecting prototype for Write a block of data into DSP code or data RAM using pre(). Prototype was for dspxfr_one_seg() instead sound/pci/hda/patch_ca0132.c:3397: warning: expecting prototype for data overlay to DSP memories(). Prototype was for dspxfr_image() instead sound/hda/hdac_regmap.c:393: warning: expecting prototype for snd_hdac_regmap_init(). Prototype was for snd_hdac_regmap_exit() instead sound/hda/ext/hdac_ext_controller.c:142: warning: expecting prototype for snd_hdac_ext_bus_get_link_index(). Prototype was for snd_hdac_ext_bus_get_link() instead sound/hda/ext/hdac_ext_stream.c:140: warning: expecting prototype for snd_hdac_ext_linkstream_start(). Prototype was for snd_hdac_ext_link_stream_start() instead Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/ext/hdac_ext_controller.c | 2 +- sound/hda/ext/hdac_ext_stream.c | 2 +- sound/hda/hdac_regmap.c | 2 +- sound/pci/hda/hda_codec.c | 6 +++--- sound/pci/hda/hda_generic.c | 4 ++-- sound/pci/hda/hda_jack.c | 4 ++-- sound/pci/hda/patch_ca0132.c | 9 ++++----- 7 files changed, 14 insertions(+), 15 deletions(-) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index a9bd39b93697..b2df7b4f9227 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -133,7 +133,7 @@ void snd_hdac_link_free_all(struct hdac_bus *bus) EXPORT_SYMBOL_GPL(snd_hdac_link_free_all); /** - * snd_hdac_ext_bus_get_link_index - get link based on codec name + * snd_hdac_ext_bus_get_link - get link based on codec name * @bus: the pointer to HDAC bus object * @codec_name: codec name */ diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index c4d54a838773..0c005d67fa89 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -133,7 +133,7 @@ void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_decouple); /** - * snd_hdac_ext_linkstream_start - start a stream + * snd_hdac_ext_link_stream_start - start a stream * @stream: HD-audio ext core stream to start */ void snd_hdac_ext_link_stream_start(struct hdac_ext_stream *stream) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index d75f31eb9d78..fe3587547cfe 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -386,7 +386,7 @@ int snd_hdac_regmap_init(struct hdac_device *codec) EXPORT_SYMBOL_GPL(snd_hdac_regmap_init); /** - * snd_hdac_regmap_init - Release the regmap from HDA codec + * snd_hdac_regmap_exit - Release the regmap from HDA codec * @codec: the codec object */ void snd_hdac_regmap_exit(struct hdac_device *codec) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9b755062d841..2026f1ccaf5a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3483,7 +3483,7 @@ EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power); */ /** - * snd_hda_input_mux_info_info - Info callback helper for the input-mux enum + * snd_hda_input_mux_info - Info callback helper for the input-mux enum * @imux: imux helper object * @uinfo: pointer to get/store the data */ @@ -3506,7 +3506,7 @@ int snd_hda_input_mux_info(const struct hda_input_mux *imux, EXPORT_SYMBOL_GPL(snd_hda_input_mux_info); /** - * snd_hda_input_mux_info_put - Put callback helper for the input-mux enum + * snd_hda_input_mux_put - Put callback helper for the input-mux enum * @codec: the HDA codec * @imux: imux helper object * @ucontrol: pointer to get/store the data @@ -3941,7 +3941,7 @@ unsigned int snd_hda_correct_pin_ctl(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_correct_pin_ctl); /** - * _snd_hda_pin_ctl - Helper to set pin ctl value + * _snd_hda_set_pin_ctl - Helper to set pin ctl value * @codec: the HDA codec * @pin: referred pin NID * @val: pin control value to set diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 5e40944e7342..8b7c5508f368 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3923,7 +3923,7 @@ static void vmaster_update_mute_led(void *private_data, int enabled) } /** - * snd_dha_gen_add_mute_led_cdev - Create a LED classdev and enable as vmaster mute LED + * snd_hda_gen_add_mute_led_cdev - Create a LED classdev and enable as vmaster mute LED * @codec: the HDA codec * @callback: the callback for LED classdev brightness_set_blocking */ @@ -4074,7 +4074,7 @@ static int add_micmute_led_hook(struct hda_codec *codec) } /** - * snd_dha_gen_add_micmute_led_cdev - Create a LED classdev and enable as mic-mute LED + * snd_hda_gen_add_micmute_led_cdev - Create a LED classdev and enable as mic-mute LED * @codec: the HDA codec * @callback: the callback for LED classdev brightness_set_blocking * diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index b8b568046592..ac00866d8032 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -213,7 +213,7 @@ static void jack_detect_update(struct hda_codec *codec, } /** - * snd_hda_set_dirty_all - Mark all the cached as dirty + * snd_hda_jack_set_dirty_all - Mark all the cached as dirty * @codec: the HDA codec * * This function sets the dirty flag to all entries of jack table. @@ -293,7 +293,7 @@ find_callback_from_list(struct hda_jack_tbl *jack, } /** - * snd_hda_jack_detect_enable_mst - enable the jack-detection + * snd_hda_jack_detect_enable_callback_mst - enable the jack-detection * @codec: the HDA codec * @nid: pin NID to enable * @func: callback function to register diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 7e62aed172a9..c966f49fa942 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2338,7 +2338,7 @@ static int dspio_send_scp_message(struct hda_codec *codec, } /** - * Prepare and send the SCP message to DSP + * dspio_scp - Prepare and send the SCP message to DSP * @codec: the HDA codec * @mod_id: ID of the DSP module to send the command * @src_id: ID of the source @@ -2865,7 +2865,7 @@ static int dsp_dma_stop(struct hda_codec *codec, } /** - * Allocate router ports + * dsp_allocate_router_ports - Allocate router ports * * @codec: the HDA codec * @num_chans: number of channels in the stream @@ -3178,8 +3178,7 @@ static int dspxfr_hci_write(struct hda_codec *codec, } /** - * Write a block of data into DSP code or data RAM using pre-allocated - * DMA engine. + * dspxfr_one_seg - Write a block of data into DSP code or data RAM using pre-allocated DMA engine. * * @codec: the HDA codec * @fls: pointer to a fast load image @@ -3376,7 +3375,7 @@ static int dspxfr_one_seg(struct hda_codec *codec, } /** - * Write the entire DSP image of a DSP code/data overlay to DSP memories + * dspxfr_image - Write the entire DSP image of a DSP code/data overlay to DSP memories * * @codec: the HDA codec * @fls_data: pointer to a fast load image -- 2.25.1

2 1

[PATCH 0/4]: ALSA: usb-audio: Add Pioneer DJM-850 support
by Nicolas MURE 02 Mar '21

02 Mar '21

Hello, This patches serie adds support for the Pioneer DJM-850 mixer. The mixer has a 24bits (little endian) sound card, which features 4 stereo channels for capture and 4 stereo channels for playback. The mixer also has controls to specify the capture source for each channels, and also a control to set the sample rate (44.1kHz, 48kHz, 96kHz). Thank you :) Signed-off-by: Nicolas MURE <nicolas.mure2019(a)gmail.com>

2 11

[PATCH 0/4] drop unneeded snd_soc_dai_set_drvdata
by Julia Lawall 02 Mar '21

02 Mar '21

snd_soc_dai_set_drvdata is not needed when the set data comes from snd_soc_dai_get_drvdata or dev_get_drvdata. --- sound/soc/fsl/fsl_micfil.c | 2 -- sound/soc/fsl/fsl_sai.c | 2 -- sound/soc/fsl/fsl_xcvr.c | 1 - sound/soc/mxs/mxs-saif.c | 10 ---------- sound/soc/pxa/mmp-sspa.c | 1 - sound/soc/sunxi/sun4i-i2s.c | 2 -- 6 files changed, 18 deletions(-)

3 6

[PATCH][next] ASoC: codecs: lpass-rx-macro: remove redundant initialization of variable hph_pwr_mode
by Colin King 02 Mar '21

02 Mar '21

From: Colin Ian King <colin.king(a)canonical.com> The variable hph_pwr_mode is being initialized with a value that is never read and it is being updated later with a new value. The initialization is redundant and can be removed. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/lpass-rx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 8c04b3b2c907..76909c50d7b9 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -2038,7 +2038,7 @@ static int rx_macro_load_compander_coeff(struct snd_soc_component *component, { u16 comp_coeff_lsb_reg, comp_coeff_msb_reg; int i; - int hph_pwr_mode = HPH_LOHIFI; + int hph_pwr_mode; if (!rx->comp_enabled[comp]) return 0; -- 2.30.0

3 2

[PATCH 0/6] ASoC: core: remove cppcheck warnings
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

This is the first batch of cleanups to make cppcheck more usable, currently we have way too many warnings that drown real issues. Pierre-Louis Bossart (6): ASoC: soc-ops: remove useless assignment ASoC: soc-pcm: remove redundant assignment ASoC: soc-pcm: remove shadowing variable ASoC: soc-pcm: add error log ASoC: soc-topology: clarify expression ASoC: generic: simple-card-utils: remove useless assignment sound/soc/generic/simple-card-utils.c | 2 +- sound/soc/soc-ops.c | 2 +- sound/soc/soc-pcm.c | 4 ++-- sound/soc/soc-topology.c | 16 ++++++++-------- 4 files changed, 12 insertions(+), 12 deletions(-) -- 2.25.1

2 7

[PATCH 00/10] ASoC: qcom: remove cppcheck warnings
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

Second batch of cleanups for Qualcomm SOCs and codecs. The only functional change is the addition of a missing error check in the last patch. Pierre-Louis Bossart (10): ASoC: qcom: lpass-hdmi: remove useless return ASoC: qcom: lpass-platform: remove useless assignment ASoC: qcom: q6dsp-dai: clarify expression ASoC: qcom: q6afe: remove useless assignments ASoC: qcom: q6afe: align function prototype ASoC: qcom: q6asm: align function prototypes ASoC: wcd-clsh-v2: align function prototypes ASoC: wcd9335: clarify return value ASoC: wcd934x: remove useless return ASoC: lpass-wsa-macro: add missing test sound/soc/codecs/lpass-wsa-macro.c | 2 ++ sound/soc/codecs/wcd-clsh-v2.h | 6 +++--- sound/soc/codecs/wcd9335.c | 2 +- sound/soc/codecs/wcd934x.c | 2 -- sound/soc/qcom/lpass-hdmi.c | 4 ---- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- sound/soc/qcom/qdsp6/q6afe.c | 5 ++--- sound/soc/qcom/qdsp6/q6afe.h | 2 +- sound/soc/qcom/qdsp6/q6asm.h | 6 +++--- 10 files changed, 14 insertions(+), 19 deletions(-) -- 2.25.1

3 12

[PATCH 0/6] ASoC: samsung: remove cppcheck warnings
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

No functional changes except for patch 2 and 3 where missing error checks were added for consistency. Pierre-Louis Bossart (6): ASoC: samsung: i2s: remove unassigned variable ASoC: samsung: s3c24xx_simtec: add missing error check ASoC: samsung: smdk_wm8994: add missing return ASoC: samsung: snow: remove useless test ASoC: samsung: tm2_wm5110: check of_parse return value ASoC: samsung: tm2_wm5510: remove shadowing variable sound/soc/samsung/i2s.c | 3 +-- sound/soc/samsung/s3c24xx_simtec.c | 5 +++++ sound/soc/samsung/smdk_wm8994.c | 1 + sound/soc/samsung/snow.c | 5 +---- sound/soc/samsung/tm2_wm5110.c | 3 +-- 5 files changed, 9 insertions(+), 8 deletions(-) -- 2.25.1

4 21

[PATCH 0/8] ASoC: sh: remove cppcheck warnings
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

No functional changes except for patch 4 where a missing error check was added for consistency. Pierre-Louis Bossart (8): ASoC: sh: dma-sh7760: remove unused variable ASoC: sh: rcar: align function prototypes ASoC: sh: rcar: simplify return ASoC: sh: rcar: core: rename shadowing variables ASoC: sh: rcar: ctu: add missing error check ASoC: sh: rcar: ssi: remove redundant assignment ASoC: sh: siu_pcm: remove useless assignment ASoC: sh: siu_pcm: remove unused variable sound/soc/sh/dma-sh7760.c | 1 - sound/soc/sh/rcar/core.c | 14 +++++++------- sound/soc/sh/rcar/ctu.c | 2 ++ sound/soc/sh/rcar/rsnd.h | 6 +++--- sound/soc/sh/rcar/ssi.c | 1 - sound/soc/sh/siu_pcm.c | 6 +----- 6 files changed, 13 insertions(+), 17 deletions(-) -- 2.25.1

3 13

[PATCH 0/9] ASoC: fsl: remove cppcheck warnings
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

Nothing critical and no functional changes. The only change that needs attention if the 'fsl_ssi: remove unnecessary tests' patch, where variables are to zero, then tested to set register fields. Either the tests are indeed redundant or the entire programming sequence is incorrect. Pierre-Louis Bossart (9): ASoC: fsl: fsl_asrc: remove useless assignment ASoC: fsl: fsl_dma: remove unused variable ASoC: fsl: fsl_easrc: remove useless assignments ASoC: fsl: fsl_esai: clarify expression ASoC: fsl: fsl_ssi: remove unnecessary tests ASoC: fsl: imx-hdmi: remove unused structure members ASoC: fsl: mpc5200: signed parameter in snprintf format ASoC: fsl: mpc8610: remove useless assignment ASoC: fsl: p1022_ds: remove useless assignment sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_dma.c | 3 --- sound/soc/fsl/fsl_easrc.c | 6 +++--- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_ssi.c | 5 ++--- sound/soc/fsl/imx-hdmi.c | 4 ---- sound/soc/fsl/mpc5200_dma.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- 9 files changed, 10 insertions(+), 18 deletions(-) -- 2.25.1

2 10

[PATCH] ASoC: constify of_phandle_args in snd_soc_get_dai_name()
by Krzysztof Kozlowski 02 Mar '21

02 Mar '21

The pointer to of_phandle_args passed to snd_soc_get_dai_name() and of_xlate_dai_name() implementations is not modified. Since it is being used only to translate passed OF node to a DAI name, it should not be modified, so mark it as const for correctness and safer code. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- include/sound/soc-component.h | 4 ++-- include/sound/soc.h | 2 +- sound/soc/meson/aiu-acodec-ctrl.c | 2 +- sound/soc/meson/aiu-codec-ctrl.c | 2 +- sound/soc/meson/aiu.c | 4 ++-- sound/soc/meson/aiu.h | 2 +- sound/soc/qcom/lpass-cpu.c | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- sound/soc/soc-component.c | 2 +- sound/soc/soc-core.c | 2 +- 10 files changed, 12 insertions(+), 12 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5b47768222b7..7dc75b39287f 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -101,7 +101,7 @@ struct snd_soc_component_driver { /* DT */ int (*of_xlate_dai_name)(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name); int (*of_xlate_dai_id)(struct snd_soc_component *comment, struct device_node *endpoint); @@ -450,7 +450,7 @@ void snd_soc_component_remove(struct snd_soc_component *component); int snd_soc_component_of_xlate_dai_id(struct snd_soc_component *component, struct device_node *ep); int snd_soc_component_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name); int snd_soc_component_compr_open(struct snd_compr_stream *cstream); void snd_soc_component_compr_free(struct snd_compr_stream *cstream, diff --git a/include/sound/soc.h b/include/sound/soc.h index bd38015d6c6d..78609ab331c8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1219,7 +1219,7 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, struct device_node **bitclkmaster, struct device_node **framemaster); int snd_soc_get_dai_id(struct device_node *ep); -int snd_soc_get_dai_name(struct of_phandle_args *args, +int snd_soc_get_dai_name(const struct of_phandle_args *args, const char **dai_name); int snd_soc_of_get_dai_name(struct device_node *of_node, const char **dai_name); diff --git a/sound/soc/meson/aiu-acodec-ctrl.c b/sound/soc/meson/aiu-acodec-ctrl.c index 7078197e0cc5..27a6d3259c50 100644 --- a/sound/soc/meson/aiu-acodec-ctrl.c +++ b/sound/soc/meson/aiu-acodec-ctrl.c @@ -159,7 +159,7 @@ static const struct snd_kcontrol_new aiu_acodec_ctrl_controls[] = { }; static int aiu_acodec_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name) { return aiu_of_xlate_dai_name(component, args, dai_name, AIU_ACODEC); diff --git a/sound/soc/meson/aiu-codec-ctrl.c b/sound/soc/meson/aiu-codec-ctrl.c index 4b773d3e8b07..c3ea733fce91 100644 --- a/sound/soc/meson/aiu-codec-ctrl.c +++ b/sound/soc/meson/aiu-codec-ctrl.c @@ -125,7 +125,7 @@ static const struct snd_soc_dapm_route aiu_hdmi_ctrl_routes[] = { }; static int aiu_hdmi_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name) { return aiu_of_xlate_dai_name(component, args, dai_name, AIU_HDMI); diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index dc35ca79021c..ba15d5762b0b 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -42,7 +42,7 @@ static const struct snd_soc_dapm_route aiu_cpu_dapm_routes[] = { }; int aiu_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name, unsigned int component_id) { @@ -72,7 +72,7 @@ int aiu_of_xlate_dai_name(struct snd_soc_component *component, } static int aiu_cpu_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name) { return aiu_of_xlate_dai_name(component, args, dai_name, AIU_CPU); diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h index 87aa19ac4af3..393b6c2307e4 100644 --- a/sound/soc/meson/aiu.h +++ b/sound/soc/meson/aiu.h @@ -45,7 +45,7 @@ struct aiu { SNDRV_PCM_FMTBIT_S24_LE) int aiu_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name, unsigned int component_id); diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c642e5f8f28c..4762286b33fe 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -340,7 +340,7 @@ int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai) EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_probe); static int asoc_qcom_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name) { struct lpass_data *drvdata = snd_soc_component_get_drvdata(component); diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 4e1f101281e7..e8915519f427 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1315,7 +1315,7 @@ static struct snd_soc_dai_driver q6afe_dais[] = { }; static int q6afe_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name) { int id = args->args[0]; diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 159bf88b9f8c..8415e9bd2932 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -370,7 +370,7 @@ int snd_soc_component_of_xlate_dai_id(struct snd_soc_component *component, } int snd_soc_component_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name) { if (component->driver->of_xlate_dai_name) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f6d4e99b590c..50190654edda 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2994,7 +2994,7 @@ int snd_soc_get_dai_id(struct device_node *ep) } EXPORT_SYMBOL_GPL(snd_soc_get_dai_id); -int snd_soc_get_dai_name(struct of_phandle_args *args, +int snd_soc_get_dai_name(const struct of_phandle_args *args, const char **dai_name) { struct snd_soc_component *pos; -- 2.25.1

2 1

[PATCH 0/5] soc-pcm: tidyup snd_pcm_hardware setup for FE/BE
by Kuninori Morimoto 02 Mar '21

02 Mar '21

Hi Mark These patches tidyup snd_pcm_hardware setup for FE/BE. [1/5] changes behavior, but I think it is bug-fix. Kuninori Morimoto (5): ASoC: soc-pcm: remove strange format storing ASoC: soc-pcm: unpack dpcm_init_runtime_hw() ASoC: soc-pcm: add dpcm_runtime_setup_fe() ASoC: soc-pcm: add dpcm_runtime_setup() ASoC: soc-pcm: unpack dpcm_set_fe_runtime() sound/soc/soc-pcm.c | 79 +++++++++++++++++++++------------------------ 1 file changed, 36 insertions(+), 43 deletions(-) -- 2.25.1

2 6

[PATCH] ASoC: fsl_sai: Add pm qos cpu latency support
by Shengjiu Wang 02 Mar '21

02 Mar '21

On SoCs such as i.MX7ULP, cpuidle has some levels which may disable system/bus clocks, so need to add pm_qos to prevent cpuidle from entering low level idles and make sure system/bus clocks are enabled when sai is active. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 12 ++++++++++++ sound/soc/fsl/fsl_sai.h | 4 ++++ 2 files changed, 16 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5e65b456d3e2..a22679dea578 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -10,6 +10,7 @@ #include <linux/module.h> #include <linux/of_address.h> #include <linux/of_device.h> +#include <linux/pm_qos.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/slab.h> @@ -1177,6 +1178,7 @@ static const struct fsl_sai_soc_data fsl_sai_vf610_data = { .fifo_depth = 32, .reg_offset = 0, .mclk0_is_mclk1 = false, + .flags = 0, }; static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { @@ -1185,6 +1187,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { .fifo_depth = 32, .reg_offset = 0, .mclk0_is_mclk1 = true, + .flags = 0, }; static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { @@ -1193,6 +1196,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { .fifo_depth = 16, .reg_offset = 8, .mclk0_is_mclk1 = false, + .flags = PMQOS_CPU_LATENCY, }; static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { @@ -1201,6 +1205,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { .fifo_depth = 128, .reg_offset = 8, .mclk0_is_mclk1 = false, + .flags = 0, }; static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { @@ -1209,6 +1214,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { .fifo_depth = 64, .reg_offset = 0, .mclk0_is_mclk1 = false, + .flags = 0, }; static const struct of_device_id fsl_sai_ids[] = { @@ -1235,6 +1241,9 @@ static int fsl_sai_runtime_suspend(struct device *dev) clk_disable_unprepare(sai->bus_clk); + if (sai->soc_data->flags & PMQOS_CPU_LATENCY) + cpu_latency_qos_remove_request(&sai->pm_qos_req); + regcache_cache_only(sai->regmap, true); return 0; @@ -1264,6 +1273,9 @@ static int fsl_sai_runtime_resume(struct device *dev) goto disable_tx_clk; } + if (sai->soc_data->flags & PMQOS_CPU_LATENCY) + cpu_latency_qos_add_request(&sai->pm_qos_req, 0); + regcache_cache_only(sai->regmap, false); regcache_mark_dirty(sai->regmap); regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), FSL_SAI_CSR_SR); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index ff2619f1b214..bc60030967dd 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -216,12 +216,15 @@ #define FSL_SAI_MAXBURST_TX 6 #define FSL_SAI_MAXBURST_RX 6 +#define PMQOS_CPU_LATENCY BIT(0) + struct fsl_sai_soc_data { bool use_imx_pcm; bool use_edma; bool mclk0_is_mclk1; unsigned int fifo_depth; unsigned int reg_offset; + unsigned int flags; }; /** @@ -273,6 +276,7 @@ struct fsl_sai { struct snd_dmaengine_dai_dma_data dma_params_tx; struct fsl_sai_verid verid; struct fsl_sai_param param; + struct pm_qos_request pm_qos_req; }; #define TX 1 -- 2.27.0

2 1

[PATCH] ASoC: fsl_xcvr: move reset assert into runtime_resume
by Shengjiu Wang 02 Mar '21

02 Mar '21

From: Viorel Suman <viorel.suman(a)nxp.com> Move reset assert into runtime_resume since we cannot rely on reset assert state when the device is put out from suspend. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index dd228b421e2c..4857e0df8681 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1243,10 +1243,6 @@ static __maybe_unused int fsl_xcvr_runtime_suspend(struct device *dev) if (ret < 0) dev_err(dev, "Failed to assert M0+ core: %d\n", ret); - ret = reset_control_assert(xcvr->reset); - if (ret < 0) - dev_err(dev, "Failed to assert M0+ reset: %d\n", ret); - regcache_cache_only(xcvr->regmap, true); clk_disable_unprepare(xcvr->spba_clk); @@ -1262,6 +1258,12 @@ static __maybe_unused int fsl_xcvr_runtime_resume(struct device *dev) struct fsl_xcvr *xcvr = dev_get_drvdata(dev); int ret; + ret = reset_control_assert(xcvr->reset); + if (ret < 0) { + dev_err(dev, "Failed to assert M0+ reset: %d\n", ret); + return ret; + } + ret = clk_prepare_enable(xcvr->ipg_clk); if (ret) { dev_err(dev, "failed to start IPG clock.\n"); -- 2.27.0

2 1

[PATCH v3] ASoC: rt1316: Add RT1316 SDCA vendor-specific driver
by shumingf＠realtek.com 02 Mar '21

02 Mar '21

From: Shuming Fan <shumingf(a)realtek.com> This is the initial amplifier driver for rt1316 SDCA version. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt1316-sdw.c | 744 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1316-sdw.h | 55 +++ 4 files changed, 807 insertions(+) create mode 100644 sound/soc/codecs/rt1316-sdw.c create mode 100644 sound/soc/codecs/rt1316-sdw.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e4cf14e66a51..ae833a54bcb3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -182,6 +182,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT711_SDW imply SND_SOC_RT715_SDW imply SND_SOC_RT1308_SDW + imply SND_SOC_RT1316_SDW imply SND_SOC_SGTL5000 imply SND_SOC_SI476X imply SND_SOC_SIMPLE_AMPLIFIER @@ -1144,6 +1145,11 @@ config SND_SOC_RT1308_SDW depends on I2C && SOUNDWIRE select REGMAP_SOUNDWIRE +config SND_SOC_RT1316_SDW + tristate "Realtek RT1316 Codec - SDW" + depends on SOUNDWIRE + select REGMAP_SOUNDWIRE + config SND_SOC_RT5514 tristate depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 81357dc62ea0..1585bc1d9a25 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -173,6 +173,7 @@ snd-soc-rt1015p-objs := rt1015p.o snd-soc-rt1305-objs := rt1305.o snd-soc-rt1308-objs := rt1308.o snd-soc-rt1308-sdw-objs := rt1308-sdw.o +snd-soc-rt1316-sdw-objs := rt1316-sdw.o snd-soc-rt274-objs := rt274.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o @@ -487,6 +488,7 @@ obj-$(CONFIG_SND_SOC_RT1015P) += snd-soc-rt1015p.o obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o obj-$(CONFIG_SND_SOC_RT1308) += snd-soc-rt1308.o obj-$(CONFIG_SND_SOC_RT1308_SDW) += snd-soc-rt1308-sdw.o +obj-$(CONFIG_SND_SOC_RT1316_SDW) += snd-soc-rt1316-sdw.o obj-$(CONFIG_SND_SOC_RT274) += snd-soc-rt274.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c new file mode 100644 index 000000000000..a6fb34a48f33 --- /dev/null +++ b/sound/soc/codecs/rt1316-sdw.c @@ -0,0 +1,744 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt1316-sdw.c -- rt1316 SDCA ALSA SoC amplifier audio driver +// +// Copyright(c) 2021 Realtek Semiconductor Corp. +// +// +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/pm_runtime.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include "rt1316-sdw.h" + +static const struct reg_default rt1316_reg_defaults[] = { + { 0x3004, 0x00 }, + { 0x3005, 0x00 }, + { 0x3206, 0x00 }, + { 0xc001, 0x00 }, + { 0xc002, 0x00 }, + { 0xc003, 0x00 }, + { 0xc004, 0x00 }, + { 0xc005, 0x00 }, + { 0xc006, 0x00 }, + { 0xc007, 0x00 }, + { 0xc008, 0x00 }, + { 0xc009, 0x00 }, + { 0xc00a, 0x00 }, + { 0xc00b, 0x00 }, + { 0xc00c, 0x00 }, + { 0xc00d, 0x00 }, + { 0xc00e, 0x00 }, + { 0xc00f, 0x00 }, + { 0xc010, 0xa5 }, + { 0xc011, 0x00 }, + { 0xc012, 0xff }, + { 0xc013, 0xff }, + { 0xc014, 0x40 }, + { 0xc015, 0x00 }, + { 0xc016, 0x00 }, + { 0xc017, 0x00 }, + { 0xc605, 0x30 }, + { 0xc700, 0x0a }, + { 0xc701, 0xaa }, + { 0xc702, 0x1a }, + { 0xc703, 0x0a }, + { 0xc710, 0x80 }, + { 0xc711, 0x00 }, + { 0xc712, 0x3e }, + { 0xc713, 0x80 }, + { 0xc714, 0x80 }, + { 0xc715, 0x06 }, + { 0xd101, 0x00 }, + { 0xd102, 0x30 }, + { 0xd103, 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_UDMPU21, RT1316_SDCA_CTL_UDMPU_CLUSTER, 0), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_L), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_R), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_XU24, RT1316_SDCA_CTL_BYPASS, 0), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE23, RT1316_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE22, RT1316_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE24, RT1316_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, +}; + +static const struct reg_sequence rt1316_blind_write[] = { + { 0xc710, 0x17 }, + { 0xc711, 0x80 }, + { 0xc712, 0x26 }, + { 0xc713, 0x06 }, + { 0xc714, 0x80 }, + { 0xc715, 0x06 }, + { 0xc702, 0x0a }, + { 0xc703, 0x0a }, + { 0xc001, 0x45 }, + { 0xc003, 0x00 }, + { 0xc004, 0x11 }, + { 0xc005, 0x00 }, + { 0xc006, 0x00 }, + { 0xc106, 0x00 }, + { 0xc007, 0x11 }, + { 0xc008, 0x11 }, + { 0xc009, 0x00 }, + + { 0x2f0a, 0x00 }, + { 0xd101, 0xf0 }, + { 0xd103, 0x9b }, + { 0x2f36, 0x8e }, + { 0x3206, 0x80 }, + { 0x3211, 0x0b }, + { 0x3216, 0x06 }, + { 0xc614, 0x20 }, + { 0xc615, 0x0a }, + { 0xc616, 0x02 }, + { 0xc617, 0x00 }, + { 0xc60b, 0x10 }, + { 0xc60e, 0x05 }, + { 0xc102, 0x00 }, + { 0xc090, 0xb0 }, + { 0xc00f, 0x01 }, + { 0xc09c, 0x7b }, + + { 0xc602, 0x07 }, + { 0xc603, 0x07 }, + { 0xc0a3, 0x71 }, + { 0xc00b, 0x30 }, + { 0xc093, 0x80 }, + { 0xc09d, 0x80 }, + { 0xc0b0, 0x77 }, + { 0xc010, 0xa5 }, + { 0xc050, 0x83 }, + { 0x2f55, 0x03 }, + { 0x3217, 0xb5 }, + { 0x3202, 0x02 }, + + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_XU24, RT1316_SDCA_CTL_BYPASS, 0), 0x00 }, + + /* for IV sense */ + { 0x2232, 0x80 }, + { 0xc0b0, 0x77 }, + { 0xc011, 0x00 }, + { 0xc020, 0x00 }, + { 0xc023, 0x00 }, + { 0x3101, 0x00 }, + { 0x3004, 0xa0 }, + { 0x3005, 0xb1 }, + { 0xc007, 0x11 }, + { 0xc008, 0x11 }, + { 0xc009, 0x00 }, + { 0xc022, 0xd6 }, + { 0xc025, 0xd6 }, + + { 0xd001, 0x03 }, + { 0xd002, 0xbf }, + { 0xd003, 0x03 }, + { 0xd004, 0xbf }, +}; + +static bool rt1316_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2f0a: + case 0x2f36: + case 0x3203 ... 0x320e: + case 0xc000 ... 0xc7b4: + case 0xcf00 ... 0xcf03: + case 0xd101 ... 0xd103: + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_UDMPU21, RT1316_SDCA_CTL_UDMPU_CLUSTER, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_L): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_R): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE23, RT1316_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE27, RT1316_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE22, RT1316_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE24, RT1316_SDCA_CTL_REQ_POWER_STATE, 0): + return true; + default: + return false; + } +} + +static bool rt1316_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0xc000: + case 0xc093: + case 0xc09d: + case 0xc0a3: + case 0xc201: + case 0xc427 ... 0xc428: + case 0xd102: + return true; + default: + return false; + } +} + +static const struct regmap_config rt1316_sdw_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = rt1316_readable_register, + .volatile_reg = rt1316_volatile_register, + .max_register = 0x4108ffff, + .reg_defaults = rt1316_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rt1316_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt1316_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + prop->is_sdca = true; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x04; /* BITMAP: 00000100 */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + dev_dbg(&slave->dev, "%s\n", __func__); + + return 0; +} + +static int rt1316_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(dev); + + if (rt1316->hw_init) + return 0; + + if (rt1316->first_hw_init) { + regcache_cache_only(rt1316->regmap, false); + regcache_cache_bypass(rt1316->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + /* sw reset */ + regmap_write(rt1316->regmap, 0xc000, 0x02); + + /* initial settings - blind write */ + regmap_multi_reg_write(rt1316->regmap, rt1316_blind_write, + ARRAY_SIZE(rt1316_blind_write)); + + if (rt1316->first_hw_init) { + regcache_cache_bypass(rt1316->regmap, false); + regcache_mark_dirty(rt1316->regmap); + } else + rt1316->first_hw_init = true; + + /* Mark Slave initialization complete */ + rt1316->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + return 0; +} + +static int rt1316_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(&slave->dev); + + /* Update the status */ + rt1316->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + rt1316->hw_init = false; + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt1316->hw_init || rt1316->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt1316_io_init(&slave->dev, slave); +} + +static int rt1316_classd_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 rt1316_sdw_priv *rt1316 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE23, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE27, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE22, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE23, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE27, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE22, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + break; + + default: + break; + } + + return 0; +} + +static int rt1316_pde24_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 rt1316_sdw_priv *rt1316 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE24, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE24, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + break; + } + return 0; +} + +static const char * const rt1316_rx_data_ch_select[] = { + "L,R", + "L,L", + "L,R", + "L,L+R", + "R,L", + "R,R", + "R,L+R", + "L+R,L", + "L+R,R", + "L+R,L+R", +}; + +static SOC_ENUM_SINGLE_DECL(rt1316_rx_data_ch_enum, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_UDMPU21, RT1316_SDCA_CTL_UDMPU_CLUSTER, 0), 0, + rt1316_rx_data_ch_select); + +static const struct snd_kcontrol_new rt1316_snd_controls[] = { + + /* I2S Data Channel Selection */ + SOC_ENUM("RX Channel Select", rt1316_rx_data_ch_enum), + + /* XU24 Bypass Control */ + SOC_SINGLE("XU24 Bypass Switch", + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_XU24, RT1316_SDCA_CTL_BYPASS, 0), 0, 1, 0), + + /* Left/Right IV tag */ + SOC_SINGLE("Left V Tag Select", 0x3004, 0, 7, 0), + SOC_SINGLE("Left I Tag Select", 0x3004, 4, 7, 0), + SOC_SINGLE("Right V Tag Select", 0x3005, 0, 7, 0), + SOC_SINGLE("Right I Tag Select", 0x3005, 4, 7, 0), + + /* IV mixer Control */ + SOC_DOUBLE("Isense Mixer Switch", 0xc605, 2, 0, 1, 1), + SOC_DOUBLE("Vsense Mixer Switch", 0xc605, 3, 1, 1, 1), +}; + +static const struct snd_kcontrol_new rt1316_sto_dac = + SOC_DAPM_DOUBLE_R("Switch", + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_L), + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_R), + 0, 1, 1); + +static const struct snd_soc_dapm_widget rt1316_dapm_widgets[] = { + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("DP1RX", "DP1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP2TX", "DP2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SWITCH("DAC", SND_SOC_NOPM, 0, 0, &rt1316_sto_dac), + + /* Output Lines */ + SND_SOC_DAPM_PGA_E("CLASS D", SND_SOC_NOPM, 0, 0, NULL, 0, + rt1316_classd_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + + SND_SOC_DAPM_SUPPLY("PDE 24", SND_SOC_NOPM, 0, 0, + rt1316_pde24_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA("I Sense", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("V Sense", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SIGGEN("I Gen"), + SND_SOC_DAPM_SIGGEN("V Gen"), +}; + +static const struct snd_soc_dapm_route rt1316_dapm_routes[] = { + { "DAC", "Switch", "DP1RX" }, + { "CLASS D", NULL, "DAC" }, + { "SPOL", NULL, "CLASS D" }, + { "SPOR", NULL, "CLASS D" }, + + { "I Sense", NULL, "I Gen" }, + { "V Sense", NULL, "V Gen" }, + { "I Sense", NULL, "PDE 24" }, + { "V Sense", NULL, "PDE 24" }, + { "DP2TX", NULL, "I Sense" }, + { "DP2TX", NULL, "V Sense" }, +}; + +static int rt1316_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + struct sdw_stream_data *stream; + + if (!sdw_stream) + return 0; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = sdw_stream; + + /* Use tx_mask or rx_mask to configure stream tag and set dma_data */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = stream; + else + dai->capture_dma_data = stream; + + return 0; +} + +static void rt1316_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int rt1316_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1316_sdw_priv *rt1316 = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_data *stream; + int retval, port, num_channels, ch_mask; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!rt1316->sdw_slave) + return -EINVAL; + + /* SoundWire specific configuration */ + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + port = 1; + } else { + direction = SDW_DATA_DIR_TX; + port = 2; + } + + num_channels = params_channels(params); + ch_mask = (1 << num_channels) - 1; + + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = num_channels; + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + port_config.ch_mask = ch_mask; + port_config.num = port; + + retval = sdw_stream_add_slave(rt1316->sdw_slave, &stream_config, + &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + return 0; +} + +static int rt1316_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1316_sdw_priv *rt1316 = + snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt1316->sdw_slave) + return -EINVAL; + + sdw_stream_remove_slave(rt1316->sdw_slave, stream->sdw_stream); + return 0; +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static struct sdw_slave_ops rt1316_slave_ops = { + .read_prop = rt1316_read_prop, + .update_status = rt1316_update_status, +}; + +static const struct snd_soc_component_driver soc_component_sdw_rt1316 = { + .controls = rt1316_snd_controls, + .num_controls = ARRAY_SIZE(rt1316_snd_controls), + .dapm_widgets = rt1316_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt1316_dapm_widgets), + .dapm_routes = rt1316_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt1316_dapm_routes), +}; + +static const struct snd_soc_dai_ops rt1316_aif_dai_ops = { + .hw_params = rt1316_sdw_hw_params, + .hw_free = rt1316_sdw_pcm_hw_free, + .set_sdw_stream = rt1316_set_sdw_stream, + .shutdown = rt1316_sdw_shutdown, +}; + +#define RT1316_STEREO_RATES SNDRV_PCM_RATE_48000 +#define RT1316_FORMATS (SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver rt1316_sdw_dai[] = { + { + .name = "rt1316-aif", + .playback = { + .stream_name = "DP1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT1316_STEREO_RATES, + .formats = RT1316_FORMATS, + }, + .capture = { + .stream_name = "DP2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT1316_STEREO_RATES, + .formats = RT1316_FORMATS, + }, + .ops = &rt1316_aif_dai_ops, + }, +}; + +static int rt1316_sdw_init(struct device *dev, struct regmap *regmap, + struct sdw_slave *slave) +{ + struct rt1316_sdw_priv *rt1316; + int ret; + + rt1316 = devm_kzalloc(dev, sizeof(*rt1316), GFP_KERNEL); + if (!rt1316) + return -ENOMEM; + + dev_set_drvdata(dev, rt1316); + rt1316->sdw_slave = slave; + rt1316->regmap = regmap; + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt1316->hw_init = false; + rt1316->first_hw_init = false; + + ret = devm_snd_soc_register_component(dev, + &soc_component_sdw_rt1316, + rt1316_sdw_dai, + ARRAY_SIZE(rt1316_sdw_dai)); + + dev_dbg(&slave->dev, "%s\n", __func__); + + return ret; +} + +static int rt1316_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *regmap; + + /* Regmap Initialization */ + regmap = devm_regmap_init_sdw(slave, &rt1316_sdw_regmap); + if (!regmap) + return -EINVAL; + + return rt1316_sdw_init(&slave->dev, regmap, slave); +} + +static const struct sdw_device_id rt1316_id[] = { + SDW_SLAVE_ENTRY_EXT(0x025d, 0x1316, 0x3, 0x1, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt1316_id); + +static int __maybe_unused rt1316_dev_suspend(struct device *dev) +{ + struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(dev); + + if (!rt1316->hw_init) + return 0; + + regcache_cache_only(rt1316->regmap, true); + + return 0; +} + +#define RT1316_PROBE_TIMEOUT 5000 + +static int __maybe_unused rt1316_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt1316->hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT1316_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt1316->regmap, false); + regcache_sync(rt1316->regmap); + + return 0; +} + +static const struct dev_pm_ops rt1316_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt1316_dev_suspend, rt1316_dev_resume) + SET_RUNTIME_PM_OPS(rt1316_dev_suspend, rt1316_dev_resume, NULL) +}; + +static struct sdw_driver rt1316_sdw_driver = { + .driver = { + .name = "rt1316-sdca", + .owner = THIS_MODULE, + .pm = &rt1316_pm, + }, + .probe = rt1316_sdw_probe, + .ops = &rt1316_slave_ops, + .id_table = rt1316_id, +}; +module_sdw_driver(rt1316_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT1316 driver SDCA SDW"); +MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt1316-sdw.h b/sound/soc/codecs/rt1316-sdw.h new file mode 100644 index 000000000000..cbcdaa8f8cfa --- /dev/null +++ b/sound/soc/codecs/rt1316-sdw.h @@ -0,0 +1,55 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt1316-sdw.h -- RT1316 SDCA ALSA SoC audio driver header + * + * Copyright(c) 2021 Realtek Semiconductor Corp. + */ + +#ifndef __RT1316_SDW_H__ +#define __RT1316_SDW_H__ + +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/soundwire/sdw_registers.h> +#include <sound/soc.h> + +/* RT1316 SDCA Control - function number */ +#define FUNC_NUM_SMART_AMP 0x04 + +/* RT1316 SDCA entity */ +#define RT1316_SDCA_ENT_PDE23 0x31 +#define RT1316_SDCA_ENT_PDE27 0x32 +#define RT1316_SDCA_ENT_PDE22 0x33 +#define RT1316_SDCA_ENT_PDE24 0x34 +#define RT1316_SDCA_ENT_XU24 0x24 +#define RT1316_SDCA_ENT_FU21 0x03 +#define RT1316_SDCA_ENT_UDMPU21 0x02 + +/* RT1316 SDCA control */ +#define RT1316_SDCA_CTL_SAMPLE_FREQ_INDEX 0x10 +#define RT1316_SDCA_CTL_REQ_POWER_STATE 0x01 +#define RT1316_SDCA_CTL_BYPASS 0x01 +#define RT1316_SDCA_CTL_FU_MUTE 0x01 +#define RT1316_SDCA_CTL_FU_VOLUME 0x02 +#define RT1316_SDCA_CTL_UDMPU_CLUSTER 0x10 + +/* RT1316 SDCA channel */ +#define CH_L 0x01 +#define CH_R 0x02 + +struct rt1316_sdw_priv { + struct snd_soc_component *component; + struct regmap *regmap; + struct sdw_slave *sdw_slave; + enum sdw_slave_status status; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +#endif /* __RT1316_SDW_H__ */ -- 2.29.0

2 1

[PATCH] ASoC: SOF: Intel: hda: turn off display power in resume
by Kai Vehmanen 02 Mar '21

02 Mar '21

Turn off display power at the end of controller resume flow. This is now possible with the changes done in commit 87fc20e4a0cb ("ASoC: SOF: Intel: hda: use hdac_ext fine-grained link management"). As codec driver is able to request the HDA link to be brought back up, the controller no longer needs to blindly keep display power enabled. Co-developed-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Keyon Jie <yang.jie(a)intel.com> --- sound/soc/sof/intel/hda-dsp.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 5788fe356960..9900a058ba6e 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -678,7 +678,7 @@ static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume) if (ret < 0) { dev_err(sdev->dev, "error: failed to start controller after resume\n"); - return ret; + goto cleanup; } #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) @@ -704,6 +704,10 @@ static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume) hda_dsp_ctrl_ppcap_enable(sdev, true); hda_dsp_ctrl_ppcap_int_enable(sdev, true); +cleanup: + /* display codec can powered off after controller init */ + hda_codec_i915_display_power(sdev, false); + return 0; } base-commit: 9fd914d917da05641b42cab7d40bdf8ab06dac3b -- 2.29.2

2 1

[PATCH 0/4] ASoC: rt*: Constify static structs
by Rikard Falkeborn 02 Mar '21

02 Mar '21

Constify a number of static structs that are never modified in RealTek codecs. The most important patches are the first two, which constifies snd_soc_dai_ops and sdw_slave_ops, both which contain function pointers. The other two patches are for good measure, since I was already touching the code there. When doing this, I discovered sound/soc/codecs/rt1016.c is not in a Makefile, so there is not really any way to build it (I added locally to the Makefile to compile-test my changes). Is this expected or an oversight? Rikard Falkeborn (4): ASoC: rt*: Constify static struct sdw_slave_ops ASoC: rt*: Constify static struct snd_soc_dai_ops ASoC: rt*: Constify static struct acpi_device_id ASoc: rt5631: Constify static struct coeff_clk_div sound/soc/codecs/rt1011.c | 2 +- sound/soc/codecs/rt1015.c | 4 ++-- sound/soc/codecs/rt1016.c | 4 ++-- sound/soc/codecs/rt1305.c | 2 +- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt1308.c | 2 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 4 ++-- sound/soc/codecs/rt700-sdw.c | 2 +- sound/soc/codecs/rt700.c | 2 +- sound/soc/codecs/rt711-sdw.c | 2 +- sound/soc/codecs/rt711.c | 2 +- sound/soc/codecs/rt715-sdw.c | 2 +- sound/soc/codecs/rt715.c | 2 +- 14 files changed, 17 insertions(+), 17 deletions(-) -- 2.30.1

2 5

[PATCH 0/5] ASoC: rsnd: cleanup ppcheck warning for Renesas sound driver
by Kuninori Morimoto 02 Mar '21

02 Mar '21

Hi Mark, Pierre-Louis These patches are based on below patch-set which from Pierre-Louis, and cleanup cppcheck warnings for Rensas sound driver. [3/5] has Reported-by Pierre-Louis tag. Subject: [PATCH 0/8] ASoC: sh: remove cppcheck warnings Date: Fri, 19 Feb 2021 17:16:27 -0600 Kuninori Morimoto (5): ASoC: rsnd: cleanup ppcheck warning for ssiu.c ASoC: rsnd: cleanup ppcheck warning for ssi.c ASoC: rsnd: cleanup ppcheck warning for core.c ASoC: rsnd: cleanup ppcheck warning for cmd.c ASoC: rsnd: cleanup ppcheck warning for adg.c sound/soc/sh/rcar/adg.c | 15 +++++++------- sound/soc/sh/rcar/cmd.c | 15 +++++++------- sound/soc/sh/rcar/core.c | 32 +++++++++++++--------------- sound/soc/sh/rcar/ssi.c | 45 ++++++++++++++++++++-------------------- sound/soc/sh/rcar/ssiu.c | 22 +++++++++++--------- 5 files changed, 64 insertions(+), 65 deletions(-) -- 2.25.1

2 6

[PATCH] ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probe
by Benjamin Rood 02 Mar '21

02 Mar '21

According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has the following bit field definitions: | BITS | FIELD | RW | RESET | DEFINITION | | 15 | RSVD | RO | 0x0 | Reserved | | 14 | RSVD | RW | 0x1 | Reserved | | 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode | | 11:10 | RSVD | RO | 0x0 | Reserved | | 9:8 | LBI_RESP | RW | 0x1 | Integrator Response | | 7:6 | RSVD | RO | 0x0 | Reserved | | 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode | | 4:1 | RSVD | RO | 0x0 | Reserved | | 0 | EN | RW | 0x0 | Enable/Disable AVC | The original default value written to the DAP_AVC_CTRL register during sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to bits 4 and 10, which are defined as RESERVED. It would also not set bits 12 and 14 to their correct RESET values of 0x1, and instead set them to 0x0. While the DAP_AVC module is effectively disabled because the EN bit is 0, this default value is still writing invalid values to registers that are marked as read-only and RESERVED as well as not setting bits 12 and 14 to their correct default values as defined by the datasheet. The correct value that should be written to the DAP_AVC_CTRL register is 0x5100, which configures the register bits to the default values defined by the datasheet, and prevents any writes to bits defined as 'read-only'. Generally speaking, it is best practice to NOT attempt to write values to registers/bits defined as RESERVED, as it generally produces unwanted/undefined behavior, or errors. Also, all credit for this patch should go to my colleague Dan MacDonald <dmacdonald(a)curbellmedical.com> for finding this error in the first place. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Signed-off-by: Benjamin Rood <benjaminjrood(a)gmail.com> --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 4d6ff8114622..4c0e87e22b97 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -71,7 +71,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, { SGTL5000_DAP_MIX_CHAN, 0x0000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_CTRL, 0x5100 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, { SGTL5000_DAP_AVC_DECAY, 0x0050 }, -- 2.25.1

3 2

[PATCH] sound: soc/uniphier: Simplify the return expression of uniphier_aio_startup
by dingsenjie＠163.com 02 Mar '21

02 Mar '21

From: dingsenjie <dingsenjie(a)yulong.com> Simplify the return expression in the aio-cpu.c. Signed-off-by: dingsenjie <dingsenjie(a)yulong.com> --- sound/soc/uniphier/aio-cpu.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c index 25c40c2..cf98141 100644 --- a/sound/soc/uniphier/aio-cpu.c +++ b/sound/soc/uniphier/aio-cpu.c @@ -256,17 +256,12 @@ static int uniphier_aio_startup(struct snd_pcm_substream *substream, { struct uniphier_aio *aio = uniphier_priv(dai); struct uniphier_aio_sub *sub = &aio->sub[substream->stream]; - int ret; sub->substream = substream; sub->pass_through = 0; sub->use_mmap = true; - ret = aio_init(sub); - if (ret) - return ret; - - return 0; + return aio_init(sub); } static void uniphier_aio_shutdown(struct snd_pcm_substream *substream, -- 1.9.1

2 2

[PATCH 0/5] AsoC: rt5640/rt5651: Volume control fixes
by Hans de Goede 02 Mar '21

02 Mar '21

Hi All, Here is a series of rt5640/rt5651 volume-control fixes which I wrote while working on a bytcr-rt5640 UCM profile patch-series adding hardware-volume control to devices using this UCM profile. The UCM series will also work on older kernels, but it works best on kernels with this series applied, giving e.g. finer grained volume control and support for hardware muting the outputs. Regards, Hans Hans de Goede (5): ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10 ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10 ASoC: rt5640: Add emulated 'DAC1 Playback Switch' control ASoC: rt5640: Rename 'Mono DAC Playback Volume' to 'DAC2 Playback Volume' ASoC: Intel: bytcr_rt5640: Add used AIF to the components string sound/soc/codecs/rt5640.c | 106 +++++++++++++++++++++++--- sound/soc/codecs/rt5640.h | 4 + sound/soc/codecs/rt5651.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 11 ++- 4 files changed, 111 insertions(+), 14 deletions(-) -- 2.30.1

2 10

[PATCH] ASoC: Intel: boards: sof-wm8804: add check for PLL setting
by Colin King 02 Mar '21

02 Mar '21

From: Colin Ian King <colin.king(a)canonical.com> Currently the return from snd_soc_dai_set_pll is not checking for failure, this is the only driver in the kernel that ignores this, so it probably should be added for sake of completeness. Fix this by adding an error return check. Addresses-Coverity: ("Unchecked return value") Fixes: f139546fb7d4 ("ASoC: Intel: boards: sof-wm8804: support for Hifiberry Digiplus boards") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/boards/sof_wm8804.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_wm8804.c b/sound/soc/intel/boards/sof_wm8804.c index a46ba13e8eb0..6a181e45143d 100644 --- a/sound/soc/intel/boards/sof_wm8804.c +++ b/sound/soc/intel/boards/sof_wm8804.c @@ -124,7 +124,11 @@ static int sof_wm8804_hw_params(struct snd_pcm_substream *substream, } snd_soc_dai_set_clkdiv(codec_dai, WM8804_MCLK_DIV, mclk_div); - snd_soc_dai_set_pll(codec_dai, 0, 0, sysclk, mclk_freq); + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, sysclk, mclk_freq); + if (ret < 0) { + dev_err(rtd->card->dev, "Failed to set WM8804 PLL\n"); + return ret; + } ret = snd_soc_dai_set_sysclk(codec_dai, WM8804_TX_CLKSRC_PLL, sysclk, SND_SOC_CLOCK_OUT); -- 2.30.0

3 2

[PATCH] ASoC: es8316: Simplify adc_pga_gain_tlv table
by Hans de Goede 02 Mar '21

02 Mar '21

Most steps in this table are steps of 3dB (300 centi-dB), so we can simplify the table. This not only reduces the amount of space it takes inside the kernel, this also makes alsa-lib's mixer code actually accept the table, where as before this change alsa-lib saw the "ADC PGA Gain" control as a control without a dB scale. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/es8316.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 1ce84f12a4cf..9ab973958fc7 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -63,13 +63,8 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 1, 1, TLV_DB_SCALE_ITEM(0, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(250, 0, 0), 3, 3, TLV_DB_SCALE_ITEM(450, 0, 0), - 4, 4, TLV_DB_SCALE_ITEM(700, 0, 0), - 5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0), - 6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0), - 7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0), - 8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0), - 9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0), - 10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0), + 4, 7, TLV_DB_SCALE_ITEM(700, 300, 0), + 8, 10, TLV_DB_SCALE_ITEM(1800, 300, 0), ); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv, -- 2.30.1

2 1

[PATCH 0/7] Add I2S-MCC support for Microchip's SAMA7G5
by Codrin Ciubotariu 01 Mar '21

01 Mar '21

SAMA7G5 includes an updated version of I2S-MCC, found previously on SAM9X60. This controller includes 8 data pins, 4 for playback and 4 for capture. For I2S and LEFT_J formats, these pins can be used to send/receive up to 8 audio channels. For DSP_A, with TDM, any pins pair (DIN/DOUT) from these 4 can be selected to send/receive data. This version also includes 2 FIFOs (send and receive). This patch set starts by moving the driver's bindings to yaml and continues with adding a new compatible for the SAMA7G5 variant, followed by the changes needed for I2S/LEFT_J support, TDM pin pair selection and FIFO support, exclusively for SAMA7G5. Codrin Ciubotariu (7): ASoC: convert Microchip I2SMCC binding to yaml dt-bindings: mchp,i2s-mcc: Add SAMA7G5 to binding ASoC: mchp-i2s-mcc: Add compatible for SAMA7G5 ASoC: mchp-i2s-mcc: Add multi-channel support for I2S and LEFT_J formats dt-bindings: mchp,i2s-mcc: Add property to specify pin pair for TDM ASoC: mchp-i2s-mcc: Add support to select TDM pins ASoC: mchp-i2s-mcc: Add FIFOs support .../bindings/sound/mchp,i2s-mcc.yaml | 108 ++++++++++++ .../bindings/sound/mchp-i2s-mcc.txt | 43 ----- sound/soc/atmel/Kconfig | 3 + sound/soc/atmel/mchp-i2s-mcc.c | 161 +++++++++++++++--- 4 files changed, 252 insertions(+), 63 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mchp,i2s-mcc.yaml delete mode 100644 Documentation/devicetree/bindings/sound/mchp-i2s-mcc.txt -- 2.27.0

3 9

[PATCH] ALSA: usb-audio: Fix Pioneer DJM devices URB_CONTROL request direction to set samplerate
by Nicolas MURE 01 Mar '21

01 Mar '21

As requested in this comment : https://mailman.alsa-project.org/pipermail/alsa-devel/2021-March/181275.html This commit only contains the fix about the `URB_CONTROL` request direction to set the samplerate of Pioneer DJM devices (`URB_CONTROL out`). Signed-off-by: Nicolas MURE <nicolas.mure2019(a)gmail.com> --- sound/usb/quirks.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 9ba4682ebc48..737b2729c0d3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1482,7 +1482,7 @@ static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs, usb_set_interface(subs->dev, 0, 1); // we should derive windex from fmt-sync_ep but it's not set snd_usb_ctl_msg(subs->stream->chip->dev, - usb_rcvctrlpipe(subs->stream->chip->dev, 0), + usb_sndctrlpipe(subs->stream->chip->dev, 0), 0x01, 0x22, 0x0100, windex, &sr, 0x0003); return 0; } -- 2.29.2

2 1

[PATCH v4 0/2] Dell-hardware-privacy-implementation-for-dell-laptop
by Perry Yuan 01 Mar '21

01 Mar '21

From: Perry Yuan <perry_yuan(a)dell.com> Hi All, This patch set is a new driver for dell mobile platform ,which has the hardware privacy feature. For micmute led control, the hotkey is Fn + F4, it is a hardware based mute state, and new privacy will prevent micphone void input from hardware layer, any application cannot get voice data when micmute activated. Camera mute use a new hardware design to control the camrea shutter. When video is muted, no OS application should be functionally able to capture images of the person/environment in front of the system Older history: [1]https://patchwork.kernel.org/project/platform-driver-x86/patch/2020122813… [2]https://patchwork.kernel.org/project/alsa-devel/patch/20201103125859.8759… [3]https://www.spinics.net/lists/alsa-devel/msg120537.html [4]https://github.com/thesofproject/linux/pull/2660 Perry Yuan (2): platform/x86: dell-privacy: Add support for Dell hardware privacy ASoC: rt715:add micmute led state control supports .../testing/sysfs-platform-dell-privacy-wmi | 32 ++ drivers/platform/x86/Kconfig | 17 + drivers/platform/x86/Makefile | 4 +- drivers/platform/x86/dell-laptop.c | 26 +- drivers/platform/x86/dell-privacy-acpi.c | 164 +++++++++ drivers/platform/x86/dell-privacy-wmi.c | 340 ++++++++++++++++++ drivers/platform/x86/dell-privacy-wmi.h | 35 ++ drivers/platform/x86/dell-wmi.c | 29 +- sound/soc/codecs/rt715-sdca.c | 12 + sound/soc/codecs/rt715.c | 12 + 10 files changed, 655 insertions(+), 16 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi create mode 100644 drivers/platform/x86/dell-privacy-acpi.c create mode 100644 drivers/platform/x86/dell-privacy-wmi.c create mode 100644 drivers/platform/x86/dell-privacy-wmi.h -- 2.25.1

1 0

[PATCH] sound/usb generate midi streaming substream names from jack names
by George Harker 01 Mar '21

01 Mar '21

A number of devices have named substreams which are hard to remember / decypher from <device> MIDI n names. Eg. Korg puts a pass through on one substream and iConnectivity devices name the connections. This makes it easier to connect to the correct device. Devices which handle naming through quirks are unaffected by this change. Addresses TODO comment in sound/usb/midi.c Signed-off-by: George Harker <george(a)george-graphics.co.uk> --- sound/usb/midi.c | 122 ++++++++++++++++++++++++++++++++++++++--------- sound/usb/midi.h | 2 + 2 files changed, 101 insertions(+), 23 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 0c23fa6d8..c6651a566 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -47,6 +47,7 @@ #include <linux/usb.h> #include <linux/wait.h> #include <linux/usb/audio.h> +#include <linux/usb/midi.h> #include <linux/module.h> #include <sound/core.h> @@ -77,23 +78,6 @@ MODULE_AUTHOR("Clemens Ladisch <clemens(a)ladisch.de>"); MODULE_DESCRIPTION("USB Audio/MIDI helper module"); MODULE_LICENSE("Dual BSD/GPL"); - -struct usb_ms_header_descriptor { - __u8 bLength; - __u8 bDescriptorType; - __u8 bDescriptorSubtype; - __u8 bcdMSC[2]; - __le16 wTotalLength; -} __attribute__ ((packed)); - -struct usb_ms_endpoint_descriptor { - __u8 bLength; - __u8 bDescriptorType; - __u8 bDescriptorSubtype; - __u8 bNumEmbMIDIJack; - __u8 baAssocJackID[]; -} __attribute__ ((packed)); - struct snd_usb_midi_in_endpoint; struct snd_usb_midi_out_endpoint; struct snd_usb_midi_endpoint; @@ -1756,12 +1740,68 @@ static void snd_usbmidi_get_port_info(struct snd_rawmidi *rmidi, int number, } } +static struct usb_midi_in_jack_descriptor *find_usb_in_jack_descriptor( + struct usb_host_interface *hostif, uint8_t jack_id) +{ + unsigned char *extra = hostif->extra; + int extralen = hostif->extralen; + + while (extralen > 4) { + struct usb_midi_in_jack_descriptor *injd = + (struct usb_midi_in_jack_descriptor *)extra; + + if (injd->bLength > 4 && + injd->bDescriptorType == USB_DT_CS_INTERFACE && + injd->bDescriptorSubtype == UAC_MIDI_IN_JACK && + injd->bJackID == jack_id) + return injd; + if (!extra[0]) + break; + extralen -= extra[0]; + extra += extra[0]; + } + return NULL; +} + +static struct usb_midi_out_jack_descriptor *find_usb_out_jack_descriptor( + struct usb_host_interface *hostif, uint8_t jack_id) +{ + unsigned char *extra = hostif->extra; + int extralen = hostif->extralen; + + while (extralen > 4) { + struct usb_midi_out_jack_descriptor *outjd = + (struct usb_midi_out_jack_descriptor *)extra; + + if (outjd->bLength > 4 && + outjd->bDescriptorType == USB_DT_CS_INTERFACE && + outjd->bDescriptorSubtype == UAC_MIDI_OUT_JACK && + outjd->bJackID == jack_id) + return outjd; + if (!extra[0]) + break; + extralen -= extra[0]; + extra += extra[0]; + } + return NULL; +} + static void snd_usbmidi_init_substream(struct snd_usb_midi *umidi, - int stream, int number, + int stream, int number, int jack_id, struct snd_rawmidi_substream **rsubstream) { struct port_info *port_info; const char *name_format; + struct usb_interface *intf; + struct usb_host_interface *hostif; + struct usb_midi_in_jack_descriptor *injd; + struct usb_midi_out_jack_descriptor *outjd; + uint8_t jack_name_buf[32]; + uint8_t *default_jack_name = "MIDI"; + uint8_t *jack_name = default_jack_name; + uint8_t iJack; + size_t sz; + int res; struct snd_rawmidi_substream *substream = snd_usbmidi_find_substream(umidi, stream, number); @@ -1771,11 +1811,35 @@ static void snd_usbmidi_init_substream(struct snd_usb_midi *umidi, return; } - /* TODO: read port name from jack descriptor */ + intf = umidi->iface; + if (intf && jack_id >= 0) { + hostif = intf->cur_altsetting; + iJack = 0; + if (stream != SNDRV_RAWMIDI_STREAM_OUTPUT) { + /* in jacks connect to outs */ + outjd = find_usb_out_jack_descriptor(hostif, jack_id); + if (outjd) { + sz = USB_DT_MIDI_OUT_SIZE(outjd->bNrInputPins); + iJack = *(((uint8_t *) outjd) + sz - 1); + } + } else { + /* and out jacks connect to ins */ + injd = find_usb_in_jack_descriptor(hostif, jack_id); + if (injd) + iJack = injd->iJack; + } + if (iJack != 0) { + res = usb_string(umidi->dev, iJack, jack_name_buf, 32); + if (res) + jack_name = jack_name_buf; + } + } + port_info = find_port_info(umidi, number); - name_format = port_info ? port_info->name : "%s MIDI %d"; + name_format = port_info ? port_info->name : + (jack_name != default_jack_name ? "%s %s" : "%s %s %d"); snprintf(substream->name, sizeof(substream->name), - name_format, umidi->card->shortname, number + 1); + name_format, umidi->card->shortname, jack_name, number + 1); *rsubstream = substream; } @@ -1810,6 +1874,7 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi, snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_OUTPUT, out_ports, + endpoints[i].assoc_out_jacks[j], &umidi->endpoints[i].out->ports[j].substream); ++out_ports; } @@ -1817,6 +1882,7 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi, snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_INPUT, in_ports, + endpoints[i].assoc_in_jacks[j], &umidi->endpoints[i].in->ports[j].substream); ++in_ports; } @@ -1862,7 +1928,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, struct usb_host_endpoint *hostep; struct usb_endpoint_descriptor *ep; struct usb_ms_endpoint_descriptor *ms_ep; - int i, epidx; + int i, j, epidx; intf = umidi->iface; if (!intf) @@ -1875,7 +1941,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ms_header->bDescriptorType == USB_DT_CS_INTERFACE && ms_header->bDescriptorSubtype == UAC_HEADER) dev_dbg(&umidi->dev->dev, "MIDIStreaming version %02x.%02x\n", - ms_header->bcdMSC[1], ms_header->bcdMSC[0]); + ((uint8_t *)&ms_header->bcdMSC)[1], ((uint8_t *)&ms_header->bcdMSC)[0]); else dev_warn(&umidi->dev->dev, "MIDIStreaming interface descriptor not found\n"); @@ -1911,6 +1977,10 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, endpoints[epidx].out_interval = 1; endpoints[epidx].out_cables = (1 << ms_ep->bNumEmbMIDIJack) - 1; + for (j = 0; j < ms_ep->bNumEmbMIDIJack; ++j) + endpoints[epidx].assoc_out_jacks[j] = ms_ep->baAssocJackID[j]; + for (; j < 0x10; ++j) + endpoints[epidx].assoc_out_jacks[j] = -1; dev_dbg(&umidi->dev->dev, "EP %02X: %d jack(s)\n", ep->bEndpointAddress, ms_ep->bNumEmbMIDIJack); } else { @@ -1928,6 +1998,10 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, endpoints[epidx].in_interval = 1; endpoints[epidx].in_cables = (1 << ms_ep->bNumEmbMIDIJack) - 1; + for (j = 0; j < ms_ep->bNumEmbMIDIJack; ++j) + endpoints[epidx].assoc_in_jacks[j] = ms_ep->baAssocJackID[j]; + for (; j < 0x10; ++j) + endpoints[epidx].assoc_in_jacks[j] = -1; dev_dbg(&umidi->dev->dev, "EP %02X: %d jack(s)\n", ep->bEndpointAddress, ms_ep->bNumEmbMIDIJack); } @@ -2244,11 +2318,13 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi *umidi, snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_OUTPUT, cable, + -1 /* prevent trying to find jack */, &umidi->endpoints[cable & 1].out->ports[cable].substream); if (endpoint->in_cables & (1 << cable)) snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_INPUT, cable, + -1 /* prevent trying to find jack */, &umidi->endpoints[0].in->ports[cable].substream); } return 0; diff --git a/sound/usb/midi.h b/sound/usb/midi.h index 8c38aec22..3f153195c 100644 --- a/sound/usb/midi.h +++ b/sound/usb/midi.h @@ -13,6 +13,8 @@ struct snd_usb_midi_endpoint_info { uint8_t in_interval; uint16_t out_cables; /* bitmask */ uint16_t in_cables; /* bitmask */ + int16_t assoc_in_jacks[16]; + int16_t assoc_out_jacks[16]; }; /* for QUIRK_MIDI_YAMAHA, data is NULL */ -- 2.20.1

2 5

[PATCH] ALSA: usb-audio: use Corsair Virtuoso mapping for Corsair Virtuoso SE.
by Andrea Fagiani 01 Mar '21

01 Mar '21

The Corsair Virtuoso SE RGB Wireless is a USB headset with a mic and a sidetone feature. Assign the Corsair Virtuoso name map to the SE product ids as well, in order to label its mixer appropriately and allow userspace to pick the correct volume controls. Signed-off-by: Andrea Fagiani <andfagiani(a)gmail.com> --- sound/usb/mixer_maps.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index a7212f16660e..646deb6244b1 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -536,6 +536,16 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x05a7, 0x1020), .map = bose_companion5_map, }, + { + /* Corsair Virtuoso SE (wired mode) */ + .id = USB_ID(0x1b1c, 0x0a3d), + .map = corsair_virtuoso_map, + }, + { + /* Corsair Virtuoso SE (wireless mode) */ + .id = USB_ID(0x1b1c, 0x0a3e), + .map = corsair_virtuoso_map, + }, { /* Corsair Virtuoso (wired mode) */ .id = USB_ID(0x1b1c, 0x0a41), -- 2.25.1

2 1

[PATCH v2] ALSA: hda/hdmi: let new platforms assign the pcm slot dynamically
by Hui Wang 01 Mar '21

01 Mar '21

If the platform set the dyn_pcm_assign to true, it will call hdmi_find_pcm_slot() to find a pcm slot when hdmi/dp monitor is connected and need to create a pcm. So far only intel_hsw_common_init() and patch_nvhdmi() set the dyn_pcm_assign to true, here we let tgl platforms assign the pcm slot dynamically first, if the driver runs for a period of time and there is no regression reported, we could set no_fixed_assgin to true in the intel_hsw_common_init(), and then set it to true in the patch_nvhdmi(). This change comes from the discussion between Takashi and Kai Vehmanen. Please refer to: https://github.com/alsa-project/alsa-lib/pull/118 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- Did the test with this patch on 2 TGL machines, and with both legacy HDA driver and sof driver, plug 1 monitor into the machine, plug 2 monitors into the machine and plug 2 monitors into a dp-mst hub, they all worked as expcted, the first 2 pcm slots was assigned to monitors, and playback worked well. sound/pci/hda/patch_hdmi.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 97adff0cbcab..1aca815b920d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -157,6 +157,7 @@ struct hdmi_spec { bool dyn_pin_out; bool dyn_pcm_assign; + bool no_fixed_assign; bool intel_hsw_fixup; /* apply Intel platform-specific fixups */ /* * Non-generic VIA/NVIDIA specific @@ -1345,6 +1346,12 @@ static int hdmi_find_pcm_slot(struct hdmi_spec *spec, { int i; + /* on the new machines, try to assign the pcm slot dynamically, + * not use the preferred fixed map anymore. + */ + if (spec->no_fixed_assign) + goto last_try; + /* * generic_hdmi_build_pcms() may allocate extra PCMs on some * platforms (with maximum of 'num_nids + dev_num - 1') @@ -1374,6 +1381,7 @@ static int hdmi_find_pcm_slot(struct hdmi_spec *spec, return i; } + last_try: /* the last try; check the empty slots in pins */ for (i = 0; i < spec->num_nids; i++) { if (!test_bit(i, &spec->pcm_bitmap)) @@ -2988,8 +2996,16 @@ static int patch_i915_tgl_hdmi(struct hda_codec *codec) * the index indicate the port number. */ static const int map[] = {0x4, 0x6, 0x8, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf}; + int ret; - return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); + ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); + if (!ret) { + struct hdmi_spec *spec = codec->spec; + + spec->no_fixed_assign = true; + } + + return ret; } /* Intel Baytrail and Braswell; with eld notifier */ -- 2.25.1

3 3

conf: USB - add "Cmedia Audio" to USB-Audio.pcm.iec958_device
by GitHub pull_request - opened 01 Mar '21

01 Mar '21

alsa-project/alsa-lib pull request #122 was opened from jason77-wang: Otherwise, there will be a "Digital Output(S/PDIF)-Cmedia Audio" from Gnome UI, but there is no this physical interface on the card. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/122 Patch URL : https://github.com/alsa-project/alsa-lib/pull/122.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Turning an external amplifier on or off when audio is played
by Amr Bekhit 28 Feb '21

28 Feb '21

Hello, I'm working with an embedded system that uses the SGTL5000 to play audio. We have an amplifier connected to the DAC output, which we can turn off using a GPIO pin. I'd like to automatically enable the amplifier before playing a sound and disable it after playback has completed. Are there any hooks that would allow me to do that in userspace? If not, is there something in kernel-space that would allow me to accomplish this? Thanks, Amr

1 0

broadwell-rt286, bdw-rt5677: Fix inconsistency in card detection
by GitHub pull_request - edited 28 Feb '21

28 Feb '21

alsa-project/alsa-ucm-conf pull request #85 was edited from dpward: This conveniently provides a workaround for a bug in kernel 5.10+: https://bugzilla.kernel.org/show_bug.cgi?id=211985 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/85 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/85.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

broadwell-rt286, bdw-rt5677: Fix inconsistency in card detection
by GitHub pull_request - edited 28 Feb '21

28 Feb '21

alsa-project/alsa-ucm-conf pull request #85 was edited from dpward: Fixes: https://bugzilla.redhat.com/show_bug.cgi?id=1933229 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/85 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/85.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ALSA: hda/realtek: Enable headset mic of Acer SWIFT with ALC256
by chris.chiu＠canonical.com 28 Feb '21

28 Feb '21

From: Chris Chiu <chris.chiu(a)canonical.com> The Acer SWIFT Swift SF314-54/55 laptops with ALC256 cannot detect both the headset mic and the internal mic. Introduce new fixup to enable the jack sense and the headset mic. However, the internal mic actually connects to Intel SST audio. It still needs Intel SST support to make internal mic capture work. Signed-off-by: Chris Chiu <chris.chiu(a)canonical.com> --- v1 -> v2: remove unnecessary aamix fixup sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1927605f0f7e..4871507cd4bf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6406,6 +6406,7 @@ enum { ALC236_FIXUP_DELL_AIO_HEADSET_MIC, ALC282_FIXUP_ACER_DISABLE_LINEOUT, ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, + ALC256_FIXUP_ACER_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -7853,6 +7854,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_ACER_MIC_NO_PRESENCE, }, + [ALC256_FIXUP_ACER_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x02a1113c }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x90a1092f }, /* use as internal mic */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7879,9 +7890,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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), + SND_PCI_QUIRK(0x1025, 0x1269, "Acer SWIFT SF314-54", ALC256_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x129c, "Acer SWIFT SF314-55", ALC256_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), -- 2.20.1

4 4

[PATCH] ALSA: ctxfi: cthw20k2: fix mask on conf to allow 4 bits
by Colin King 28 Feb '21

28 Feb '21

From: Colin Ian King <colin.king(a)canonical.com> Currently the mask operation on variable conf is just 3 bits so the switch statement case value of 8 is unreachable dead code. The function daio_mgr_dao_init can be passed a 4 bit value, function dao_rsc_init calls it with conf set to: conf = (desc->msr & 0x7) | (desc->passthru << 3); so clearly when desc->passthru is set to 1 then conf can be at least 8. Fix this by changing the mask to 0xf. Fixes: 8cc72361481f ("ALSA: SB X-Fi driver merge") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/ctxfi/cthw20k2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index a855fb8c58bd..55af8ef29838 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -991,7 +991,7 @@ static int daio_mgr_dao_init(void *blk, unsigned int idx, unsigned int conf) if (idx < 4) { /* S/PDIF output */ - switch ((conf & 0x7)) { + switch ((conf & 0xf)) { case 1: set_field(&ctl->txctl[idx], ATXCTL_NUC, 0); break; -- 2.30.0

2 1

[PATCH] ALSA: usb-audio: Allow modifying parameters with succeeding hw_params calls
by Takashi Iwai 28 Feb '21

28 Feb '21

The recent fix for the hw constraints for implicit feedback streams via commit e4ea77f8e53f ("ALSA: usb-audio: Always apply the hw constraints for implicit fb sync") added the check of the matching endpoints and whether those EPs are already opened. This is needed and correct, per se, even for the normal streams without the implicit feedback, as the endpoint setup is exclusive. However, it's reported that there seem applications that behave in unexpected ways to update the hw_params without clearing the previous setup via hw_free, and those hit a problem now: then hw_params is called with still the previous EP setup kept, hence it's restricted with the previous own setup. Although the obvious fix is to call snd_pcm_hw_free() API in the application side, it's a kind of unwelcome change. This patch tries to ease the situation: in the endpoint check, we add a couple of more conditions and now skip the endpoint that is being used only by the stream in question itself. That is, in addition to the presence check of ep (ep->cur_audiofmt is non-NULL), when the following conditions are met, we skip such an ep: - ep->opened == 1, and - ep->cur_audiofmt == subs->cur_audiofmt. subs->cur_audiofmt is non-NULL only if it's a re-setup of hw_params, and ep->cur_audiofmt points to the currently set up parameters. So if those match, it must be this stream itself. Fixes: e4ea77f8e53f ("ALSA: usb-audio: Always apply the hw constraints for implicit fb sync") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211941 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index bf5a0f3c1fad..e5311b6bb3f6 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -845,13 +845,19 @@ get_sync_ep_from_substream(struct snd_usb_substream *subs) list_for_each_entry(fp, &subs->fmt_list, list) { ep = snd_usb_get_endpoint(chip, fp->endpoint); - if (ep && ep->cur_rate) - return ep; + if (ep && ep->cur_audiofmt) { + /* if EP is already opened solely for this substream, + * we still allow us to change the parameter; otherwise + * this substream has to follow the existing parameter + */ + if (ep->cur_audiofmt != subs->cur_audiofmt || ep->opened > 1) + return ep; + } if (!fp->implicit_fb) continue; /* for the implicit fb, check the sync ep as well */ ep = snd_usb_get_endpoint(chip, fp->sync_ep); - if (ep && ep->cur_rate) + if (ep && ep->cur_audiofmt) return ep; } return NULL; -- 2.26.2

1 0

[PATCH RFC] x86: remove toolchain check for X32 ABI capability
by Masahiro Yamada 28 Feb '21

28 Feb '21

This commit reverts 0bf6276392e9 ("x32: Warn and disable rather than error if binutils too old"). The help text in arch/x86/Kconfig says enabling the X32 ABI support needs binutils 2.22 or later. This is met because the minimal binutils version is 2.23 according to Documentation/process/changes.rst. I would not say I am not familiar with toolchain configuration, but I checked the configure.tgt code in binutils. The elf32_x86_64 emulation mode seems to be included when it is configured for the x86_64-*-linux-* target. I also tried lld and llvm-objcopy, and succeeded in building x32 VDSO. I removed the compile-time check in arch/x86/Makefile, in the hope of elf32_x86_64 being always supported. With this, CONFIG_X86_X32 and CONFIG_X86_X32_ABI will be equivalent. Rename the former to the latter. Signed-off-by: Masahiro Yamada <masahiroy(a)kernel.org> --- arch/x86/Kconfig | 8 ++------ arch/x86/Makefile | 16 ---------------- arch/x86/include/asm/syscall_wrapper.h | 6 +++--- arch/x86/include/asm/vdso.h | 2 +- arch/x86/kernel/process_64.c | 2 +- fs/fuse/file.c | 2 +- fs/xfs/xfs_ioctl32.c | 2 +- sound/core/control_compat.c | 16 ++++++++-------- sound/core/pcm_compat.c | 20 ++++++++++---------- 9 files changed, 27 insertions(+), 47 deletions(-) diff --git a/arch/x86/Kconfig b/arch/x86/Kconfig index 2792879d398e..7272cba2744c 100644 --- a/arch/x86/Kconfig +++ b/arch/x86/Kconfig @@ -2865,7 +2865,7 @@ config IA32_AOUT help Support old a.out binaries in the 32bit emulation. -config X86_X32 +config X86_X32_ABI bool "x32 ABI for 64-bit mode" depends on X86_64 help @@ -2874,10 +2874,6 @@ config X86_X32 full 64-bit register file and wide data path while leaving pointers at 32 bits for smaller memory footprint. - You will need a recent binutils (2.22 or later) with - elf32_x86_64 support enabled to compile a kernel with this - option set. - config COMPAT_32 def_bool y depends on IA32_EMULATION || X86_32 @@ -2886,7 +2882,7 @@ config COMPAT_32 config COMPAT def_bool y - depends on IA32_EMULATION || X86_X32 + depends on IA32_EMULATION || X86_X32_ABI if COMPAT config COMPAT_FOR_U64_ALIGNMENT diff --git a/arch/x86/Makefile b/arch/x86/Makefile index 2d6d5a28c3bf..e163549f5be7 100644 --- a/arch/x86/Makefile +++ b/arch/x86/Makefile @@ -125,22 +125,6 @@ else KBUILD_CFLAGS += -mcmodel=kernel endif -ifdef CONFIG_X86_X32 - x32_ld_ok := $(call try-run,\ - /bin/echo -e '1: .quad 1b' | \ - $(CC) $(KBUILD_AFLAGS) -c -x assembler -o "$$TMP" - && \ - $(OBJCOPY) -O elf32-x86-64 "$$TMP" "$$TMPO" && \ - $(LD) -m elf32_x86_64 "$$TMPO" -o "$$TMP",y,n) - ifeq ($(x32_ld_ok),y) - CONFIG_X86_X32_ABI := y - KBUILD_AFLAGS += -DCONFIG_X86_X32_ABI - KBUILD_CFLAGS += -DCONFIG_X86_X32_ABI - else - $(warning CONFIG_X86_X32 enabled but no binutils support) - endif -endif -export CONFIG_X86_X32_ABI - # # If the function graph tracer is used with mcount instead of fentry, # '-maccumulate-outgoing-args' is needed to prevent a GCC bug diff --git a/arch/x86/include/asm/syscall_wrapper.h b/arch/x86/include/asm/syscall_wrapper.h index a84333adeef2..69bf87c41a0b 100644 --- a/arch/x86/include/asm/syscall_wrapper.h +++ b/arch/x86/include/asm/syscall_wrapper.h @@ -158,7 +158,7 @@ extern long __ia32_sys_ni_syscall(const struct pt_regs *regs); #endif /* CONFIG_IA32_EMULATION */ -#ifdef CONFIG_X86_X32 +#ifdef CONFIG_X86_X32_ABI /* * For the x32 ABI, we need to create a stub for compat_sys_*() which is aware * of the x86-64-style parameter ordering of x32 syscalls. The syscalls common @@ -176,12 +176,12 @@ extern long __ia32_sys_ni_syscall(const struct pt_regs *regs); #define __X32_COMPAT_SYS_NI(name) \ __SYS_NI(x32, compat_sys_##name) -#else /* CONFIG_X86_X32 */ +#else /* CONFIG_X86_X32_ABI */ #define __X32_COMPAT_SYS_STUB0(name) #define __X32_COMPAT_SYS_STUBx(x, name, ...) #define __X32_COMPAT_COND_SYSCALL(name) #define __X32_COMPAT_SYS_NI(name) -#endif /* CONFIG_X86_X32 */ +#endif /* CONFIG_X86_X32_ABI */ #ifdef CONFIG_COMPAT diff --git a/arch/x86/include/asm/vdso.h b/arch/x86/include/asm/vdso.h index 98aa103eb4ab..2963a2f5dbc4 100644 --- a/arch/x86/include/asm/vdso.h +++ b/arch/x86/include/asm/vdso.h @@ -37,7 +37,7 @@ struct vdso_image { extern const struct vdso_image vdso_image_64; #endif -#ifdef CONFIG_X86_X32 +#ifdef CONFIG_X86_X32_ABI extern const struct vdso_image vdso_image_x32; #endif diff --git a/arch/x86/kernel/process_64.c b/arch/x86/kernel/process_64.c index d08307df69ad..a93b6f4296fc 100644 --- a/arch/x86/kernel/process_64.c +++ b/arch/x86/kernel/process_64.c @@ -656,7 +656,7 @@ void set_personality_64bit(void) static void __set_personality_x32(void) { -#ifdef CONFIG_X86_X32 +#ifdef CONFIG_X86_X32_ABI if (current->mm) current->mm->context.flags = 0; diff --git a/fs/fuse/file.c b/fs/fuse/file.c index 8cccecb55fb8..c53c620d1a7a 100644 --- a/fs/fuse/file.c +++ b/fs/fuse/file.c @@ -2797,7 +2797,7 @@ long fuse_do_ioctl(struct file *file, unsigned int cmd, unsigned long arg, #else if (flags & FUSE_IOCTL_COMPAT) { inarg.flags |= FUSE_IOCTL_32BIT; -#ifdef CONFIG_X86_X32 +#ifdef CONFIG_X86_X32_ABI if (in_x32_syscall()) inarg.flags |= FUSE_IOCTL_COMPAT_X32; #endif diff --git a/fs/xfs/xfs_ioctl32.c b/fs/xfs/xfs_ioctl32.c index 33c09ec8e6c0..e8038bc966e7 100644 --- a/fs/xfs/xfs_ioctl32.c +++ b/fs/xfs/xfs_ioctl32.c @@ -233,7 +233,7 @@ xfs_compat_ioc_fsbulkstat( inumbers_fmt_pf inumbers_func = xfs_fsinumbers_fmt_compat; bulkstat_one_fmt_pf bs_one_func = xfs_fsbulkstat_one_fmt_compat; -#ifdef CONFIG_X86_X32 +#ifdef CONFIG_X86_X32_ABI if (in_x32_syscall()) { /* * ... but on x32 the input xfs_fsop_bulkreq has pointers diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 1d708aab9c98..5d1b94bda2cd 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -153,7 +153,7 @@ struct snd_ctl_elem_value32 { unsigned char reserved[128]; }; -#ifdef CONFIG_X86_X32 +#ifdef CONFIG_X86_X32_ABI /* x32 has a different alignment for 64bit values from ia32 */ struct snd_ctl_elem_value_x32 { struct snd_ctl_elem_id id; @@ -165,7 +165,7 @@ struct snd_ctl_elem_value_x32 { } value; unsigned char reserved[128]; }; -#endif /* CONFIG_X86_X32 */ +#endif /* CONFIG_X86_X32_ABI */ /* get the value type and count of the control */ static int get_ctl_type(struct snd_card *card, struct snd_ctl_elem_id *id, @@ -350,7 +350,7 @@ static int snd_ctl_elem_write_user_compat(struct snd_ctl_file *file, return ctl_elem_write_user(file, data32, &data32->value); } -#ifdef CONFIG_X86_X32 +#ifdef CONFIG_X86_X32_ABI static int snd_ctl_elem_read_user_x32(struct snd_card *card, struct snd_ctl_elem_value_x32 __user *data32) { @@ -362,7 +362,7 @@ static int snd_ctl_elem_write_user_x32(struct snd_ctl_file *file, { return ctl_elem_write_user(file, data32, &data32->value); } -#endif /* CONFIG_X86_X32 */ +#endif /* CONFIG_X86_X32_ABI */ /* add or replace a user control */ static int snd_ctl_elem_add_compat(struct snd_ctl_file *file, @@ -421,10 +421,10 @@ enum { SNDRV_CTL_IOCTL_ELEM_WRITE32 = _IOWR('U', 0x13, struct snd_ctl_elem_value32), SNDRV_CTL_IOCTL_ELEM_ADD32 = _IOWR('U', 0x17, struct snd_ctl_elem_info32), SNDRV_CTL_IOCTL_ELEM_REPLACE32 = _IOWR('U', 0x18, struct snd_ctl_elem_info32), -#ifdef CONFIG_X86_X32 +#ifdef CONFIG_X86_X32_ABI SNDRV_CTL_IOCTL_ELEM_READ_X32 = _IOWR('U', 0x12, struct snd_ctl_elem_value_x32), SNDRV_CTL_IOCTL_ELEM_WRITE_X32 = _IOWR('U', 0x13, struct snd_ctl_elem_value_x32), -#endif /* CONFIG_X86_X32 */ +#endif /* CONFIG_X86_X32_ABI */ }; static inline long snd_ctl_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg) @@ -463,12 +463,12 @@ static inline long snd_ctl_ioctl_compat(struct file *file, unsigned int cmd, uns return snd_ctl_elem_add_compat(ctl, argp, 0); case SNDRV_CTL_IOCTL_ELEM_REPLACE32: return snd_ctl_elem_add_compat(ctl, argp, 1); -#ifdef CONFIG_X86_X32 +#ifdef CONFIG_X86_X32_ABI case SNDRV_CTL_IOCTL_ELEM_READ_X32: return snd_ctl_elem_read_user_x32(ctl->card, argp); case SNDRV_CTL_IOCTL_ELEM_WRITE_X32: return snd_ctl_elem_write_user_x32(ctl, argp); -#endif /* CONFIG_X86_X32 */ +#endif /* CONFIG_X86_X32_ABI */ } down_read(&snd_ioctl_rwsem); diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 590a46a9e78d..937f5117a81f 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -147,13 +147,13 @@ static int snd_pcm_ioctl_channel_info_compat(struct snd_pcm_substream *substream return err; } -#ifdef CONFIG_X86_X32 +#ifdef CONFIG_X86_X32_ABI /* X32 ABI has the same struct as x86-64 for snd_pcm_channel_info */ static int snd_pcm_channel_info_user(struct snd_pcm_substream *substream, struct snd_pcm_channel_info __user *src); #define snd_pcm_ioctl_channel_info_x32(s, p) \ snd_pcm_channel_info_user(s, p) -#endif /* CONFIG_X86_X32 */ +#endif /* CONFIG_X86_X32_ABI */ struct compat_snd_pcm_status64 { snd_pcm_state_t state; @@ -373,7 +373,7 @@ static int snd_pcm_ioctl_xfern_compat(struct snd_pcm_substream *substream, return err; } -#ifdef CONFIG_X86_X32 +#ifdef CONFIG_X86_X32_ABI /* X32 ABI has 64bit timespec and 64bit alignment */ struct snd_pcm_mmap_status_x32 { snd_pcm_state_t state; @@ -464,7 +464,7 @@ static int snd_pcm_ioctl_sync_ptr_x32(struct snd_pcm_substream *substream, return 0; } -#endif /* CONFIG_X86_X32 */ +#endif /* CONFIG_X86_X32_ABI */ /* */ @@ -484,10 +484,10 @@ enum { SNDRV_PCM_IOCTL_READN_FRAMES32 = _IOR('A', 0x53, struct snd_xfern32), SNDRV_PCM_IOCTL_STATUS_COMPAT64 = _IOR('A', 0x20, struct compat_snd_pcm_status64), SNDRV_PCM_IOCTL_STATUS_EXT_COMPAT64 = _IOWR('A', 0x24, struct compat_snd_pcm_status64), -#ifdef CONFIG_X86_X32 +#ifdef CONFIG_X86_X32_ABI SNDRV_PCM_IOCTL_CHANNEL_INFO_X32 = _IOR('A', 0x32, struct snd_pcm_channel_info), SNDRV_PCM_IOCTL_SYNC_PTR_X32 = _IOWR('A', 0x23, struct snd_pcm_sync_ptr_x32), -#endif /* CONFIG_X86_X32 */ +#endif /* CONFIG_X86_X32_ABI */ }; static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg) @@ -531,10 +531,10 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l case __SNDRV_PCM_IOCTL_SYNC_PTR32: return snd_pcm_common_ioctl(file, substream, cmd, argp); case __SNDRV_PCM_IOCTL_SYNC_PTR64: -#ifdef CONFIG_X86_X32 +#ifdef CONFIG_X86_X32_ABI if (in_x32_syscall()) return snd_pcm_ioctl_sync_ptr_x32(substream, argp); -#endif /* CONFIG_X86_X32 */ +#endif /* CONFIG_X86_X32_ABI */ return snd_pcm_common_ioctl(file, substream, cmd, argp); case SNDRV_PCM_IOCTL_HW_REFINE32: return snd_pcm_ioctl_hw_params_compat(substream, 1, argp); @@ -566,10 +566,10 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l return snd_pcm_status_user_compat64(substream, argp, false); case SNDRV_PCM_IOCTL_STATUS_EXT_COMPAT64: return snd_pcm_status_user_compat64(substream, argp, true); -#ifdef CONFIG_X86_X32 +#ifdef CONFIG_X86_X32_ABI case SNDRV_PCM_IOCTL_CHANNEL_INFO_X32: return snd_pcm_ioctl_channel_info_x32(substream, argp); -#endif /* CONFIG_X86_X32 */ +#endif /* CONFIG_X86_X32_ABI */ } return -ENOIOCTLCMD; -- 2.27.0

3 5

broadwell-rt286, bdw-rt5677: detect card when using Catpt driver
by GitHub pull_request - opened 27 Feb '21

27 Feb '21

alsa-project/alsa-ucm-conf pull request #85 was opened from dpward: Fixes: https://bugzilla.redhat.com/show_bug.cgi?id=1933229 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/85 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/85.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: usb-audio: Drop bogus dB range in too low level
by Takashi Iwai 27 Feb '21

27 Feb '21

Some USB audio firmware seem to report broken dB values for the volume controls, and this screws up applications like PulseAudio who blindly trusts the given data. For example, Edifier G2000 reports a PCM volume from -128dB to -127dB, and this results in barely inaudible sound. This patch adds a sort of sanity check at parsing the dB values in USB-audio driver and disables the dB reporting if the range looks bogus. Here, we assume -96dB as the bottom line of the max dB. Note that, if one can figure out that proper dB range later, it can be patched in the mixer maps. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211929 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index b1c78db0d470..b004b2e63a5d 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1307,6 +1307,17 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, /* totally crap, return an error */ return -EINVAL; } + } else { + /* if the max volume is too low, it's likely a bogus range; + * here we use -96dB as the threshold + */ + if (cval->dBmax <= -9600) { + usb_audio_info(cval->head.mixer->chip, + "%d:%d: bogus dB values (%d/%d), disabling dB reporting\n", + cval->head.id, mixer_ctrl_intf(cval->head.mixer), + cval->dBmin, cval->dBmax); + cval->dBmin = cval->dBmax = 0; + } } return 0; -- 2.26.2

1 0

[PATCH] alsa-tools/hdspmixer: enhance preset save and add hardware ouput loopback buttons
by fassl 27 Feb '21

27 Feb '21

>From 3c887b31fa57ca3004c2eaf0ee5abc0b3f94da70 Mon Sep 17 00:00:00 2001 From: Jasmin Fazlic <superfassl(a)gmail.com> Date: Thu, 4 Feb 2021 15:58:33 +0100 Subject: [PATCH 1/2] alsa-tools/hdspmixer: enhance saving of presets Changing the version in the file header would make a preset file not readable by older versions of the tool. If we just append new data always at the end of the save procedure we should have no problems reading them with different versions, as they all just read to a certain point and ignore the rest of the file. This patch implements the logic to save the presets first to a file called file_name.tmp and appends any extra data that would come after in a possibly present file_name file. Any data written by newer versions would remain in the preset file and from now on no old version should remove data written by newer versions. Also since we write to a temporary file and rename afterwards an extra feature is gained of not corrupting the preset should we crash. Signed-off-by: Jasmin Fazlic <superfassl(a)gmail.com> --- hdspmixer/src/HDSPMixerWindow.cxx | 98 ++++++++++++++++++++----------- 1 file changed, 64 insertions(+), 34 deletions(-) diff --git a/hdspmixer/src/HDSPMixerWindow.cxx b/hdspmixer/src/HDSPMixerWindow.cxx index 9efc25d..3b3d668 100644 --- a/hdspmixer/src/HDSPMixerWindow.cxx +++ b/hdspmixer/src/HDSPMixerWindow.cxx @@ -353,18 +353,25 @@ void HDSPMixerWindow::save() sizeof(inputs->strips[0]->data[0][0][0]->fader_pos) / sizeof(inputs->strips[0]->data[0][0][0]->fader_pos[0])); - - FILE *file; - - if ((file = fopen(file_name, "w")) == NULL) { - fl_alert("Error opening file %s for saving", file_name); + FILE *in,*out; + + /* We want to append any existing extra data that might got written by a + * newer version to this file, therefore write our data to file_name.tmp + * and append the old data. Also this way we would not corrupt the file + * should we crash. + */ + std::string const tmp = file_name + std::string(".tmp"); + char const * const tmpc = tmp.c_str(); + + if ((out = fopen(tmpc, "w")) == NULL) { + fl_alert("Error opening file %s for saving", tmpc); } if (dirty) { inputs->buttons->presets->save_preset(current_preset+1); } /* since hdspmixer 1.11, we also store the meter level settings. Indicate * the new on-disk structure via a small header */ - if (fwrite((void *)&header, sizeof(char), sizeof(header), file) != + if (fwrite((void *)&header, sizeof(char), sizeof(header), out) != sizeof(header)) { goto save_error; } @@ -374,99 +381,122 @@ void HDSPMixerWindow::save() for (int preset = 0; preset < 8; ++preset) { for (int channel = 0; channel < HDSP_MAX_CHANNELS; ++channel) { /* inputs pans and volumes */ - if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->pan_pos[0]), sizeof(int), pan_array_size, file) != pan_array_size) { + if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->pan_pos[0]), sizeof(int), pan_array_size, out) != pan_array_size) { goto save_error; } - if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->fader_pos[0]), sizeof(int), pan_array_size, file) != pan_array_size) { + if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->fader_pos[0]), sizeof(int), pan_array_size, out) != pan_array_size) { goto save_error; } /* playbacks pans and volumes */ - if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->pan_pos[0]), sizeof(int), pan_array_size, file) != pan_array_size) { + if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->pan_pos[0]), sizeof(int), pan_array_size, out) != pan_array_size) { goto save_error; } - if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->fader_pos[0]), sizeof(int), pan_array_size, file) != pan_array_size) { + if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->fader_pos[0]), sizeof(int), pan_array_size, out) != pan_array_size) { goto save_error; } /* inputs mute/solo/dest */ - if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->mute), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->mute), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->solo), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->solo), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->dest), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->dest), sizeof(int), 1, out) != 1) { goto save_error; } /* playbacks mute/solo/dest */ - if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->mute), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->mute), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->solo), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->solo), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->dest), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->dest), sizeof(int), 1, out) != 1) { goto save_error; } /* outputs volumes */ - if (fwrite((void *)&(outputs->strips[channel]->data[card][speed][preset]->fader_pos), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(outputs->strips[channel]->data[card][speed][preset]->fader_pos), sizeof(int), 1, out) != 1) { goto save_error; } } /* Lineouts */ - if (fwrite((void *)&(outputs->strips[HDSP_MAX_CHANNELS]->data[card][speed][preset]->fader_pos), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(outputs->strips[HDSP_MAX_CHANNELS]->data[card][speed][preset]->fader_pos), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(outputs->strips[HDSP_MAX_CHANNELS+1]->data[card][speed][preset]->fader_pos), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(outputs->strips[HDSP_MAX_CHANNELS+1]->data[card][speed][preset]->fader_pos), sizeof(int), 1, out) != 1) { goto save_error; } /* Global settings */ - if (fwrite((void *)&(data[card][speed][preset]->input), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->input), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->output), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->output), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->playback), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->playback), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->submix), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->submix), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->submix_value), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->submix_value), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->solo), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->solo), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->mute), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->mute), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->last_destination), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->last_destination), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->rmsplus3), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->rmsplus3), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->numbers), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->numbers), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->over), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->over), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->level), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->level), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->rate), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->rate), sizeof(int), 1, out) != 1) { goto save_error; } } } } - fclose(file); + + /* If the file we want to write already exists it could be possible that it + * was saved with a newer version. If that is the case we just append its + * content to the new output file and that way ensure that we don't lose any + * data the new version wrote. + */ + if ((in = fopen(file_name, "r")) != NULL) { + if (!fseek(in, ftell(out), SEEK_SET)) { + char buff[512]; + size_t read; + while ((read = fread(&buff, sizeof(char), sizeof(buff), in)) != 0) + fwrite(buff, sizeof(char), read, out); + if (ferror(in) || ferror(out)) + fl_alert("Error appending %s to %s", file_name, tmpc); + } + fclose(in); + } + + fclose(out); + + if (rename(tmpc, file_name)) + fl_alert("Error renaming %s to %s", tmpc, file_name); + ::remove(tmpc); + return; save_error: - fclose(file); + fclose(out); fl_alert("Error saving presets to file %s", file_name); return; } -- 2.27.0 >From 2dbf462a9126da817db40f4d62207b47e245f3e7 Mon Sep 17 00:00:00 2001 From: Jasmin Fazlic <superfassl(a)gmail.com> Date: Thu, 4 Feb 2021 17:00:22 +0100 Subject: [PATCH 2/2] alsa-tools/hdspmixer: add output loopback buttons This patch adds "LPBK" buttons to the output strip channels allowing to toggle the hardware output loopback state. Efforts were made to hide the buttons for not yet enabled devices but due to difficulties in implementation was cancelled for the time being. Should it be a no-go efforts should be made to fix this and hide them, although I would think fixing the driver for other devices would be an effort with much more merit. This is currently only enabled for HDSP9632 devices with following driver patch: https://github.com/tiwai/sound/commit/da2a040ee7cfe1dd57d5bec7906cb979c5787… Signed-off-by: Jasmin Fazlic <superfassl(a)gmail.com> --- hdspmixer/pixmaps/loopback.xpm | 69 ++++ hdspmixer/pixmaps/output.xpm | 498 +++++++++++++++----------- hdspmixer/pixmaps/output_r.xpm | 498 +++++++++++++++----------- hdspmixer/src/HDSPMixerCard.cxx | 34 ++ hdspmixer/src/HDSPMixerCard.h | 2 + hdspmixer/src/HDSPMixerLoopback.cxx | 133 +++++++ hdspmixer/src/HDSPMixerLoopback.h | 48 +++ hdspmixer/src/HDSPMixerOutput.cxx | 1 + hdspmixer/src/HDSPMixerOutput.h | 3 + hdspmixer/src/HDSPMixerOutputData.h | 1 + hdspmixer/src/HDSPMixerOutputs.cxx | 4 +- hdspmixer/src/HDSPMixerPresetData.cxx | 1 + hdspmixer/src/HDSPMixerPresetData.h | 1 + hdspmixer/src/HDSPMixerPresets.cxx | 2 + hdspmixer/src/HDSPMixerWindow.cxx | 68 ++++ hdspmixer/src/Makefile.am | 2 + hdspmixer/src/defines.h | 2 +- hdspmixer/src/pixmaps.cxx | 1 + hdspmixer/src/pixmaps.h | 1 + 19 files changed, 931 insertions(+), 438 deletions(-) create mode 100644 hdspmixer/pixmaps/loopback.xpm create mode 100644 hdspmixer/src/HDSPMixerLoopback.cxx create mode 100644 hdspmixer/src/HDSPMixerLoopback.h diff --git a/hdspmixer/pixmaps/loopback.xpm b/hdspmixer/pixmaps/loopback.xpm new file mode 100644 index 0000000..a327467 --- /dev/null +++ b/hdspmixer/pixmaps/loopback.xpm @@ -0,0 +1,69 @@ +/* XPM */ +char const * loopback_xpm[] = { +"34 15 51 1", +" c None", +". c #2E3038", +"+ c #FFFB7C", +"@ c #FDF97B", +"# c #2E2D11", +"$ c #000000", +"% c #383716", +"& c #898740", +"* c #F1EE75", +"= c #22210A", +"- c #727034", +"; c #E2DE6D", +"> c #D5D267", +", c #787637", +"' c #FBF77A", +") c #AAA750", +"! c #ABA851", +"~ c #CDC962", +"{ c #83813C", +"] c #C8C560", +"^ c #84823D", +"/ c #F7F378", +"( c #949145", +"_ c #B7B457", +": c #949245", +"< c #353414", +"[ c #87853F", +"} c #F1ED75", +"| c #3B3A17", +"1 c #DFDB6C", +"2 c #161505", +"3 c #4B4A20", +"4 c #E5E16E", +"5 c #FEFA7C", +"6 c #E8E470", +"7 c #535124", +"8 c #FAF679", +"9 c #E7E370", +"0 c #66642E", +"a c #E4E06E", +"b c #F6F277", +"c c #EAE671", +"d c #E0DC6C", +"e c #B9B658", +"f c #5F5D2A", +"g c #EDE973", +"h c #69672F", +"i c #DDD96A", +"j c #333342", +"k c #32323E", +"l c #313136", +"..................................", +".++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++.", +".+++@#++++@$$%&*@$$=-;+@#++>,*+++.", +".+++@#++++@#+')!@#+@~{+@#+]^/++++.", +".+++@#++++@#+'!!@#+@~(+@#_:'+++++.", +".+++@#++++@$$<[}@$$$|1+@234++++++.", +".+++@#++++@#++++@#+5678@#90a+++++.", +".+++@#++++@#++++@#+59#b@#+c0d++++.", +".+++@$$$$e@#++++@$$2f]+@#++ghi+++.", +".++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++.", +"..jklllllllllkj....jklllllllllkj.."}; diff --git a/hdspmixer/pixmaps/output.xpm b/hdspmixer/pixmaps/output.xpm index 8f7cd65..c827b1c 100644 --- a/hdspmixer/pixmaps/output.xpm +++ b/hdspmixer/pixmaps/output.xpm @@ -1,220 +1,282 @@ /* XPM */ char const * output_xpm[] = { -"36 208 9 1", -" c #595966", -". c #2E3038", -"+ c #000000", -"@ c #474951", -"# c #404044", -"$ c #7A7A8F", -"% c #27272B", -"& c #616176", -"* c #737384", -" ", -" .................................. 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", -" "}; +"36 224 55 1", +" c None", +". c #595966", +"+ c #2E3038", +"@ c #000000", +"# c #474951", +"$ c #404044", +"% c #7A7A8F", +"& c #27272B", +"* c #616176", +"= c #737384", +"- c #757585", +"; c #D4D4D4", +"> c #D6D6D6", +", c #D3D3D4", +"' c #C3C3C6", +") c #848491", +"! c #D5D5D5", +"~ c #CACACB", +"{ c #93939C", +"] c #9D9DA5", +"^ c #C8C8CA", +"/ c #787888", +"( c #B6B6BA", +"_ c #A3A3AA", +": c #C5C5C7", +"< c #A7A7AE", +"[ c #C4C4C6", +"} c #7D7D8C", +"| c #797989", +"1 c #C0C0C2", +"2 c #AFAFB4", +"3 c #BFBFC2", +"4 c #D3D3D3", +"5 c #95959F", +"6 c #D5D5D6", +"7 c #D1D1D2", +"8 c #91919B", +"9 c #757586", +"0 c #8E8E99", +"a c #D1D1D1", +"b c #8F8F99", +"c c #CBCBCD", +"d c #92929C", +"e c #7F7F8D", +"f c #8C8C97", +"g c #96969F", +"h c #AFAFB5", +"i c #CECECF", +"j c #A6A6AD", +"k c #8A8A96", +"l c #CCCCCD", +"m c #9898A1", +"n c #333342", +"o c #32323E", +"p c #313136", +"....................................", +".++++++++++++++++++++++++++++++++++.", 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+".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +"....................................", +".++++++++++++++++++++++++++++++++++.", +".+================================+.", +".+================================+.", +".+================================+.", +".+===-;====->>,')->>!~{=-;==]^)===+.", +".+===-;====-;=/((-;=-_:=-;=<[}====+.", +".+===-;====-;=|((-;=-_1=-;23/=====+.", +".+===-;====->>,[)->>>45=-678======+.", +".+===-;====-;====-;=90a|-;bcd=====+.", +".+===-;====-;====-;=9b;e-;=fcg====+.", +".+===->>>>h-;====->>6ij=-;==klm===+.", +".+================================+.", +".+================================+.", +".+================================+.", +".++nopppppppppon++++nopppppppppon++.", +"...................................."}; diff --git a/hdspmixer/src/HDSPMixerCard.cxx b/hdspmixer/src/HDSPMixerCard.cxx index ce40ba7..72232c8 100644 --- a/hdspmixer/src/HDSPMixerCard.cxx +++ b/hdspmixer/src/HDSPMixerCard.cxx @@ -231,6 +231,8 @@ void HDSPMixerCard::adjustSettings() { /* should never happen */ break; } + + max_channels = sizeof(channel_map_mf_ss); } if (type == Digiface) { @@ -253,6 +255,8 @@ void HDSPMixerCard::adjustSettings() { /* should never happen */ break; } + + max_channels = sizeof(channel_map_df_ss); } if (type == RPM) { @@ -263,6 +267,8 @@ void HDSPMixerCard::adjustSettings() { channel_map_input = channel_map_playback = channel_map_rpm; dest_map = dest_map_rpm; meter_map_input = meter_map_playback = channel_map_rpm; + + max_channels = sizeof(channel_map_rpm); } @@ -286,6 +292,8 @@ void HDSPMixerCard::adjustSettings() { /* should never happen */ break; } + + max_channels = sizeof(channel_map_df_ss); } if (type == H9632) { @@ -312,6 +320,8 @@ void HDSPMixerCard::adjustSettings() { meter_map_input = meter_map_playback = channel_map_h9632_qs; break; } + + max_channels = sizeof(channel_map_h9632_ss); } if (HDSPeMADI == type) { @@ -341,6 +351,7 @@ void HDSPMixerCard::adjustSettings() { break; } + max_channels = sizeof(channel_map_unity_ss); } if (HDSPeAIO == type) { @@ -379,6 +390,7 @@ void HDSPMixerCard::adjustSettings() { break; } + max_channels = sizeof(channel_map_aio_out_ss); } if (HDSP_AES == type) { @@ -394,6 +406,7 @@ void HDSPMixerCard::adjustSettings() { meter_map_input = channel_map_aes32; meter_map_playback = channel_map_aes32; + max_channels = sizeof(channel_map_aes32); } if (HDSPeRayDAT == type) { @@ -426,6 +439,7 @@ void HDSPMixerCard::adjustSettings() { break; } + max_channels = sizeof(channel_map_raydat_ss); } window_width = (channels_playback+2)*STRIP_WIDTH; @@ -545,3 +559,23 @@ int HDSPMixerCard::initializeCard(HDSPMixerWindow *w) return 0; } +int HDSPMixerCard::supportsLoopback() const +{ + int err = 0; + snd_ctl_elem_value_t *elemval; + snd_ctl_elem_id_t * elemid; + snd_ctl_t *handle; + snd_ctl_elem_value_alloca(&elemval); + snd_ctl_elem_id_alloca(&elemid); + if ((err = snd_ctl_open(&handle, name, SND_CTL_NONBLOCK)) < 0) + return err; + + snd_ctl_elem_id_set_name(elemid, "Output Loopback"); + snd_ctl_elem_id_set_interface(elemid, SND_CTL_ELEM_IFACE_HWDEP); + snd_ctl_elem_id_set_index(elemid, 0); + snd_ctl_elem_value_set_id(elemval, elemid); + err = snd_ctl_elem_read(handle, elemval); + snd_ctl_close(handle); + + return err; +} diff --git a/hdspmixer/src/HDSPMixerCard.h b/hdspmixer/src/HDSPMixerCard.h index faaeefa..eecfca3 100644 --- a/hdspmixer/src/HDSPMixerCard.h +++ b/hdspmixer/src/HDSPMixerCard.h @@ -52,6 +52,7 @@ public: HDSPMixerCard(int cardtype, int id, char *shortname); int channels_input, channels_playback, window_width, window_height, card_id; int channels_output; + int max_channels; int type; int last_preset; /* Last activated preset before switching to another card */ int last_dirty; /* Last dirty flag before switching to another card */ @@ -68,6 +69,7 @@ public: void adjustSettings(); void getAeb(); hdsp_9632_aeb_t h9632_aeb; + int supportsLoopback() const; }; #endif diff --git a/hdspmixer/src/HDSPMixerLoopback.cxx b/hdspmixer/src/HDSPMixerLoopback.cxx new file mode 100644 index 0000000..fcfec2a --- /dev/null +++ b/hdspmixer/src/HDSPMixerLoopback.cxx @@ -0,0 +1,133 @@ +/* + * HDSPMixer + * + * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#pragma implementation +#include "HDSPMixerLoopback.h" + +HDSPMixerLoopback::HDSPMixerLoopback(int x, int y, int idx):Fl_Widget(x, y, 34, 15) +{ + basew = (HDSPMixerWindow *)window(); + index = idx; +} + +void HDSPMixerLoopback::draw() +{ + if (_loopback == 1) + fl_draw_pixmap(loopback_xpm, x(), y()); +} + +int HDSPMixerLoopback::get() +{ + auto const card { basew->cards[basew->current_card] }; + + if (card->supportsLoopback() != 0) + return -1; + + if (index >= card->max_channels) + return -1; + + int err; + snd_ctl_elem_value_t *elemval; + snd_ctl_elem_id_t * elemid; + snd_ctl_t *handle; + snd_ctl_elem_value_alloca(&elemval); + snd_ctl_elem_id_alloca(&elemid); + char const * const name = basew->cards[basew->current_card]->name; + if ((err = snd_ctl_open(&handle, name, SND_CTL_NONBLOCK)) < 0) { + fprintf(stderr, "Error accessing ctl interface on card %s\n.", name); + return -1; + } + + snd_ctl_elem_id_set_name(elemid, "Output Loopback"); + snd_ctl_elem_id_set_interface(elemid, SND_CTL_ELEM_IFACE_HWDEP); + snd_ctl_elem_id_set_index(elemid, index); + snd_ctl_elem_value_set_id(elemval, elemid); + if ((err = snd_ctl_elem_read(handle, elemval)) < 0) + fprintf(stderr, "cannot read loopback: %d\n", err); + else + _loopback = snd_ctl_elem_value_get_integer(elemval, 0); + + snd_ctl_close(handle); + + return _loopback; +} + +void HDSPMixerLoopback::set(int l) +{ + auto const card { basew->cards[basew->current_card] }; + + if (card->supportsLoopback() != 0) + return; + + if (index >= card->max_channels) + return; + + if (l != _loopback) { + int err; + + snd_ctl_elem_id_t *id; + snd_ctl_elem_value_t *ctl; + snd_ctl_t *handle; + + snd_ctl_elem_value_alloca(&ctl); + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_id_set_name(id, "Output Loopback"); + snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_HWDEP); + snd_ctl_elem_id_set_device(id, 0); + snd_ctl_elem_id_set_index(id, index); + snd_ctl_elem_value_set_id(ctl, id); + + if ((err = snd_ctl_open( + &handle, basew->cards[basew->current_card]->name, SND_CTL_NONBLOCK)) < 0) { + fprintf(stderr, "Alsa error 1: %s\n", snd_strerror(err)); + return; + } + + snd_ctl_elem_value_set_integer(ctl, 0, l); + if ((err = snd_ctl_elem_write(handle, ctl)) < 0) { + fprintf(stderr, "Alsa error 2: %s\n", snd_strerror(err)); + snd_ctl_close(handle); + return; + } + + _loopback = l; + + snd_ctl_close(handle); + + redraw(); + } +} + +int HDSPMixerLoopback::handle(int e) +{ + int button3 = Fl::event_button3(); + switch (e) { + case FL_PUSH: + set(!_loopback); + if (button3) + relative->set(_loopback); + basew->checkState(); + redraw(); + return 1; + default: + return Fl_Widget::handle(e); + } +} + diff --git a/hdspmixer/src/HDSPMixerLoopback.h b/hdspmixer/src/HDSPMixerLoopback.h new file mode 100644 index 0000000..07f1f0c --- /dev/null +++ b/hdspmixer/src/HDSPMixerLoopback.h @@ -0,0 +1,48 @@ +/* + * HDSPMixer + * + * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#pragma interface +#ifndef HDSPMixerLoopback_H +#define HDSPMixerLoopback_H + +#include <FL/Fl.H> +#include <FL/Fl_Widget.H> +#include "HDSPMixerWindow.h" +#include "pixmaps.h" + +class HDSPMixerWindow; + +class HDSPMixerLoopback:public Fl_Widget +{ +private: + HDSPMixerWindow *basew; + int _loopback{-1}; +public: + HDSPMixerLoopback *relative; + int index; + HDSPMixerLoopback(int x, int y, int idx); + void draw(); + int handle(int e); + int get(); + void set(int l); +}; + +#endif + diff --git a/hdspmixer/src/HDSPMixerOutput.cxx b/hdspmixer/src/HDSPMixerOutput.cxx index 0053fe0..7f80a74 100644 --- a/hdspmixer/src/HDSPMixerOutput.cxx +++ b/hdspmixer/src/HDSPMixerOutput.cxx @@ -204,6 +204,7 @@ HDSPMixerOutput::HDSPMixerOutput(int x, int y, int w, int h, int num):Fl_Group(x peak = new HDSPMixerPeak(x+3, y+4, 0); gain = new HDSPMixerGain(x+3, y+175, 0); meter = new HDSPMixerMeter(x+20, y+27, false, peak); + loopback = new HDSPMixerLoopback(x+1, y+208, out_num); end(); } diff --git a/hdspmixer/src/HDSPMixerOutput.h b/hdspmixer/src/HDSPMixerOutput.h index 6278cfd..1e04ae5 100644 --- a/hdspmixer/src/HDSPMixerOutput.h +++ b/hdspmixer/src/HDSPMixerOutput.h @@ -27,6 +27,7 @@ #include <FL/fl_draw.H> #include <alsa/sound/hdsp.h> #include "HDSPMixerFader.h" +#include "HDSPMixerLoopback.h" #include "HDSPMixerPeak.h" #include "HDSPMixerGain.h" #include "HDSPMixerMeter.h" @@ -36,6 +37,7 @@ class HDSPMixerFader; class HDSPMixerGain; +class HDSPMixerLoopback; class HDSPMixerPeak; class HDSPMixerMeter; class HDSPMixerOutputData; @@ -56,6 +58,7 @@ public: HDSPMixerFader *fader; HDSPMixerGain *gain; HDSPMixerMeter *meter; + HDSPMixerLoopback *loopback; HDSPMixerOutput(int x, int y, int w, int h, int out); void draw(); void draw_background(); diff --git a/hdspmixer/src/HDSPMixerOutputData.h b/hdspmixer/src/HDSPMixerOutputData.h index 885047b..75b9109 100644 --- a/hdspmixer/src/HDSPMixerOutputData.h +++ b/hdspmixer/src/HDSPMixerOutputData.h @@ -26,6 +26,7 @@ class HDSPMixerOutputData { public: int fader_pos; + int loopback; HDSPMixerOutputData(); }; diff --git a/hdspmixer/src/HDSPMixerOutputs.cxx b/hdspmixer/src/HDSPMixerOutputs.cxx index 0b4e7f4..342685d 100644 --- a/hdspmixer/src/HDSPMixerOutputs.cxx +++ b/hdspmixer/src/HDSPMixerOutputs.cxx @@ -25,13 +25,15 @@ HDSPMixerOutputs::HDSPMixerOutputs(int x, int y, int w, int h, int nchans):Fl_Gr { int i; for (i = 0; i < HDSP_MAX_CHANNELS+2; i += 2) { - strips[i] = new HDSPMixerOutput((i*STRIP_WIDTH), y, STRIP_WIDTH, SMALLSTRIP_HEIGHT, i); + strips[i] = new HDSPMixerOutput((i*STRIP_WIDTH), y, STRIP_WIDTH, SMALLSTRIP_HEIGHT, i); strips[i+1] = new HDSPMixerOutput(((i+1)*STRIP_WIDTH), y, STRIP_WIDTH, SMALLSTRIP_HEIGHT, i+1); /* Setup linked stereo channels */ strips[i]->fader->relative = strips[i+1]->fader; strips[i+1]->fader->relative = strips[i]->fader; strips[i]->fader->gain = strips[i]->gain; strips[i+1]->fader->gain = strips[i+1]->gain; + strips[i]->loopback->relative = strips[i+1]->loopback; + strips[i+1]->loopback->relative = strips[i]->loopback; } empty_aebo[0] = new HDSPMixerEmpty((nchans-6)*STRIP_WIDTH, y, 2*STRIP_WIDTH, SMALLSTRIP_HEIGHT, 0); diff --git a/hdspmixer/src/HDSPMixerPresetData.cxx b/hdspmixer/src/HDSPMixerPresetData.cxx index 276d101..6359732 100644 --- a/hdspmixer/src/HDSPMixerPresetData.cxx +++ b/hdspmixer/src/HDSPMixerPresetData.cxx @@ -36,5 +36,6 @@ HDSPMixerPresetData::HDSPMixerPresetData() over = 3; level = 0; rate = 1; + loopback = 0; } diff --git a/hdspmixer/src/HDSPMixerPresetData.h b/hdspmixer/src/HDSPMixerPresetData.h index 58536d3..3b46b92 100644 --- a/hdspmixer/src/HDSPMixerPresetData.h +++ b/hdspmixer/src/HDSPMixerPresetData.h @@ -38,6 +38,7 @@ public: int over; int rate; int rmsplus3; + int loopback; HDSPMixerPresetData(); }; diff --git a/hdspmixer/src/HDSPMixerPresets.cxx b/hdspmixer/src/HDSPMixerPresets.cxx index aeeb9c9..82654d9 100644 --- a/hdspmixer/src/HDSPMixerPresets.cxx +++ b/hdspmixer/src/HDSPMixerPresets.cxx @@ -143,6 +143,7 @@ void HDSPMixerPresets::save_preset(int prst) { basew->playbacks->strips[i]->data[card][speed][p]->dest = basew->playbacks->strips[i]->targets->selected; basew->outputs->strips[i]->data[card][speed][p]->fader_pos = basew->outputs->strips[i]->fader->pos[0]; + basew->outputs->strips[i]->data[card][speed][p]->loopback = basew->outputs->strips[i]->loopback->get(); } /* Line outs */ basew->outputs->strips[HDSP_MAX_CHANNELS]->data[card][speed][p]->fader_pos = basew->outputs->strips[HDSP_MAX_CHANNELS]->fader->pos[0]; @@ -188,6 +189,7 @@ void HDSPMixerPresets::restore_preset(int prst) { basew->playbacks->strips[i]->targets->selected = basew->playbacks->strips[i]->data[card][speed][p]->dest; basew->outputs->strips[i]->fader->pos[0] = basew->outputs->strips[i]->data[card][speed][p]->fader_pos; + basew->outputs->strips[i]->loopback->set(basew->outputs->strips[i]->data[card][speed][p]->loopback); } /* Line outs */ basew->outputs->strips[HDSP_MAX_CHANNELS]->fader->pos[0] = basew->outputs->strips[HDSP_MAX_CHANNELS+1]->data[card][speed][p]->fader_pos; diff --git a/hdspmixer/src/HDSPMixerWindow.cxx b/hdspmixer/src/HDSPMixerWindow.cxx index 3b3d668..4a911c1 100644 --- a/hdspmixer/src/HDSPMixerWindow.cxx +++ b/hdspmixer/src/HDSPMixerWindow.cxx @@ -471,6 +471,29 @@ void HDSPMixerWindow::save() } } + /* Output loopback data */ + for (int channel = 0; channel < HDSP_MAX_CHANNELS; ++channel) { + auto const strip = outputs->strips[channel]; + + for (int card = 0; card < MAX_CARDS; ++card) { + auto const data = strip->data[card]; + + for (int speed = 0; speed < 3; ++speed) { + auto const spd = data[speed]; + + for (int preset = 0; preset < 8; ++preset) { + auto const data = spd[preset]; + + if (fwrite((void *)&(data->loopback), + sizeof(int), + 1, + out) != 1) + goto save_error; + } + } + } + } + /* If the file we want to write already exists it could be possible that it * was saved with a newer version. If that is the case we just append its * content to the new output file and that way ensure that we don't lose any @@ -519,6 +542,7 @@ void HDSPMixerWindow::load() bool ondisk_v1 = false; int pan_array_size = 14; /* old (pre 1.0.24) HDSP_MAX_DEST */ int channels_per_card = 26; /* old (pre 1.0.24) HDSP_MAX_CHANNELS */ + bool res = true; if (fread(&buffer, sizeof(char), sizeof(buffer), file) != sizeof(buffer)) { goto load_error; @@ -647,6 +671,46 @@ void HDSPMixerWindow::load() } } } + + /* Output loopback data */ + for (int channel = 0; channel < HDSP_MAX_CHANNELS; ++channel) { + auto const strip = outputs->strips[channel]; + + for (int card = 0; card < MAX_CARDS; ++card) { + auto const data = strip->data[card]; + + for (int speed = 0; speed < 3; ++speed) { + auto const spd = data[speed]; + + for (int preset = 0; preset < 8; ++preset) { + auto const data = spd[preset]; + + /* TODO: Somewhere we get a value of 5 from, investigate + * this another day. For now just reset it here and + * continue looping to reset the value. + */ + data->loopback = 0; + + if (feof(file)) { + res = true; + continue; + } + + if (ferror(file)) { + res = false; + continue; + } + + if (fread((void *)&(data->loopback), sizeof(int), 1, file) != 1) + res = false; + } + } + } + } + + if (!res) + goto load_error; + fclose(file); setTitleWithFilename(); resetMixer(); @@ -844,6 +908,8 @@ void HDSPMixerWindow::restoreDefaults(int card) } outputs->strips[i]->data[card][speed][preset]->fader_pos = (preset != 4) ? 137*CF : 0; outputs->strips[i+1]->data[card][speed][preset]->fader_pos = (preset != 4) ? 137*CF : 0; + outputs->strips[i]->data[card][speed][preset]->loopback = 0; + outputs->strips[i+1]->data[card][speed][preset]->loopback = 0; if (preset == 3 || preset == 7) { inputs->strips[i]->data[card][speed][preset]->mute = 1; inputs->strips[i+1]->data[card][speed][preset]->mute = 1; @@ -1051,6 +1117,8 @@ void HDSPMixerWindow::checkState() /* Outputs row */ if (outputs->strips[i]->data[current_card][speed][p]->fader_pos != outputs->strips[i]->fader->pos[0]) corrupt++; + if (outputs->strips[i]->data[current_card][speed][p]->loopback != outputs->strips[i]->loopback->get()) + corrupt++; } /* Global settings */ diff --git a/hdspmixer/src/Makefile.am b/hdspmixer/src/Makefile.am index e80a8ac..ebc4c3c 100644 --- a/hdspmixer/src/Makefile.am +++ b/hdspmixer/src/Makefile.am @@ -15,6 +15,8 @@ hdspmixer_SOURCES = \ HDSPMixerEmpty.h \ HDSPMixerOutput.cxx \ HDSPMixerOutput.h \ + HDSPMixerLoopback.cxx \ + HDSPMixerLoopback.h \ HDSPMixerIOMixer.cxx \ HDSPMixerIOMixer.h \ HDSPMixerSelector.cxx \ diff --git a/hdspmixer/src/defines.h b/hdspmixer/src/defines.h index af5c382..e76141b 100644 --- a/hdspmixer/src/defines.h +++ b/hdspmixer/src/defines.h @@ -34,7 +34,7 @@ #define STRIP_WIDTH 36 #define FULLSTRIP_HEIGHT 253 -#define SMALLSTRIP_HEIGHT 208 +#define SMALLSTRIP_HEIGHT 224 #define MENU_HEIGHT 20 #define MIN_WIDTH 2*STRIP_WIDTH diff --git a/hdspmixer/src/pixmaps.cxx b/hdspmixer/src/pixmaps.cxx index 2f7c589..34f8834 100644 --- a/hdspmixer/src/pixmaps.cxx +++ b/hdspmixer/src/pixmaps.cxx @@ -47,6 +47,7 @@ #include "../pixmaps/over.xpm" #include "../pixmaps/peak.xpm" #include "../pixmaps/solo.xpm" +#include "../pixmaps/loopback.xpm" #include "../pixmaps/iomixer_r.xpm" #include "../pixmaps/output_r.xpm" #include "../pixmaps/matrix_black.xpm" diff --git a/hdspmixer/src/pixmaps.h b/hdspmixer/src/pixmaps.h index b980a62..3548ba3 100644 --- a/hdspmixer/src/pixmaps.h +++ b/hdspmixer/src/pixmaps.h @@ -49,6 +49,7 @@ extern char const * output_xpm[]; extern char const * over_xpm[]; extern char const * peak_xpm[]; extern char const * solo_xpm[]; +extern char const * loopback_xpm[]; extern char const * iomixer_r_xpm[]; extern char const * output_r_xpm[]; extern char const * matrix_white_xpm[]; -- 2.27.0

3 5

[PATCH] ALSA: usb-audio: Don't abort even if the clock rate differs
by Takashi Iwai 27 Feb '21

27 Feb '21

The commit 93db51d06b32 ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3") changed the behavior of the function set_sample_rate_v2v3() slightly to treat the inconsistent sample rate as an error. It was done by assumption that the sample rate validation should have been done at the parser phase as implemented in that patch. But the validation is later selectively enabled only for certain devices as it causes a regression (the commit fe773b8711e3 "ALSA: usb-audio: workaround for iface reset issue"), and now the inconsistency surfaced as a fatal error while it worked in the past as is, as reported for FiiO M3K DAC. For recovering from the regression, change set_sample_rate_v2v3() again to ignore the sample rate difference as non-error. BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1182633 Fixes: 93db51d06b32 ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 8243652d5604..a746802d0ac3 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -652,10 +652,10 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, cur_rate = prev_rate; if (cur_rate != rate) { - usb_audio_warn(chip, - "%d:%d: freq mismatch (RO clock): req %d, clock runs @%d\n", - fmt->iface, fmt->altsetting, rate, cur_rate); - return -ENXIO; + usb_audio_dbg(chip, + "%d:%d: freq mismatch: req %d, clock runs @%d\n", + fmt->iface, fmt->altsetting, rate, cur_rate); + /* continue processing */ } validation: -- 2.26.2

1 0

[PATCH v5 0/9] ALSA: add virtio sound driver
by Anton Yakovlev 27 Feb '21

27 Feb '21

This series implements a driver part of the virtio sound device specification v8 [1]. The driver supports PCM playback and capture substreams, jack and channel map controls. A message-based transport is used to write/read PCM frames to/from a device. As a device part was used OpenSynergy proprietary implementation. v5 changes: - Fixed another bunch of sparse warnings (replaced virtio_cread() -> virtio_cread_le()), no functional changes. (Sorry, I didn't know how to run sparse locally, now everything should be fixed) [1] https://lists.oasis-open.org/archives/virtio-dev/202003/msg00185.html Anton Yakovlev (9): uapi: virtio_ids: add a sound device type ID from OASIS spec ALSA: virtio: add virtio sound driver ALSA: virtio: handling control messages ALSA: virtio: build PCM devices and substream hardware descriptors ALSA: virtio: handling control and I/O messages for the PCM device ALSA: virtio: PCM substream operators ALSA: virtio: introduce jack support ALSA: virtio: introduce PCM channel map support ALSA: virtio: introduce device suspend/resume support MAINTAINERS | 9 + include/uapi/linux/virtio_ids.h | 1 + include/uapi/linux/virtio_snd.h | 334 +++++++++++++++++++++ sound/Kconfig | 2 + sound/Makefile | 3 +- sound/virtio/Kconfig | 10 + sound/virtio/Makefile | 13 + sound/virtio/virtio_card.c | 462 +++++++++++++++++++++++++++++ sound/virtio/virtio_card.h | 113 ++++++++ sound/virtio/virtio_chmap.c | 219 ++++++++++++++ sound/virtio/virtio_ctl_msg.c | 310 ++++++++++++++++++++ sound/virtio/virtio_ctl_msg.h | 78 +++++ sound/virtio/virtio_jack.c | 233 +++++++++++++++ sound/virtio/virtio_pcm.c | 498 ++++++++++++++++++++++++++++++++ sound/virtio/virtio_pcm.h | 120 ++++++++ sound/virtio/virtio_pcm_msg.c | 392 +++++++++++++++++++++++++ sound/virtio/virtio_pcm_ops.c | 491 +++++++++++++++++++++++++++++++ 17 files changed, 3287 insertions(+), 1 deletion(-) create mode 100644 include/uapi/linux/virtio_snd.h create mode 100644 sound/virtio/Kconfig create mode 100644 sound/virtio/Makefile create mode 100644 sound/virtio/virtio_card.c create mode 100644 sound/virtio/virtio_card.h create mode 100644 sound/virtio/virtio_chmap.c create mode 100644 sound/virtio/virtio_ctl_msg.c create mode 100644 sound/virtio/virtio_ctl_msg.h create mode 100644 sound/virtio/virtio_jack.c create mode 100644 sound/virtio/virtio_pcm.c create mode 100644 sound/virtio/virtio_pcm.h create mode 100644 sound/virtio/virtio_pcm_msg.c create mode 100644 sound/virtio/virtio_pcm_ops.c -- 2.30.0

3 27

ASoc: soc_core.c stream direction from snd_soc_dai
by Matt Flax 26 Feb '21

26 Feb '21

Hi there, A large number of audio codecs allow different formats for playback and capture. This becomes very useful when there are different latencies between playback and capture hardware data lines. For example digital isolation chips typically have a 1 bit delay in propagation as the bit clock rate gets faster for higher sample rates. By setting the capture and playback formats to differ by one or two bit clock cycles, the delay problem is solved. There doesn't seem to be a simple way to detect stream direction in the codec driver's set_fmt function. The snd_soc_runtime_set_dai_fmt function : https://github.com/torvalds/linux/blob/master/sound/soc/soc-core.c#L1480 calls the snd_soc_dai_set_fmt function : https://github.com/torvalds/linux/blob/master/sound/soc/soc-dai.c#L101 which calls the set_fmt function : https://github.com/torvalds/linux/blob/master/include/sound/soc-dai.h#L189 The snd_soc_dai_ops set_fmt function is defined as : int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); Is there a simple way to find the stream direction from a snd_soc_dai ? If the stream direction can be detected then the playback and capture formats can be set independently for the codec. It there a different way to set the playback and capture formats for the codec independently at runtime, depending on the sample rate ? Matt

5 8

[PATCH alsa-ucm-conf 0/8] codecs/rt5640: Cleanup + HW volume control support
by Hans de Goede 26 Feb '21

26 Feb '21

Hi All, Here is a patch series consisting of some codecs/rt5640 cleanups and addition of HW volume control support to the rt5640 .conf snippets. Note patch 7 and 8 depend on a new "aif:%d" part being added to the components string, the kernel patches for this are pending upstream in the "[PATCH 0/5] AsoC: rt5640/rt5651: Volume control fixes" patch series. Patches 7 and 8 should probably not be merged until the kernel patches have landed, because theoretically the new component string part could change as part of the review of the kernel patches. Regards, Hans Hans de Goede (8): codecs/rt5640: Cleanup: unify Stereo / Mono ADC handling codecs/rt5640: Cleanup: Initially disable all inputs and outputs codecs/rt5640: Cleanup: Move 'SPK MIX' setup to main EnableSequene codecs/rt5640: Drop bogus 'DIG MIXL DAC L2 Switch' setting codecs/rt5640: Unify capture volume for AIF1 and AIF2 recording paths codecs/rt5640: Add hardware volume-control support codecs/rt5640: Use the new "aif:%d" part of the components string codecs/rt5640: Specify Playback/CaptureMasterElem for HW volume-control ucm2/bytcr-rt5640/HiFi.conf | 46 +++++++++++++++++ ucm2/codecs/rt5640/DigitalMics.conf | 18 ++++++- ucm2/codecs/rt5640/EnableSeq-AIF-unknown.conf | 7 +++ ucm2/codecs/rt5640/EnableSeq-AIF1.conf | 7 +++ ucm2/codecs/rt5640/EnableSeq-AIF2.conf | 7 +++ ucm2/codecs/rt5640/EnableSeq.conf | 50 ++++++++++++++----- ucm2/codecs/rt5640/HeadPhones.conf | 15 +++++- ucm2/codecs/rt5640/HeadsetMic.conf | 24 ++++++--- ucm2/codecs/rt5640/IN1-InternalMic.conf | 27 ++++++---- ucm2/codecs/rt5640/IN3-InternalMic.conf | 27 ++++++---- ucm2/codecs/rt5640/MonoSpeaker.conf | 17 +++++-- ucm2/codecs/rt5640/Speaker.conf | 17 +++++-- 12 files changed, 211 insertions(+), 51 deletions(-) create mode 100644 ucm2/codecs/rt5640/EnableSeq-AIF-unknown.conf create mode 100644 ucm2/codecs/rt5640/EnableSeq-AIF1.conf create mode 100644 ucm2/codecs/rt5640/EnableSeq-AIF2.conf -- 2.30.1

1 8

Possible alsa bug around USB drivers
by Steven Skeard 25 Feb '21

25 Feb '21

Hey all, The last few weeks I've been plagued by inconsistent audio on my PopOS 20.10 desktop. Sometimes on boot everything would be fine, but other times I would have no working audio and only dummy devices in the audio menus. I eventually found that if I specify the modprobe order so that the USB driver (eg. for my webcam) loads after the onboard motherboard driver, that audio will work consistently. Here are the related links: - Reddit thread where I posted the original problem and the solution: https://www.reddit.com/r/pop_os/comments/lmvbj3/audio_not_working_after_upg… - A related thread where it sounds like an arch user has the same problem: https://bbs.archlinux.org/viewtopic.php?id=260373 - The alsa info I uploaded while under the effects of the problem: http://alsa-project.org/db/?f=a23ccbb818fec6eaf08b032504eac5d5d8a30589 I'm not really sure where this problem originated but I figured you guys might be able to point me in the right direction if it is not an alsa problem. I don't think I messed with the original alsa-base.conf but I suppose my memory could be failing me here. I'm not sure where this file comes from but I'm guessing it could be either a pop os, ubuntu, or alsa default configuration? Thanks, Steven Skeard

1 0

[PATCH] ASoC: rt1015p: add acpi device id for rt1015p
by jack.yu＠realtek.com 25 Feb '21

25 Feb '21

From: Jack Yu <jack.yu(a)realtek.com> Add acpi device id for rt1015p. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015p.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/rt1015p.c b/sound/soc/codecs/rt1015p.c index 671f2a2130fe..4a00e7a4ef6f 100644 --- a/sound/soc/codecs/rt1015p.c +++ b/sound/soc/codecs/rt1015p.c @@ -4,6 +4,7 @@ // // Copyright 2020 The Linux Foundation. All rights reserved. +#include <linux/acpi.h> #include <linux/delay.h> #include <linux/device.h> #include <linux/err.h> @@ -130,10 +131,21 @@ static const struct of_device_id rt1015p_device_id[] = { MODULE_DEVICE_TABLE(of, rt1015p_device_id); #endif +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt1015p_acpi_match[] = { + { "RT1015", 0}, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt1015p_acpi_match); +#endif + static struct platform_driver rt1015p_platform_driver = { .driver = { .name = "rt1015p", .of_match_table = of_match_ptr(rt1015p_device_id), +#ifdef CONFIG_ACPI + .acpi_match_table = ACPI_PTR(rt1015p_acpi_match), +#endif }, .probe = rt1015p_platform_probe, }; -- 2.29.0

3 2

[PATCH] ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 10-p0XX OVCD current threshold
by Hans de Goede 25 Feb '21

25 Feb '21

When I added the quirk for the "HP Pavilion x2 10-p0XX" I copied the byt_rt5640_quirk_table[] entry for the HP Pavilion x2 10-k0XX / 10-n0XX models since these use almost the same settings. While doing this I accidentally also copied and kept the non-standard OVCD_TH_1500UA setting used on those models. This too low threshold is causing headsets to often be seen as headphones (without a headset-mic) and when correctly identified it is causing ghost play/pause button-presses to get detected. Correct the HP Pavilion x2 10-p0XX quirk to use the default OVCD_TH_2000UA setting, fixing these problems. Fixes: fbdae7d6d04d ("ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 Detachable quirks") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 24ced8e4a574..1f6a636571c2 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -581,7 +581,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { }, .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | BYT_RT5640_JD_SRC_JD1_IN4P | - BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MCLK_EN), }, -- 2.30.1

3 2

[PATCH v2] ALSA: hda/realtek: Enable headset mic of Acer SWIFT with ALC256
by chris.chiu＠canonical.com 25 Feb '21

25 Feb '21

From: Chris Chiu <chris.chiu(a)canonical.com> The Acer SWIFT Swift SF314-54/55 laptops with ALC256 cannot detect both the headset mic and the internal mic. Introduce new fixup to enable the jack sense and the headset mic. However, the internal mic actually connects to Intel SST audio. It still needs Intel SST support to make internal mic capture work. Signed-off-by: Chris Chiu <chris.chiu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1927605f0f7e..4871507cd4bf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6406,6 +6406,7 @@ enum { ALC236_FIXUP_DELL_AIO_HEADSET_MIC, ALC282_FIXUP_ACER_DISABLE_LINEOUT, ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, + ALC256_FIXUP_ACER_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -7853,6 +7854,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_ACER_MIC_NO_PRESENCE, }, + [ALC256_FIXUP_ACER_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x02a1113c }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x90a1092f }, /* use as internal mic */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7879,9 +7890,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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), + SND_PCI_QUIRK(0x1025, 0x1269, "Acer SWIFT SF314-54", ALC256_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x129c, "Acer SWIFT SF314-55", ALC256_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), -- 2.20.1

1 0

[PATCH] ALSA: hda/realtek: Enable MICs of Acer SWIFT with ALC256
by Chris Chiu 25 Feb '21

25 Feb '21

The Acer SWIFT Swift SF314-54/55 with ALC256 cannot detect the headset microphone and suffers the noise problem in audio capture. This patch enables the headset jack sense and fixes the noise problem with aamix fixup. Signed-off-by: Chris Chiu <chris.chiu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290645516313..57e4dbcd76a9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6372,6 +6372,8 @@ enum { ALC236_FIXUP_DELL_AIO_HEADSET_MIC, ALC282_FIXUP_ACER_DISABLE_LINEOUT, ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, + ALC256_FIXUP_ACER_SWIFT_NO_MIC_PRESENCE, + ALC256_FIXUP_ACER_SWIFT_DISABLE_AAMIX, }; static const struct hda_fixup alc269_fixups[] = { @@ -7815,6 +7817,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_ACER_MIC_NO_PRESENCE, }, + [ALC256_FIXUP_ACER_SWIFT_NO_MIC_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x02a1113c }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x90a1092f }, /* use as internal mic */ + { }, + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, + [ALC256_FIXUP_ACER_SWIFT_DISABLE_AAMIX] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC256_FIXUP_ACER_SWIFT_NO_MIC_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7841,9 +7859,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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), + SND_PCI_QUIRK(0x1025, 0x1269, "Acer SWIFT SF314-54", ALC256_FIXUP_ACER_SWIFT_DISABLE_AAMIX), SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x129c, "Acer SWIFT SF314-55", ALC256_FIXUP_ACER_SWIFT_DISABLE_AAMIX), SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), -- 2.20.1

2 2

[PATCH] ALSA: hda/hdmi: let new platforms assign the pcm slot dynamically
by Hui Wang 25 Feb '21

25 Feb '21

If the platform set the dyn_pcm_assign to true, it will call hdmi_find_pcm_slot() to find a pcm slot when hdmi/dp monitor is connected and need to create a pcm. So far only intel_hsw_common_init() and patch_nvhdmi() set the dyn_pcm_assign to true, here we let tgl platforms assign the pcm slot dynamically first, if the driver runs for a period of time and there is no regression reported, we could set no_fixed_assgin to true in the intel_hsw_common_init(), and then set it to true in the patch_nvhdmi(). This change comes from the discussion between Takashi and Kai Vehmanen. Please refer to: https://github.com/alsa-project/alsa-lib/pull/118 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e405be7929e3..379a4d786b3b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -157,6 +157,7 @@ struct hdmi_spec { bool dyn_pin_out; bool dyn_pcm_assign; + bool no_fixed_assign; bool intel_hsw_fixup; /* apply Intel platform-specific fixups */ /* * Non-generic VIA/NVIDIA specific @@ -1345,6 +1346,12 @@ static int hdmi_find_pcm_slot(struct hdmi_spec *spec, { int i; + /* on the new machines, try to assign the pcm slot dynamically, + * not use the preferred fixed map anymore. + */ + if (spec->no_fixed_assign) + goto last_try; + /* * generic_hdmi_build_pcms() may allocate extra PCMs on some * platforms (with maximum of 'num_nids + dev_num - 1') @@ -1374,6 +1381,7 @@ static int hdmi_find_pcm_slot(struct hdmi_spec *spec, return i; } + last_try: /* the last try; check the empty slots in pins */ for (i = 0; i < spec->num_nids; i++) { if (!test_bit(i, &spec->pcm_bitmap)) @@ -2937,6 +2945,9 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, spec->port_num = port_num; spec->intel_hsw_fixup = true; + if (port_num > 6) + spec->no_fixed_assign = true; + intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); -- 2.25.1

4 10

[PATCH 0/4] ASoC: rt5670: Various kcontrol fixes
by Hans de Goede 24 Feb '21

24 Feb '21

Hi All, While working on adding hardware-volume control support to the UCM profile for the rt5672 and on adding LED trigger support to the rt5670 codec driver. I hit / noticed a couple of issues this series fixes these issues. Regards, Hans Hans de Goede (4): ASoC: rt5670: Remove 'OUT Channel Switch' control ASoC: rt5670: Remove 'HP Playback Switch' control ASoC: rt5670: Remove ADC vol-ctrl mute bits poking from Sto1 ADC mixer settings ASoC: rt5670: Add emulated 'DAC1 Playback Switch' control sound/soc/codecs/rt5670.c | 110 +++++++++++++++++++++++++++++++++----- sound/soc/codecs/rt5670.h | 9 ++-- 2 files changed, 101 insertions(+), 18 deletions(-) -- 2.30.1

3 8

[PATCH 1/2] ASoC: ak4458: Add MODULE_DEVICE_TABLE
by Shengjiu Wang 24 Feb '21

24 Feb '21

Add missed MODULE_DEVICE_TABLE for the driver can be loaded automatically at boot. Fixes: 08660086eff9 ("ASoC: ak4458: Add support for AK4458 DAC driver") Cc: <stable(a)vger.kernel.org> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/ak4458.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 472caad17012..85a1d00894a9 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -812,6 +812,7 @@ static const struct of_device_id ak4458_of_match[] = { { .compatible = "asahi-kasei,ak4497", .data = &ak4497_drvdata}, { }, }; +MODULE_DEVICE_TABLE(of, ak4458_of_match); static struct i2c_driver ak4458_i2c_driver = { .driver = { -- 2.27.0

2 2

[PATCH AUTOSEL 4.19 26/26] ASoC: Intel: bytcr_rt5640: Add quirk for the Acer One S1002 tablet
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit c58947af08aedbdee0fce5ea6e6bf3e488ae0e2c ] The Acer One S1002 tablet is using an analog mic on IN1 and has its jack-detect connected to JD2_IN4N, instead of using the default IN3 for its internal mic and JD1_IN4P for jack-detect. Note it is also using AIF2 instead of AIF1 which is somewhat unusual, this is correctly advertised in the ACPI CHAN package, so the speakers do work without the quirk. Add a quirk for the mic and jack-detect settings. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210216213555.36555-5-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 910214ab140e5..8a943de1e5b55 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -409,6 +409,19 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Acer One 10 S1002 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Acer"), + DMI_MATCH(DMI_PRODUCT_NAME, "One S1002"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Acer"), -- 2.27.0

1 0

[PATCH AUTOSEL 4.19 25/26] ASoC: Intel: bytcr_rt5640: Add quirk for the Voyo Winpad A15 tablet
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit e1317cc9ca4ac20262895fddb065ffda4fc29cfb ] The Voyo Winpad A15 tablet uses a Bay Trail (non CR) SoC, so it is using SSP2 (AIF1) and it mostly works with the defaults. But instead of using DMIC1 it is using an analog mic on IN1, add a quirk for this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210216213555.36555-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 4dd1941d4147f..910214ab140e5 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -756,6 +756,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, + { /* Voyo Winpad A15 */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Aptio CRB"), + /* Above strings are too generic, also match on BIOS date */ + DMI_MATCH(DMI_BIOS_DATE, "11/20/2014"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), -- 2.27.0

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[PATCH AUTOSEL 4.19 24/26] ASoC: Intel: bytcr_rt5640: Add quirk for the Estar Beauty HD MID 7316R tablet
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit bdea43fc0436c9e98fdfe151c2ed8a3fc7277404 ] The Estar Beauty HD MID 7316R tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210216213555.36555-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index ec630127ef2f3..4dd1941d4147f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -510,6 +510,16 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MONO_SPEAKER | BYT_RT5640_MCLK_EN), }, + { /* Estar Beauty HD MID 7316R */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Estar"), + DMI_MATCH(DMI_PRODUCT_NAME, "eSTAR BEAUTY HD Intel Quad core"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.27.0

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[PATCH AUTOSEL 5.4 40/40] ASoC: Intel: bytcr_rt5640: Add quirk for the Acer One S1002 tablet
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit c58947af08aedbdee0fce5ea6e6bf3e488ae0e2c ] The Acer One S1002 tablet is using an analog mic on IN1 and has its jack-detect connected to JD2_IN4N, instead of using the default IN3 for its internal mic and JD1_IN4P for jack-detect. Note it is also using AIF2 instead of AIF1 which is somewhat unusual, this is correctly advertised in the ACPI CHAN package, so the speakers do work without the quirk. Add a quirk for the mic and jack-detect settings. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210216213555.36555-5-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 00e8d589a7246..9ee610504bac9 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -400,6 +400,19 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Acer One 10 S1002 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Acer"), + DMI_MATCH(DMI_PRODUCT_NAME, "One S1002"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Acer"), -- 2.27.0

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[PATCH AUTOSEL 5.4 39/40] ASoC: Intel: bytcr_rt5651: Add quirk for the Jumper EZpad 7 tablet
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit df8359c512fa770ffa6b0b0309807d9b9825a47f ] Add a DMI quirk for the Jumper EZpad 7 tablet, this tablet has a jack-detect switch which reads 1/high when a jack is inserted, rather then using the standard active-low setup which most jack-detect switches use. All other settings are using the defaults. Add a DMI-quirk setting the defaults + the BYT_RT5651_JD_NOT_INV flags for this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210216213555.36555-4-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5651.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 4606f6f582d6f..921c09cdb4800 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -435,6 +435,19 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { BYT_RT5651_SSP0_AIF1 | BYT_RT5651_MONO_SPEAKER), }, + { + /* Jumper EZpad 7 */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Jumper"), + DMI_MATCH(DMI_PRODUCT_NAME, "EZpad"), + /* Jumper12x.WJ2012.bsBKRCP05 with the version dropped */ + DMI_MATCH(DMI_BIOS_VERSION, "Jumper12x.WJ2012.bsBKRCP"), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_JD_NOT_INV), + }, { /* KIANO SlimNote 14.2 */ .callback = byt_rt5651_quirk_cb, -- 2.27.0

1 0

[PATCH AUTOSEL 5.4 38/40] ASoC: Intel: bytcr_rt5640: Add quirk for the Voyo Winpad A15 tablet
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit e1317cc9ca4ac20262895fddb065ffda4fc29cfb ] The Voyo Winpad A15 tablet uses a Bay Trail (non CR) SoC, so it is using SSP2 (AIF1) and it mostly works with the defaults. But instead of using DMIC1 it is using an analog mic on IN1, add a quirk for this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210216213555.36555-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index cdbc00c773386..00e8d589a7246 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -786,6 +786,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, + { /* Voyo Winpad A15 */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Aptio CRB"), + /* Above strings are too generic, also match on BIOS date */ + DMI_MATCH(DMI_BIOS_DATE, "11/20/2014"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), -- 2.27.0

1 0

[PATCH AUTOSEL 5.4 37/40] ASoC: Intel: bytcr_rt5640: Add quirk for the Estar Beauty HD MID 7316R tablet
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit bdea43fc0436c9e98fdfe151c2ed8a3fc7277404 ] The Estar Beauty HD MID 7316R tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210216213555.36555-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 6012367f6fe48..cdbc00c773386 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -513,6 +513,16 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MONO_SPEAKER | BYT_RT5640_MCLK_EN), }, + { /* Estar Beauty HD MID 7316R */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Estar"), + DMI_MATCH(DMI_PRODUCT_NAME, "eSTAR BEAUTY HD Intel Quad core"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.27.0

1 0

[PATCH AUTOSEL 5.4 31/40] ASoC: Intel: Add DMI quirk table to soc_intel_is_byt_cr()
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 8ade6d8b02b1ead741bd4f6c42921035caab6560 ] Some Bay Trail systems: 1. Use a non CR version of the Bay Trail SoC 2. Contain at least 6 interrupt resources so that the platform_get_resource(pdev, IORESOURCE_IRQ, 5) check to workaround non CR systems which list their IPC IRQ at index 0 despite being non CR does not work 3. Despite 1. and 2. still have their IPC IRQ at index 0 rather then 5 Add a DMI quirk table to check for the few known models with this issue, so that the right IPC IRQ index is used on these systems. Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210120214957.140232-5-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/common/soc-intel-quirks.h | 25 +++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/soc/intel/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h index 863a477d34051..645baf0ed3dd1 100644 --- a/sound/soc/intel/common/soc-intel-quirks.h +++ b/sound/soc/intel/common/soc-intel-quirks.h @@ -11,6 +11,7 @@ #if IS_ENABLED(CONFIG_X86) +#include <linux/dmi.h> #include <asm/cpu_device_id.h> #include <asm/intel-family.h> #include <asm/iosf_mbi.h> @@ -40,12 +41,36 @@ SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_L); static inline bool soc_intel_is_byt_cr(struct platform_device *pdev) { + /* + * List of systems which: + * 1. Use a non CR version of the Bay Trail SoC + * 2. Contain at least 6 interrupt resources so that the + * platform_get_resource(pdev, IORESOURCE_IRQ, 5) check below + * succeeds + * 3. Despite 1. and 2. still have their IPC IRQ at index 0 rather then 5 + * + * This needs to be here so that it can be shared between the SST and + * SOF drivers. We rely on the compiler to optimize this out in files + * where soc_intel_is_byt_cr is not used. + */ + static const struct dmi_system_id force_bytcr_table[] = { + { /* Lenovo Yoga Tablet 2 series */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_FAMILY, "YOGATablet2"), + }, + }, + {} + }; struct device *dev = &pdev->dev; int status = 0; if (!soc_intel_is_byt()) return false; + if (dmi_check_system(force_bytcr_table)) + return true; + if (iosf_mbi_available()) { u32 bios_status; -- 2.27.0

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[PATCH AUTOSEL 5.10 56/56] ASoC: Intel: bytcr_rt5640: Add quirk for the Acer One S1002 tablet
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit c58947af08aedbdee0fce5ea6e6bf3e488ae0e2c ] The Acer One S1002 tablet is using an analog mic on IN1 and has its jack-detect connected to JD2_IN4N, instead of using the default IN3 for its internal mic and JD1_IN4P for jack-detect. Note it is also using AIF2 instead of AIF1 which is somewhat unusual, this is correctly advertised in the ACPI CHAN package, so the speakers do work without the quirk. Add a quirk for the mic and jack-detect settings. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210216213555.36555-5-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 626677fa1b5c0..3af4cb87032ce 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -402,6 +402,19 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Acer One 10 S1002 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Acer"), + DMI_MATCH(DMI_PRODUCT_NAME, "One S1002"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Acer"), -- 2.27.0

1 0

[PATCH AUTOSEL 5.10 55/56] ASoC: Intel: bytcr_rt5651: Add quirk for the Jumper EZpad 7 tablet
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit df8359c512fa770ffa6b0b0309807d9b9825a47f ] Add a DMI quirk for the Jumper EZpad 7 tablet, this tablet has a jack-detect switch which reads 1/high when a jack is inserted, rather then using the standard active-low setup which most jack-detect switches use. All other settings are using the defaults. Add a DMI-quirk setting the defaults + the BYT_RT5651_JD_NOT_INV flags for this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210216213555.36555-4-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5651.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 688b5e0a49e32..bf8b87d45cb0a 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -435,6 +435,19 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { BYT_RT5651_SSP0_AIF1 | BYT_RT5651_MONO_SPEAKER), }, + { + /* Jumper EZpad 7 */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Jumper"), + DMI_MATCH(DMI_PRODUCT_NAME, "EZpad"), + /* Jumper12x.WJ2012.bsBKRCP05 with the version dropped */ + DMI_MATCH(DMI_BIOS_VERSION, "Jumper12x.WJ2012.bsBKRCP"), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_JD_NOT_INV), + }, { /* KIANO SlimNote 14.2 */ .callback = byt_rt5651_quirk_cb, -- 2.27.0

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[PATCH AUTOSEL 5.10 54/56] ASoC: Intel: bytcr_rt5640: Add quirk for the Voyo Winpad A15 tablet
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit e1317cc9ca4ac20262895fddb065ffda4fc29cfb ] The Voyo Winpad A15 tablet uses a Bay Trail (non CR) SoC, so it is using SSP2 (AIF1) and it mostly works with the defaults. But instead of using DMIC1 it is using an analog mic on IN1, add a quirk for this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210216213555.36555-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index be874d2a109b9..626677fa1b5c0 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -799,6 +799,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, + { /* Voyo Winpad A15 */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Aptio CRB"), + /* Above strings are too generic, also match on BIOS date */ + DMI_MATCH(DMI_BIOS_DATE, "11/20/2014"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), -- 2.27.0

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[PATCH AUTOSEL 5.10 53/56] ASoC: Intel: bytcr_rt5640: Add quirk for the Estar Beauty HD MID 7316R tablet
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit bdea43fc0436c9e98fdfe151c2ed8a3fc7277404 ] The Estar Beauty HD MID 7316R tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210216213555.36555-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index db3633de9122f..be874d2a109b9 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -515,6 +515,16 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MONO_SPEAKER | BYT_RT5640_MCLK_EN), }, + { /* Estar Beauty HD MID 7316R */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Estar"), + DMI_MATCH(DMI_PRODUCT_NAME, "eSTAR BEAUTY HD Intel Quad core"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.27.0

1 0

[PATCH AUTOSEL 5.10 48/56] ASoC: Intel: sof_sdw: detect DMIC number based on mach params
by Sasha Levin 24 Feb '21

24 Feb '21

From: Rander Wang <rander.wang(a)intel.com> [ Upstream commit f88dcb9b98d3f86ead04d2453475267910448bb8 ] Current driver create DMIC dai based on quirk for each platforms, so we need to add quirk for new platforms. Now driver reports DMIC number to machine driver and machine driver can create DMIC dai based on this information. The old check is reserved for some platforms may be failed to set the DMIC number in BIOS. Reviewed-by: Bard Liao <bard.liao(a)intel.com> Signed-off-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210208233336.59449-6-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 2d2b0519eee25..a30c1d7b3d95f 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -923,7 +923,7 @@ static int sof_card_dai_links_create(struct device *dev, ctx->idisp_codec = true; /* enable dmic01 & dmic16k */ - dmic_num = (sof_sdw_quirk & SOF_SDW_PCH_DMIC) ? 2 : 0; + dmic_num = (sof_sdw_quirk & SOF_SDW_PCH_DMIC || mach_params->dmic_num) ? 2 : 0; comp_num += dmic_num; dev_dbg(dev, "sdw %d, ssp %d, dmic %d, hdmi %d", sdw_be_num, ssp_num, -- 2.27.0

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[PATCH AUTOSEL 5.10 47/56] ASoC: Intel: sof-sdw: indent and add quirks consistently
by Sasha Levin 24 Feb '21

24 Feb '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 8caf37e2be761688c396c609880936a807af490f ] Use the same style for all quirks to avoid misses and errors Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20210208233336.59449-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index a8d43c87cb5a2..2d2b0519eee25 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -113,9 +113,10 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Tiger Lake Client Platform"), }, - .driver_data = (void *)(SOF_RT711_JD_SRC_JD1 | - SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC | - SOF_SSP_PORT(SOF_I2S_SSP2)), + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_RT711_JD_SRC_JD1 | + SOF_SDW_PCH_DMIC | + SOF_SSP_PORT(SOF_I2S_SSP2)), }, { .callback = sof_sdw_quirk_cb, @@ -139,7 +140,8 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Google"), DMI_MATCH(DMI_PRODUCT_NAME, "Volteer"), }, - .driver_data = (void *)(SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC | + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC | SOF_SDW_FOUR_SPK), }, { @@ -148,7 +150,8 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Google"), DMI_MATCH(DMI_PRODUCT_NAME, "Ripto"), }, - .driver_data = (void *)(SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC | + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC | SOF_SDW_FOUR_SPK), }, -- 2.27.0

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[PATCH AUTOSEL 5.10 42/56] ASoC: Intel: Add DMI quirk table to soc_intel_is_byt_cr()
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 8ade6d8b02b1ead741bd4f6c42921035caab6560 ] Some Bay Trail systems: 1. Use a non CR version of the Bay Trail SoC 2. Contain at least 6 interrupt resources so that the platform_get_resource(pdev, IORESOURCE_IRQ, 5) check to workaround non CR systems which list their IPC IRQ at index 0 despite being non CR does not work 3. Despite 1. and 2. still have their IPC IRQ at index 0 rather then 5 Add a DMI quirk table to check for the few known models with this issue, so that the right IPC IRQ index is used on these systems. Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210120214957.140232-5-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/common/soc-intel-quirks.h | 25 +++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/soc/intel/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h index b07df3059926d..a93987ab7f4d7 100644 --- a/sound/soc/intel/common/soc-intel-quirks.h +++ b/sound/soc/intel/common/soc-intel-quirks.h @@ -11,6 +11,7 @@ #if IS_ENABLED(CONFIG_X86) +#include <linux/dmi.h> #include <asm/cpu_device_id.h> #include <asm/intel-family.h> #include <asm/iosf_mbi.h> @@ -38,12 +39,36 @@ SOC_INTEL_IS_CPU(cml, KABYLAKE_L); static inline bool soc_intel_is_byt_cr(struct platform_device *pdev) { + /* + * List of systems which: + * 1. Use a non CR version of the Bay Trail SoC + * 2. Contain at least 6 interrupt resources so that the + * platform_get_resource(pdev, IORESOURCE_IRQ, 5) check below + * succeeds + * 3. Despite 1. and 2. still have their IPC IRQ at index 0 rather then 5 + * + * This needs to be here so that it can be shared between the SST and + * SOF drivers. We rely on the compiler to optimize this out in files + * where soc_intel_is_byt_cr is not used. + */ + static const struct dmi_system_id force_bytcr_table[] = { + { /* Lenovo Yoga Tablet 2 series */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_FAMILY, "YOGATablet2"), + }, + }, + {} + }; struct device *dev = &pdev->dev; int status = 0; if (!soc_intel_is_byt()) return false; + if (dmi_check_system(force_bytcr_table)) + return true; + if (iosf_mbi_available()) { u32 bios_status; -- 2.27.0

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[PATCH AUTOSEL 5.10 27/56] ASoC: Intel: bytcr_rt5640: Add new BYT_RT5640_NO_SPEAKERS quirk-flag
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 1851ccf9e155b2a6f6cca1a7bd49325f5efbd5d2 ] Some devices, like mini PCs/media/top-set boxes do not have any speakers at all, an example of the is the Mele PCG03 Mini PC. Add a new BYT_RT5640_NO_SPEAKERS quirk-flag which when sets does not add speaker routes and modifies the components and the (optional) long_name strings to reflect that there are no speakers. Cc: Rasmus Porsager <rasmus(a)beat.dk> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210109210119.159032-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 26 +++++++++++++++++++------- 1 file changed, 19 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index f790514a147dd..db3633de9122f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -71,6 +71,7 @@ enum { #define BYT_RT5640_SSP0_AIF2 BIT(21) #define BYT_RT5640_MCLK_EN BIT(22) #define BYT_RT5640_MCLK_25MHZ BIT(23) +#define BYT_RT5640_NO_SPEAKERS BIT(24) #define BYTCR_INPUT_DEFAULTS \ (BYT_RT5640_IN3_MAP | \ @@ -132,6 +133,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk JD_NOT_INV enabled\n"); if (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) dev_info(dev, "quirk MONO_SPEAKER enabled\n"); + if (byt_rt5640_quirk & BYT_RT5640_NO_SPEAKERS) + dev_info(dev, "quirk NO_SPEAKERS enabled\n"); if (byt_rt5640_quirk & BYT_RT5640_DIFF_MIC) dev_info(dev, "quirk DIFF_MIC enabled\n"); if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) { @@ -934,7 +937,7 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) ret = snd_soc_dapm_add_routes(&card->dapm, byt_rt5640_mono_spk_map, ARRAY_SIZE(byt_rt5640_mono_spk_map)); - } else { + } else if (!(byt_rt5640_quirk & BYT_RT5640_NO_SPEAKERS)) { ret = snd_soc_dapm_add_routes(&card->dapm, byt_rt5640_stereo_spk_map, ARRAY_SIZE(byt_rt5640_stereo_spk_map)); @@ -1179,6 +1182,7 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) { static const char * const map_name[] = { "dmic1", "dmic2", "in1", "in3" }; + __maybe_unused const char *spk_type; const struct dmi_system_id *dmi_id; struct byt_rt5640_private *priv; struct snd_soc_acpi_mach *mach; @@ -1186,7 +1190,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) struct acpi_device *adev; int ret_val = 0; int dai_index = 0; - int i; + int i, cfg_spk; is_bytcr = false; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -1325,16 +1329,24 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) } } + if (byt_rt5640_quirk & BYT_RT5640_NO_SPEAKERS) { + cfg_spk = 0; + spk_type = "none"; + } else if (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) { + cfg_spk = 1; + spk_type = "mono"; + } else { + cfg_spk = 2; + spk_type = "stereo"; + } + snprintf(byt_rt5640_components, sizeof(byt_rt5640_components), - "cfg-spk:%s cfg-mic:%s", - (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) ? "1" : "2", + "cfg-spk:%d cfg-mic:%s", cfg_spk, map_name[BYT_RT5640_MAP(byt_rt5640_quirk)]); byt_rt5640_card.components = byt_rt5640_components; #if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES) snprintf(byt_rt5640_long_name, sizeof(byt_rt5640_long_name), - "bytcr-rt5640-%s-spk-%s-mic", - (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) ? - "mono" : "stereo", + "bytcr-rt5640-%s-spk-%s-mic", spk_type, map_name[BYT_RT5640_MAP(byt_rt5640_quirk)]); byt_rt5640_card.long_name = byt_rt5640_long_name; #endif -- 2.27.0

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[PATCH AUTOSEL 5.11 67/67] ASoC: Intel: bytcr_rt5640: Add quirk for the Acer One S1002 tablet
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit c58947af08aedbdee0fce5ea6e6bf3e488ae0e2c ] The Acer One S1002 tablet is using an analog mic on IN1 and has its jack-detect connected to JD2_IN4N, instead of using the default IN3 for its internal mic and JD1_IN4P for jack-detect. Note it is also using AIF2 instead of AIF1 which is somewhat unusual, this is correctly advertised in the ACPI CHAN package, so the speakers do work without the quirk. Add a quirk for the mic and jack-detect settings. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210216213555.36555-5-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index ba8ea651a22e6..f00d4e417b6cf 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -402,6 +402,19 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Acer One 10 S1002 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Acer"), + DMI_MATCH(DMI_PRODUCT_NAME, "One S1002"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Acer"), -- 2.27.0

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[PATCH AUTOSEL 5.11 66/67] ASoC: Intel: bytcr_rt5651: Add quirk for the Jumper EZpad 7 tablet
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit df8359c512fa770ffa6b0b0309807d9b9825a47f ] Add a DMI quirk for the Jumper EZpad 7 tablet, this tablet has a jack-detect switch which reads 1/high when a jack is inserted, rather then using the standard active-low setup which most jack-detect switches use. All other settings are using the defaults. Add a DMI-quirk setting the defaults + the BYT_RT5651_JD_NOT_INV flags for this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210216213555.36555-4-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5651.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index f289ec8563a11..148b7b1bd3e8c 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -435,6 +435,19 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { BYT_RT5651_SSP0_AIF1 | BYT_RT5651_MONO_SPEAKER), }, + { + /* Jumper EZpad 7 */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Jumper"), + DMI_MATCH(DMI_PRODUCT_NAME, "EZpad"), + /* Jumper12x.WJ2012.bsBKRCP05 with the version dropped */ + DMI_MATCH(DMI_BIOS_VERSION, "Jumper12x.WJ2012.bsBKRCP"), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_JD_NOT_INV), + }, { /* KIANO SlimNote 14.2 */ .callback = byt_rt5651_quirk_cb, -- 2.27.0

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[PATCH AUTOSEL 5.11 65/67] ASoC: Intel: bytcr_rt5640: Add quirk for the Voyo Winpad A15 tablet
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit e1317cc9ca4ac20262895fddb065ffda4fc29cfb ] The Voyo Winpad A15 tablet uses a Bay Trail (non CR) SoC, so it is using SSP2 (AIF1) and it mostly works with the defaults. But instead of using DMIC1 it is using an analog mic on IN1, add a quirk for this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210216213555.36555-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index ee41f41c8184e..ba8ea651a22e6 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -811,6 +811,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, + { /* Voyo Winpad A15 */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Aptio CRB"), + /* Above strings are too generic, also match on BIOS date */ + DMI_MATCH(DMI_BIOS_DATE, "11/20/2014"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), -- 2.27.0

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[PATCH AUTOSEL 5.11 64/67] ASoC: Intel: bytcr_rt5640: Add quirk for the Estar Beauty HD MID 7316R tablet
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit bdea43fc0436c9e98fdfe151c2ed8a3fc7277404 ] The Estar Beauty HD MID 7316R tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210216213555.36555-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index dce2df30d4c5b..ee41f41c8184e 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -527,6 +527,16 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MONO_SPEAKER | BYT_RT5640_MCLK_EN), }, + { /* Estar Beauty HD MID 7316R */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Estar"), + DMI_MATCH(DMI_PRODUCT_NAME, "eSTAR BEAUTY HD Intel Quad core"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.27.0

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[PATCH AUTOSEL 5.11 59/67] ASoC: Intel: sof_sdw: detect DMIC number based on mach params
by Sasha Levin 24 Feb '21

24 Feb '21

From: Rander Wang <rander.wang(a)intel.com> [ Upstream commit f88dcb9b98d3f86ead04d2453475267910448bb8 ] Current driver create DMIC dai based on quirk for each platforms, so we need to add quirk for new platforms. Now driver reports DMIC number to machine driver and machine driver can create DMIC dai based on this information. The old check is reserved for some platforms may be failed to set the DMIC number in BIOS. Reviewed-by: Bard Liao <bard.liao(a)intel.com> Signed-off-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210208233336.59449-6-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 6e4b2fdb0dcf6..9983bf6356b1f 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -933,7 +933,7 @@ static int sof_card_dai_links_create(struct device *dev, ctx->idisp_codec = true; /* enable dmic01 & dmic16k */ - dmic_num = (sof_sdw_quirk & SOF_SDW_PCH_DMIC) ? 2 : 0; + dmic_num = (sof_sdw_quirk & SOF_SDW_PCH_DMIC || mach_params->dmic_num) ? 2 : 0; comp_num += dmic_num; dev_dbg(dev, "sdw %d, ssp %d, dmic %d, hdmi %d", sdw_be_num, ssp_num, -- 2.27.0

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[PATCH AUTOSEL 5.11 58/67] ASoC: Intel: sof-sdw: indent and add quirks consistently
by Sasha Levin 24 Feb '21

24 Feb '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 8caf37e2be761688c396c609880936a807af490f ] Use the same style for all quirks to avoid misses and errors Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20210208233336.59449-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 6d0d6ef711e0f..6e4b2fdb0dcf6 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -123,9 +123,10 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Tiger Lake Client Platform"), }, - .driver_data = (void *)(SOF_RT711_JD_SRC_JD1 | - SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC | - SOF_SSP_PORT(SOF_I2S_SSP2)), + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_RT711_JD_SRC_JD1 | + SOF_SDW_PCH_DMIC | + SOF_SSP_PORT(SOF_I2S_SSP2)), }, { .callback = sof_sdw_quirk_cb, @@ -149,7 +150,8 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Google"), DMI_MATCH(DMI_PRODUCT_NAME, "Volteer"), }, - .driver_data = (void *)(SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC | + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC | SOF_SDW_FOUR_SPK), }, { @@ -158,7 +160,8 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Google"), DMI_MATCH(DMI_PRODUCT_NAME, "Ripto"), }, - .driver_data = (void *)(SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC | + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC | SOF_SDW_FOUR_SPK), }, -- 2.27.0

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[PATCH AUTOSEL 5.11 52/67] ASoC: Intel: Add DMI quirk table to soc_intel_is_byt_cr()
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 8ade6d8b02b1ead741bd4f6c42921035caab6560 ] Some Bay Trail systems: 1. Use a non CR version of the Bay Trail SoC 2. Contain at least 6 interrupt resources so that the platform_get_resource(pdev, IORESOURCE_IRQ, 5) check to workaround non CR systems which list their IPC IRQ at index 0 despite being non CR does not work 3. Despite 1. and 2. still have their IPC IRQ at index 0 rather then 5 Add a DMI quirk table to check for the few known models with this issue, so that the right IPC IRQ index is used on these systems. Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210120214957.140232-5-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/common/soc-intel-quirks.h | 25 +++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/soc/intel/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h index b07df3059926d..a93987ab7f4d7 100644 --- a/sound/soc/intel/common/soc-intel-quirks.h +++ b/sound/soc/intel/common/soc-intel-quirks.h @@ -11,6 +11,7 @@ #if IS_ENABLED(CONFIG_X86) +#include <linux/dmi.h> #include <asm/cpu_device_id.h> #include <asm/intel-family.h> #include <asm/iosf_mbi.h> @@ -38,12 +39,36 @@ SOC_INTEL_IS_CPU(cml, KABYLAKE_L); static inline bool soc_intel_is_byt_cr(struct platform_device *pdev) { + /* + * List of systems which: + * 1. Use a non CR version of the Bay Trail SoC + * 2. Contain at least 6 interrupt resources so that the + * platform_get_resource(pdev, IORESOURCE_IRQ, 5) check below + * succeeds + * 3. Despite 1. and 2. still have their IPC IRQ at index 0 rather then 5 + * + * This needs to be here so that it can be shared between the SST and + * SOF drivers. We rely on the compiler to optimize this out in files + * where soc_intel_is_byt_cr is not used. + */ + static const struct dmi_system_id force_bytcr_table[] = { + { /* Lenovo Yoga Tablet 2 series */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_FAMILY, "YOGATablet2"), + }, + }, + {} + }; struct device *dev = &pdev->dev; int status = 0; if (!soc_intel_is_byt()) return false; + if (dmi_check_system(force_bytcr_table)) + return true; + if (iosf_mbi_available()) { u32 bios_status; -- 2.27.0

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[PATCH AUTOSEL 5.11 48/67] ALSA: usb-audio: Add DJM-450 to the quirks table
by Sasha Levin 24 Feb '21

24 Feb '21

From: Olivia Mackintosh <livvy(a)base.nu> [ Upstream commit 9119e5661eab2c56a96b936cde49c6740dc49ff9 ] As with most Pioneer devices, the device descriptor is vendor specific and as such, the number of channels, the PCM format, endpoints and sample rate need to be specified. This device has 8 inputs and 8 outputs and a sample rate of 48000 only. The PCM format is S24_3LE like other devices. There seems to be an appetite for reducing duplication amongs these Pioneer patches but again, I feel this is a step to be taken after support has been added as it's not completely clear where the commonalities are. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> Link: https://lore.kernel.org/r/20210202134225.3217-3-livvy@base.nu Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks-table.h | 57 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 93d55cd1a5a4c..1165a5ac60f22 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3817,6 +3817,63 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-450 + * PCM is 8 channels out @ 48 fixed (endpoint 0x01) + * and 8 channels in @ 48 fixed (endpoint 0x82). + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0013), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // outputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // inputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC #undef USB_AUDIO_DEVICE -- 2.27.0

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[PATCH AUTOSEL 5.11 47/67] ALSA: usb-audio: Add DJM450 to Pioneer format quirk
by Sasha Levin 24 Feb '21

24 Feb '21

From: Olivia Mackintosh <livvy(a)base.nu> [ Upstream commit 3b85f5fc75d564a9eb4171dcb6b8687b080cd4d5 ] Like the DJM-750, ensure that the format control message is passed to the device when opening a stream. It seems as though fmt->sync_ep is not always set when this function is called hence the passing of the value at the call site. If this can be fixed, fmt->sync_up should be used as the wvalue. There doesn't seem to be a "cpu_to_le24" type function defined hence for the open code but I did see a similar thing done in Bluez lib. Perhaps we can get these definitions defined in byteorder.h. See hci_cpu_to_le24 in include/net/bluetooth/hci.h:2543 for similar usage. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> Link: https://lore.kernel.org/r/20210202134225.3217-2-livvy@base.nu Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e196e364cef19..9ba4682ebc482 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1470,6 +1470,23 @@ static void set_format_emu_quirk(struct snd_usb_substream *subs, subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0; } +static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs, + u16 windex) +{ + unsigned int cur_rate = subs->data_endpoint->cur_rate; + u8 sr[3]; + // Convert to little endian + sr[0] = cur_rate & 0xff; + sr[1] = (cur_rate >> 8) & 0xff; + sr[2] = (cur_rate >> 16) & 0xff; + usb_set_interface(subs->dev, 0, 1); + // we should derive windex from fmt-sync_ep but it's not set + snd_usb_ctl_msg(subs->stream->chip->dev, + usb_rcvctrlpipe(subs->stream->chip->dev, 0), + 0x01, 0x22, 0x0100, windex, &sr, 0x0003); + return 0; +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1483,6 +1500,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; + case USB_ID(0x2b73, 0x0013): /* Pioneer DJM-450 */ + pioneer_djm_set_format_quirk(subs, 0x0082); + break; } } -- 2.27.0

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[PATCH AUTOSEL 5.11 34/67] ALSA: usb-audio: Add support for Pioneer DJM-750
by Sasha Levin 24 Feb '21

24 Feb '21

From: Olivia Mackintosh <livvy(a)base.nu> [ Upstream commit b952ac76a20bc0b23cd7e22de19fb407713238a3 ] This adds the Pioneer DJ DJM-750 to the quirks table and ensures skip_pioneer_sync_ep() is (also) called: this device uses the vendor ID of 0x08e4 (I'm not sure why they use multiple vendor IDs but many just like to be awkward it seems). Playback on all 8 channels works. I'll likely keep this working in the future and submit futher patches and improvements as necessary. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> Link: https://lore.kernel.org/r/20210118130621.77miiie47wp7mump@base.nu Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/implicit.c | 3 +- sound/usb/quirks-table.h | 60 ++++++++++++++++++++++++++++++++++++++++ 2 files changed, 62 insertions(+), 1 deletion(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 521cc846d9d9f..e7216d0b860d8 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -302,7 +302,8 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, /* Pioneer devices with vendor spec class */ if (attr == USB_ENDPOINT_SYNC_ASYNC && alts->desc.bInterfaceClass == USB_CLASS_VENDOR_SPEC && - USB_ID_VENDOR(chip->usb_id) == 0x2b73 /* Pioneer */) { + (USB_ID_VENDOR(chip->usb_id) == 0x2b73 || /* Pioneer */ + USB_ID_VENDOR(chip->usb_id) == 0x08e4 /* Pioneer */)) { if (skip_pioneer_sync_ep(chip, fmt, alts)) return 1; } diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c8a4bdf18207c..93d55cd1a5a4c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3757,6 +3757,66 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-750 + * 8 channels playback & 8 channels capture @ 44.1/48/96kHz S24LE + */ + USB_DEVICE_VENDOR_SPEC(0x08e4, 0x017f), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x05, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { 44100, 48000, 96000 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x86, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { 44100, 48000, 96000 } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC #undef USB_AUDIO_DEVICE -- 2.27.0

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[PATCH AUTOSEL 5.11 33/67] ASoC: Intel: bytcr_rt5640: Add new BYT_RT5640_NO_SPEAKERS quirk-flag
by Sasha Levin 24 Feb '21

24 Feb '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 1851ccf9e155b2a6f6cca1a7bd49325f5efbd5d2 ] Some devices, like mini PCs/media/top-set boxes do not have any speakers at all, an example of the is the Mele PCG03 Mini PC. Add a new BYT_RT5640_NO_SPEAKERS quirk-flag which when sets does not add speaker routes and modifies the components and the (optional) long_name strings to reflect that there are no speakers. Cc: Rasmus Porsager <rasmus(a)beat.dk> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210109210119.159032-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 26 +++++++++++++++++++------- 1 file changed, 19 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 5520d7c800196..dce2df30d4c5b 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -71,6 +71,7 @@ enum { #define BYT_RT5640_SSP0_AIF2 BIT(21) #define BYT_RT5640_MCLK_EN BIT(22) #define BYT_RT5640_MCLK_25MHZ BIT(23) +#define BYT_RT5640_NO_SPEAKERS BIT(24) #define BYTCR_INPUT_DEFAULTS \ (BYT_RT5640_IN3_MAP | \ @@ -132,6 +133,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk JD_NOT_INV enabled\n"); if (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) dev_info(dev, "quirk MONO_SPEAKER enabled\n"); + if (byt_rt5640_quirk & BYT_RT5640_NO_SPEAKERS) + dev_info(dev, "quirk NO_SPEAKERS enabled\n"); if (byt_rt5640_quirk & BYT_RT5640_DIFF_MIC) dev_info(dev, "quirk DIFF_MIC enabled\n"); if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) { @@ -946,7 +949,7 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) ret = snd_soc_dapm_add_routes(&card->dapm, byt_rt5640_mono_spk_map, ARRAY_SIZE(byt_rt5640_mono_spk_map)); - } else { + } else if (!(byt_rt5640_quirk & BYT_RT5640_NO_SPEAKERS)) { ret = snd_soc_dapm_add_routes(&card->dapm, byt_rt5640_stereo_spk_map, ARRAY_SIZE(byt_rt5640_stereo_spk_map)); @@ -1188,6 +1191,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; static const char * const map_name[] = { "dmic1", "dmic2", "in1", "in3" }; + __maybe_unused const char *spk_type; const struct dmi_system_id *dmi_id; struct byt_rt5640_private *priv; struct snd_soc_acpi_mach *mach; @@ -1196,7 +1200,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) bool sof_parent; int ret_val = 0; int dai_index = 0; - int i; + int i, cfg_spk; is_bytcr = false; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -1335,16 +1339,24 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) } } + if (byt_rt5640_quirk & BYT_RT5640_NO_SPEAKERS) { + cfg_spk = 0; + spk_type = "none"; + } else if (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) { + cfg_spk = 1; + spk_type = "mono"; + } else { + cfg_spk = 2; + spk_type = "stereo"; + } + snprintf(byt_rt5640_components, sizeof(byt_rt5640_components), - "cfg-spk:%s cfg-mic:%s", - (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) ? "1" : "2", + "cfg-spk:%d cfg-mic:%s", cfg_spk, map_name[BYT_RT5640_MAP(byt_rt5640_quirk)]); byt_rt5640_card.components = byt_rt5640_components; #if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES) snprintf(byt_rt5640_long_name, sizeof(byt_rt5640_long_name), - "bytcr-rt5640-%s-spk-%s-mic", - (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) ? - "mono" : "stereo", + "bytcr-rt5640-%s-spk-%s-mic", spk_type, map_name[BYT_RT5640_MAP(byt_rt5640_quirk)]); byt_rt5640_card.long_name = byt_rt5640_long_name; #endif -- 2.27.0

1 0

[PATCH alsa-lib 1/3] mixer: Unify simple_none: base_len() exception handling
by Hans de Goede 24 Feb '21

24 Feb '21

Unify simple_none: base_len() exception handling: 1. In the "Input Source" and "3D Control" cases the base-name is the same as the full-name and base_len() simply returns strlen(name). Instead of returning 0 when the type is unknown, set the type to CTL_SINGLE and return strlen(name). This allows removing the special case for base_len() returning 0 in simple_event_add(). 2. Move the special handling for "Capture Volume" and "Capture Switch" from simple_event_add() to base_len(), so that we handle all exceptions inside base_len(). Instead of handling some special cases in base_len() and other special cases in simple_event_add(). Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- src/mixer/simple_none.c | 43 ++++++++++++++++++++++++----------------- 1 file changed, 25 insertions(+), 18 deletions(-) diff --git a/src/mixer/simple_none.c b/src/mixer/simple_none.c index e9dc1730..7da7de1e 100644 --- a/src/mixer/simple_none.c +++ b/src/mixer/simple_none.c @@ -907,11 +907,22 @@ static const struct suf { }; #endif -/* Return base length or 0 on failure */ +/* Return base length */ static int base_len(const char *name, selem_ctl_type_t *type) { const struct suf *p; size_t nlen = strlen(name); + + /* exception: "Capture Volume" and "Capture Switch" */ + if (!strcmp(name, "Capture Volume")) { + *type = CTL_CAPTURE_VOLUME; + return strlen("Capture"); + } + if (!strcmp(name, "Capture Switch")) { + *type = CTL_CAPTURE_SWITCH; + return strlen("Capture"); + } + p = suffixes; while (p->suffix) { size_t slen = strlen(p->suffix); @@ -944,7 +955,9 @@ static int base_len(const char *name, selem_ctl_type_t *type) return strlen(name); } } - return 0; + + *type = CTL_SINGLE; + return strlen(name); } @@ -1605,8 +1618,10 @@ static int simple_add1(snd_mixer_class_t *class, const char *name, static int simple_event_add(snd_mixer_class_t *class, snd_hctl_elem_t *helem) { const char *name = snd_hctl_elem_get_name(helem); + selem_ctl_type_t type; + char ename[128]; size_t len; - selem_ctl_type_t type = CTL_SINGLE; /* to shut up warning */ + if (snd_hctl_elem_get_interface(helem) != SND_CTL_ELEM_IFACE_MIXER) return 0; if (strcmp(name, "Capture Source") == 0) { @@ -1633,22 +1648,14 @@ static int simple_event_add(snd_mixer_class_t *class, snd_hctl_elem_t *helem) } return 0; } + len = base_len(name, &type); - if (len == 0) { - return simple_add1(class, name, helem, CTL_SINGLE, 0); - } else { - char ename[128]; - if (len >= sizeof(ename)) - len = sizeof(ename) - 1; - memcpy(ename, name, len); - ename[len] = 0; - /* exception: Capture Volume and Capture Switch */ - if (type == CTL_GLOBAL_VOLUME && !strcmp(ename, "Capture")) - type = CTL_CAPTURE_VOLUME; - else if (type == CTL_GLOBAL_SWITCH && !strcmp(ename, "Capture")) - type = CTL_CAPTURE_SWITCH; - return simple_add1(class, ename, helem, type, 0); - } + if (len >= sizeof(ename)) + len = sizeof(ename) - 1; + memcpy(ename, name, len); + ename[len] = 0; + + return simple_add1(class, ename, helem, type, 0); } static int simple_event_remove(snd_hctl_elem_t *helem, -- 2.30.1

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[PATCH v2] leds: trigger: audio: Add an activate callback to ensure the initial brightness is set
by Hans de Goede 24 Feb '21

24 Feb '21

Some 2-in-1s with a detachable (USB) keyboard(dock) have mute-LEDs in the speaker- and/or mic-mute keys on the keyboard. Examples of this are the Lenovo Thinkpad10 tablet (with its USB kbd-dock) and the HP x2 10 series. The detachable nature of these keyboards means that the keyboard and thus the mute LEDs may show up after the user (or userspace restoring old mixer settings) has muted the speaker and/or mic. Current LED-class devices with a default_trigger of "audio-mute" or "audio-micmute" initialize the brightness member of led_classdev with ledtrig_audio_get() before registering the LED. This makes the software state after attaching the keyboard match the actual audio mute state, e.g. cat /sys/class/leds/foo/brightness will show the right value. But before this commit nothing was actually calling the led_classdev's brightness_set[_blocking] callback so the value returned by ledtrig_audio_get() was never actually being sent to the hw, leading to the mute LEDs staying in their default power-on state, after attaching the keyboard, even if ledtrig_audio_get() returned a different state. This could be fixed by having the individual LED drivers call brightness_set[_blocking] themselves after registering the LED, but this really is something which should be done by a led-trigger activate callback. Add an activate callback for this, fixing the issue of the mute LEDs being out of sync after (re)attaching the keyboard. Cc: Takashi Iwai <tiwai(a)suse.de> Fixes: faa2541f5b1a ("leds: trigger: Introduce audio mute LED trigger") Reviewed-by: Marek Behún <kabel(a)kernel.org> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Changes in v2: - Fix a couple of grammar errors in the commit-message - Add Marek's Reviewed-by (thank you) --- drivers/leds/trigger/ledtrig-audio.c | 37 ++++++++++++++++++++++------ 1 file changed, 29 insertions(+), 8 deletions(-) diff --git a/drivers/leds/trigger/ledtrig-audio.c b/drivers/leds/trigger/ledtrig-audio.c index f76621e88482..c6b437e6369b 100644 --- a/drivers/leds/trigger/ledtrig-audio.c +++ b/drivers/leds/trigger/ledtrig-audio.c @@ -6,10 +6,33 @@ #include <linux/kernel.h> #include <linux/leds.h> #include <linux/module.h> +#include "../leds.h" -static struct led_trigger *ledtrig_audio[NUM_AUDIO_LEDS]; static enum led_brightness audio_state[NUM_AUDIO_LEDS]; +static int ledtrig_audio_mute_activate(struct led_classdev *led_cdev) +{ + led_set_brightness_nosleep(led_cdev, audio_state[LED_AUDIO_MUTE]); + return 0; +} + +static int ledtrig_audio_micmute_activate(struct led_classdev *led_cdev) +{ + led_set_brightness_nosleep(led_cdev, audio_state[LED_AUDIO_MICMUTE]); + return 0; +} + +static struct led_trigger ledtrig_audio[NUM_AUDIO_LEDS] = { + [LED_AUDIO_MUTE] = { + .name = "audio-mute", + .activate = ledtrig_audio_mute_activate, + }, + [LED_AUDIO_MICMUTE] = { + .name = "audio-micmute", + .activate = ledtrig_audio_micmute_activate, + }, +}; + enum led_brightness ledtrig_audio_get(enum led_audio type) { return audio_state[type]; @@ -19,24 +42,22 @@ EXPORT_SYMBOL_GPL(ledtrig_audio_get); void ledtrig_audio_set(enum led_audio type, enum led_brightness state) { audio_state[type] = state; - led_trigger_event(ledtrig_audio[type], state); + led_trigger_event(&ledtrig_audio[type], state); } EXPORT_SYMBOL_GPL(ledtrig_audio_set); static int __init ledtrig_audio_init(void) { - led_trigger_register_simple("audio-mute", - &ledtrig_audio[LED_AUDIO_MUTE]); - led_trigger_register_simple("audio-micmute", - &ledtrig_audio[LED_AUDIO_MICMUTE]); + led_trigger_register(&ledtrig_audio[LED_AUDIO_MUTE]); + led_trigger_register(&ledtrig_audio[LED_AUDIO_MICMUTE]); return 0; } module_init(ledtrig_audio_init); static void __exit ledtrig_audio_exit(void) { - led_trigger_unregister_simple(ledtrig_audio[LED_AUDIO_MUTE]); - led_trigger_unregister_simple(ledtrig_audio[LED_AUDIO_MICMUTE]); + led_trigger_unregister(&ledtrig_audio[LED_AUDIO_MUTE]); + led_trigger_unregister(&ledtrig_audio[LED_AUDIO_MICMUTE]); } module_exit(ledtrig_audio_exit); -- 2.30.1

3 3

[PATCH -next] sound: n64: Fix return value check in n64audio_probe()
by Wei Yongjun 24 Feb '21

24 Feb '21

In case of error, the function devm_platform_ioremap_resource() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Fixes: 1448f8acf4cc ("sound: Add n64 driver") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/mips/snd-n64.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/mips/snd-n64.c b/sound/mips/snd-n64.c index ca6b4b99da98..e35e93157755 100644 --- a/sound/mips/snd-n64.c +++ b/sound/mips/snd-n64.c @@ -312,14 +312,14 @@ static int __init n64audio_probe(struct platform_device *pdev) } priv->mi_reg_base = devm_platform_ioremap_resource(pdev, 0); - if (!priv->mi_reg_base) { - err = -EINVAL; + if (IS_ERR(priv->mi_reg_base)) { + err = PTR_ERR(priv->mi_reg_base); goto fail_dma_alloc; } priv->ai_reg_base = devm_platform_ioremap_resource(pdev, 1); - if (!priv->ai_reg_base) { - err = -EINVAL; + if (IS_ERR(priv->ai_reg_base)) { + err = PTR_ERR(priv->ai_reg_base); goto fail_dma_alloc; }

3 2

Re: [PATCH] ASoC: rt5682: remove jack detect delay
by Takashi Iwai 23 Feb '21

23 Feb '21

On Thu, 18 Feb 2021 09:38:53 +0100, Shuming [范書銘] <shumingf(a)realtek.com> wrote: > > > There is a 250ms delay on the jack detect interrupt currently, this delay is > > observable to users who are using inline controls. It can also mask multiple > > presses which is a negative experience. > > > > Cc: Bard liao <yung-chuan.liao(a)linux.intel.com> > > Cc: Shuming Fan <shumingf(a)realtek.com> > > > > Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> > > --- > > sound/soc/codecs/rt5682-i2c.c | 2 +- > > 1 file changed, 1 insertion(+), 1 deletion(-) > > > > diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c > > index 93c1603b42f1..b15c3e7d1f59 100644 > > --- a/sound/soc/codecs/rt5682-i2c.c > > +++ b/sound/soc/codecs/rt5682-i2c.c > > @@ -78,7 +78,7 @@ static irqreturn_t rt5682_irq(int irq, void *data) > > struct rt5682_priv *rt5682 = data; > > > > mod_delayed_work(system_power_efficient_wq, > > - &rt5682->jack_detect_work, msecs_to_jiffies(250)); > > + &rt5682->jack_detect_work, 0); > > How about using the device property to adjust the delay time? > I think it should keep the workqueue to do the jack/button detection because the jack type detection will take some times to do. One might check twice (or more) if it's not certain, too. That is, check the jack immediately, and if the jack state really changed, report it so. OTOH, if the jack state doesn't change from the old state, it can retry after some delay. thanks, Takashi

3 7

[PATCH v2 0/6] ASoC: samsung: remove cppcheck warnings
by Pierre-Louis Bossart 23 Feb '21

23 Feb '21

No functional changes except for patch 3 and 4 where missing error checks were added for consistency. v2: added Krzysztof Kozlowski's tags added fix for first patch already merged as suggested by Krzysztof Kozlowski moved variable to lower scope in patch6 Pierre-Louis Bossart (6): ASoC: samsung: tm2_wm5510: fix check of of_parse return value ASoC: samsung: i2s: remove unassigned variable ASoC: samsung: s3c24xx_simtec: add missing error check ASoC: samsung: smdk_wm8994: add missing return ASoC: samsung: snow: remove useless test ASoC: samsung: tm2_wm5510: remove shadowed variable sound/soc/samsung/i2s.c | 3 +-- sound/soc/samsung/s3c24xx_simtec.c | 5 +++++ sound/soc/samsung/smdk_wm8994.c | 1 + sound/soc/samsung/snow.c | 5 +---- sound/soc/samsung/tm2_wm5110.c | 5 +++-- 5 files changed, 11 insertions(+), 8 deletions(-) -- 2.25.1

4 17

[PATCH][next] ASoC: codecs: lpass-rx-macro: Fix uninitialized variable ec_tx
by Colin King 23 Feb '21

23 Feb '21

From: Colin Ian King <colin.king(a)canonical.com> There is potential read of the uninitialized variable ec_tx if the call to snd_soc_component_read fails or returns an unrecognized w->name. To avoid this corner case, initialize ec_tx to -1 so that it is caught later when ec_tx is bounds checked. Addresses-Coverity: ("Uninitialized scalar variable") Fixes: 4f692926f562 ("ASoC: codecs: lpass-rx-macro: add dapm widgets and route") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/lpass-rx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index c9c21d22c2c4..8c04b3b2c907 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -2895,7 +2895,7 @@ static int rx_macro_enable_echo(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); u16 val, ec_hq_reg; - int ec_tx; + int ec_tx = -1; val = snd_soc_component_read(component, CDC_RX_INP_MUX_RX_MIX_CFG4); -- 2.30.0

2 1

[PATCH] ASoC: fsl_ssi: Fix TDM slot setup for I2S mode
by Alexander Shiyan 23 Feb '21

23 Feb '21

When using the driver in I2S TDM mode, the _fsl_ssi_set_dai_fmt() function rewrites the number of slots previously set by the fsl_ssi_set_dai_tdm_slot() function to 2 by default. To fix this, let's use the saved slot count value or, if TDM is not used and the slot count is not set, proceed as before. Fixes: 4f14f5c11db1 ("ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode") Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/fsl_ssi.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 404be27c15fe..1d774c876c52 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -878,6 +878,7 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream, static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt) { u32 strcr = 0, scr = 0, stcr, srcr, mask; + unsigned int slots; ssi->dai_fmt = fmt; @@ -909,10 +910,11 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt) return -EINVAL; } + slots = ssi->slots ? : 2; regmap_update_bits(ssi->regs, REG_SSI_STCCR, - SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(2)); + SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(slots)); regmap_update_bits(ssi->regs, REG_SSI_SRCCR, - SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(2)); + SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(slots)); /* Data on rising edge of bclk, frame low, 1clk before data */ strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP | SSI_STCR_TEFS; -- 2.26.2

3 2

Add audio LED support - control API and PCM softvol plugin
by GitHub pull_request - opened 23 Feb '21

23 Feb '21

alsa-project/alsa-lib pull request #121 was opened from perexg: As proposed for the driver, this set of patches can resolve the LED handling issue for the hardware without mixer switches or volume controls (like AMD ACP bridge). Request URL : https://github.com/alsa-project/alsa-lib/pull/121 Patch URL : https://github.com/alsa-project/alsa-lib/pull/121.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 00/20] Manual replacement of all strlcpy in favor of strscpy
by Romain Perier 23 Feb '21

23 Feb '21

strlcpy() copy a C-String into a sized buffer, the result is always a valid NULL-terminated that fits in the buffer, howerver it has severals issues. It reads the source buffer first, which is dangerous if it is non NULL-terminated or if the corresponding buffer is unbounded. Its safe replacement is strscpy(), as suggested in the deprecated interface [1]. We plan to make this contribution in two steps: - Firsly all cases of strlcpy's return value are manually replaced by the corresponding calls of strscpy() with the new handling of the return value (as the return code is different in case of error). - Then all other cases are automatically replaced by using coccinelle. This series covers manual replacements. [1] https://www.kernel.org/doc/html/latest/process/deprecated.html#strlcpy Romain Perier (20): cgroup: Manual replacement of the deprecated strlcpy() with return values crypto: Manual replacement of the deprecated strlcpy() with return values devlink: Manual replacement of the deprecated strlcpy() with return values dma-buf: Manual replacement of the deprecated strlcpy() with return values kobject: Manual replacement of the deprecated strlcpy() with return values ima: Manual replacement of the deprecated strlcpy() with return values SUNRPC: Manual replacement of the deprecated strlcpy() with return values kernfs: Manual replacement of the deprecated strlcpy() with return values m68k/atari: Manual replacement of the deprecated strlcpy() with return values module: Manual replacement of the deprecated strlcpy() with return values hwmon: Manual replacement of the deprecated strlcpy() with return values s390/hmcdrv: Manual replacement of the deprecated strlcpy() with return values scsi: zfcp: Manual replacement of the deprecated strlcpy() with return values target: Manual replacement of the deprecated strlcpy() with return values ALSA: usb-audio: Manual replacement of the deprecated strlcpy() with return values tracing/probe: Manual replacement of the deprecated strlcpy() with return values vt: Manual replacement of the deprecated strlcpy() with return values usb: gadget: f_midi: Manual replacement of the deprecated strlcpy() with return values usbip: usbip_host: Manual replacement of the deprecated strlcpy() with return values s390/watchdog: Manual replacement of the deprecated strlcpy() with return values arch/m68k/emu/natfeat.c | 6 +-- crypto/lrw.c | 6 +-- crypto/xts.c | 6 +-- drivers/dma-buf/dma-buf.c | 4 +- drivers/hwmon/pmbus/max20730.c | 66 +++++++++++++------------ drivers/s390/char/diag_ftp.c | 4 +- drivers/s390/char/sclp_ftp.c | 6 +-- drivers/s390/scsi/zfcp_fc.c | 8 +-- drivers/target/target_core_configfs.c | 33 ++++--------- drivers/tty/vt/keyboard.c | 5 +- drivers/usb/gadget/function/f_midi.c | 4 +- drivers/usb/gadget/function/f_printer.c | 8 +-- drivers/usb/usbip/stub_main.c | 6 +-- drivers/watchdog/diag288_wdt.c | 12 +++-- fs/kernfs/dir.c | 27 +++++----- kernel/cgroup/cgroup.c | 2 +- kernel/module.c | 4 +- kernel/trace/trace_uprobe.c | 11 ++--- lib/kobject_uevent.c | 6 +-- net/core/devlink.c | 6 +-- net/sunrpc/clnt.c | 6 ++- security/integrity/ima/ima_policy.c | 8 ++- sound/usb/card.c | 4 +- 23 files changed, 129 insertions(+), 119 deletions(-) -- 2.20.1

3 4

[PATCH v2 0/3] HID: elan: Mute LED handling fixes and improvements
by Hans de Goede 23 Feb '21

23 Feb '21

Hi All, Here is v2 of my series with mute LED handling fixes and improvements for the hid-elan driver. New in v2: - Add Marek Behún's Reviewed-by to all patches,thank you for all the reviews Marek. - Improve error handling in patch 1/3 to return -EIO when ret != 3 and >= 0 Regards, Hans Hans de Goede (3): HID: elan: Silence mute LED errors being logged when the device is unplugged HID: elan: Set default_trigger for the mute LED HID: elan: Remove elan_mute_led_get_brigtness() drivers/hid/hid-elan.c | 17 +++++------------ 1 file changed, 5 insertions(+), 12 deletions(-) -- 2.30.1

2 5

[PATCH] sound: core: fixed coding style errors
by Rajesh Kumbhakar 23 Feb '21

23 Feb '21

fixing ERROR: "foo * bar" should be "foo *bar" fixing WARNING: Missing a blank line after declarations Signed-off-by: Rajesh Kumbhakar <sssraj.sssraj(a)gmail.com> --- sound/core/hwdep_compat.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/core/hwdep_compat.c b/sound/core/hwdep_compat.c index a0b76706c..d8624a14a 100644 --- a/sound/core/hwdep_compat.c +++ b/sound/core/hwdep_compat.c @@ -36,11 +36,13 @@ enum { SNDRV_HWDEP_IOCTL_DSP_LOAD32 = _IOW('H', 0x03, struct snd_hwdep_dsp_image32) }; -static long snd_hwdep_ioctl_compat(struct file * file, unsigned int cmd, +static long snd_hwdep_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg) { struct snd_hwdep *hw = file->private_data; + void __user *argp = compat_ptr(arg); + switch (cmd) { case SNDRV_HWDEP_IOCTL_PVERSION: case SNDRV_HWDEP_IOCTL_INFO: -- 2.25.1

2 2

yoga 9i bottom speakers do not work
by GitHub issues - edited 23 Feb '21

23 Feb '21

alsa-project/alsa-lib issue #120 was edited from zany130: this might be the same issue as the c940 and legion 7i not sure, but the soundbar works but the bottom speakers don't https://linux-hardware.org/?probe=1f0bdc3a66 I was trying to get the ALSA Information Script running and the script uploaded but it wasn't connecting so here is the file [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/6026794/alsa-… Issue URL : https://github.com/alsa-project/alsa-lib/issues/120 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

yoga 9i bottom speakers do not work
by GitHub issues - opened 23 Feb '21

23 Feb '21

alsa-project/alsa-lib issue #120 was opened from zany130: this might be the same issue as the c940 and legion 7i not sure, but the soundbar works but the bottom speakers don't https://linux-hardware.org/?probe=1f0bdc3a66 Issue URL : https://github.com/alsa-project/alsa-lib/issues/120 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] sound: soc: codecs: fix kconfig dependency on GPIOLIB
by Julian Braha 22 Feb '21

22 Feb '21

When SND_SOC_DMIC is enabled and GPIOLIB is disabled, Kbuild gives the following warning: WARNING: unmet direct dependencies detected for SND_SOC_DMIC Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_AMD_RENOIR_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AMD_RENOIR [=y] This is because SND_SOC_AMD_RENOIR_MACH selects SND_SOC_DMIC, without depending on or selecting GPIOLIB, despite that config option depending on GPIOLIB. Signed-off-by: Julian Braha <julianbraha(a)gmail.com> --- sound/soc/amd/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index a6ce000fac3f..5480dff7a1ad 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -46,6 +46,7 @@ config SND_SOC_AMD_RENOIR config SND_SOC_AMD_RENOIR_MACH tristate "AMD Renoir support for DMIC" + select GPIOLIB select SND_SOC_DMIC depends on SND_SOC_AMD_RENOIR help -- 2.27.0

1 0

[PATCH 1/2] ASoC: rt1015: fix i2c communication error
by jack.yu＠realtek.com 22 Feb '21

22 Feb '21

From: Jack Yu <jack.yu(a)realtek.com> Remove 0x100 cache re-sync to solve i2c communication error. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 37b5795b00d1..90767490af82 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -209,6 +209,7 @@ static bool rt1015_volatile_register(struct device *dev, unsigned int reg) case RT1015_VENDOR_ID: case RT1015_DEVICE_ID: case RT1015_PRO_ALT: + case RT1015_MAN_I2C: case RT1015_DAC3: case RT1015_VBAT_TEST_OUT1: case RT1015_VBAT_TEST_OUT2: -- 2.29.0

2 2

Query on sysclk source
by Sameer Pujar 22 Feb '21

22 Feb '21

Hi All, I do have a query for snd_soc_dai_set_sysclk(). It takes 'clk_id' as argument which is used for sysclk source purpose in some drivers and in some it seems to be used for clock ID specifically. Please see below references. - https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/tree/soun… as sysclk source) - https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/tree/soun… as sysclk source) - https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/tree/soun… actually seems refer to specific clock id) To add snd_soc_component_sysclk() takes both 'clk_id' and 'source' arguments. Currently snd_soc_dai_set_sysclk() has following implementation. ------------------------------------------------------------------ / int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,// // unsigned int freq, int dir)// // {// // int ret;// // // if (dai->driver->ops &&// //dai->driver->ops->set_sysclk)// // ret = dai->driver->ops->set_sysclk(dai, clk_id, freq, dir);// // else// // ret = snd_soc_component_set_sysclk(dai->component, clk_id, 0,// // freq, dir);// // // return soc_dai_ret(dai, ret);// // }/ ------------------------------------------------------------------- The 'source' for snd_soc_component_set_sysclk() is hardcoded to 0 always, which may not be the case always. Please see below. - https://elixir.bootlin.com/linux/v5.11/source/sound/soc/samsung/bells.c#L161 Hence I am wondering if 'clk_id' argument is being used inconsistently for snd_soc_dai_set_sysclk() and should we rather have a separate 'source' argument for this as well? Thanks, Sameer.

2 1

[PATCH v2] ASoC: rt1316: Add RT1316 SDCA vendor-specific driver
by shumingf＠realtek.com 22 Feb '21

22 Feb '21

From: Shuming Fan <shumingf(a)realtek.com> This is the initial amplifier driver for rt1316 SDCA version. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt1316-sdw.c | 753 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1316-sdw.h | 55 +++ 4 files changed, 816 insertions(+) create mode 100644 sound/soc/codecs/rt1316-sdw.c create mode 100644 sound/soc/codecs/rt1316-sdw.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e4cf14e66a51..ae833a54bcb3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -182,6 +182,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT711_SDW imply SND_SOC_RT715_SDW imply SND_SOC_RT1308_SDW + imply SND_SOC_RT1316_SDW imply SND_SOC_SGTL5000 imply SND_SOC_SI476X imply SND_SOC_SIMPLE_AMPLIFIER @@ -1144,6 +1145,11 @@ config SND_SOC_RT1308_SDW depends on I2C && SOUNDWIRE select REGMAP_SOUNDWIRE +config SND_SOC_RT1316_SDW + tristate "Realtek RT1316 Codec - SDW" + depends on SOUNDWIRE + select REGMAP_SOUNDWIRE + config SND_SOC_RT5514 tristate depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 81357dc62ea0..1585bc1d9a25 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -173,6 +173,7 @@ snd-soc-rt1015p-objs := rt1015p.o snd-soc-rt1305-objs := rt1305.o snd-soc-rt1308-objs := rt1308.o snd-soc-rt1308-sdw-objs := rt1308-sdw.o +snd-soc-rt1316-sdw-objs := rt1316-sdw.o snd-soc-rt274-objs := rt274.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o @@ -487,6 +488,7 @@ obj-$(CONFIG_SND_SOC_RT1015P) += snd-soc-rt1015p.o obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o obj-$(CONFIG_SND_SOC_RT1308) += snd-soc-rt1308.o obj-$(CONFIG_SND_SOC_RT1308_SDW) += snd-soc-rt1308-sdw.o +obj-$(CONFIG_SND_SOC_RT1316_SDW) += snd-soc-rt1316-sdw.o obj-$(CONFIG_SND_SOC_RT274) += snd-soc-rt274.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c new file mode 100644 index 000000000000..9cf21aab1f55 --- /dev/null +++ b/sound/soc/codecs/rt1316-sdw.c @@ -0,0 +1,753 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt1316-sdw.c -- rt1316 SDCA ALSA SoC amplifier audio driver +// +// Copyright(c) 2021 Realtek Semiconductor Corp. +// +// +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/pm_runtime.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include "rt1316-sdw.h" + +static const struct reg_default rt1316_reg_defaults[] = { + { 0x3004, 0x00 }, + { 0x3005, 0x00 }, + { 0x3206, 0x00 }, + { 0xc001, 0x00 }, + { 0xc002, 0x00 }, + { 0xc003, 0x00 }, + { 0xc004, 0x00 }, + { 0xc005, 0x00 }, + { 0xc006, 0x00 }, + { 0xc007, 0x00 }, + { 0xc008, 0x00 }, + { 0xc009, 0x00 }, + { 0xc00a, 0x00 }, + { 0xc00b, 0x00 }, + { 0xc00c, 0x00 }, + { 0xc00d, 0x00 }, + { 0xc00e, 0x00 }, + { 0xc00f, 0x00 }, + { 0xc010, 0xa5 }, + { 0xc011, 0x00 }, + { 0xc012, 0xff }, + { 0xc013, 0xff }, + { 0xc014, 0x40 }, + { 0xc015, 0x00 }, + { 0xc016, 0x00 }, + { 0xc017, 0x00 }, + { 0xc605, 0x30 }, + { 0xc700, 0x0a }, + { 0xc701, 0xaa }, + { 0xc702, 0x1a }, + { 0xc703, 0x0a }, + { 0xc710, 0x80 }, + { 0xc711, 0x00 }, + { 0xc712, 0x3e }, + { 0xc713, 0x80 }, + { 0xc714, 0x80 }, + { 0xc715, 0x06 }, + { 0xd101, 0x00 }, + { 0xd102, 0x30 }, + { 0xd103, 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_UDMPU21, RT1316_SDCA_CTL_UDMPU_CLUSTER, 0), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_L), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_R), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_XU24, RT1316_SDCA_CTL_BYPASS, 0), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE23, RT1316_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE22, RT1316_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE24, RT1316_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, +}; + +static const struct reg_sequence rt1316_blind_write[] = { + { 0xc710, 0x17 }, + { 0xc711, 0x80 }, + { 0xc712, 0x26 }, + { 0xc713, 0x06 }, + { 0xc714, 0x80 }, + { 0xc715, 0x06 }, + { 0xc702, 0x0a }, + { 0xc703, 0x0a }, + { 0xc001, 0x45 }, + { 0xc003, 0x00 }, + { 0xc004, 0x11 }, + { 0xc005, 0x00 }, + { 0xc006, 0x00 }, + { 0xc106, 0x00 }, + { 0xc007, 0x11 }, + { 0xc008, 0x11 }, + { 0xc009, 0x00 }, + + { 0x2f0a, 0x00 }, + { 0xd101, 0xf0 }, + { 0xd103, 0x9b }, + { 0x2f36, 0x8e }, + { 0x3206, 0x80 }, + { 0x3211, 0x0b }, + { 0x3216, 0x06 }, + { 0xc614, 0x20 }, + { 0xc615, 0x0a }, + { 0xc616, 0x02 }, + { 0xc617, 0x00 }, + { 0xc60b, 0x10 }, + { 0xc60e, 0x05 }, + { 0xc102, 0x00 }, + { 0xc090, 0xb0 }, + { 0xc00f, 0x01 }, + { 0xc09c, 0x7b }, + + { 0xc602, 0x07 }, + { 0xc603, 0x07 }, + { 0xc0a3, 0x71 }, + { 0xc00b, 0x30 }, + { 0xc093, 0x80 }, + { 0xc09d, 0x80 }, + { 0xc0b0, 0x77 }, + { 0xc010, 0xa5 }, + { 0xc050, 0x83 }, + { 0x2f55, 0x03 }, + { 0x3217, 0xb5 }, + { 0x3202, 0x02 }, + + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_XU24, RT1316_SDCA_CTL_BYPASS, 0), 0x00 }, + + /* for IV sense */ + { 0x2232, 0x80 }, + { 0xc0b0, 0x77 }, + { 0xc011, 0x00 }, + { 0xc020, 0x00 }, + { 0xc023, 0x00 }, + { 0x3101, 0x00 }, + { 0x3004, 0xa0 }, + { 0x3005, 0xb1 }, + { 0xc007, 0x11 }, + { 0xc008, 0x11 }, + { 0xc009, 0x00 }, + { 0xc022, 0xd6 }, + { 0xc025, 0xd6 }, + + { 0xd001, 0x03 }, + { 0xd002, 0xbf }, + { 0xd003, 0x03 }, + { 0xd004, 0xbf }, +}; + +static bool rt1316_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2f0a: + case 0x2f36: + case 0x3203 ... 0x320e: + case 0xc000 ... 0xc7b4: + case 0xcf00 ... 0xcf03: + case 0xd101 ... 0xd103: + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_UDMPU21, RT1316_SDCA_CTL_UDMPU_CLUSTER, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_L): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_R): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE23, RT1316_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE27, RT1316_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE22, RT1316_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE24, RT1316_SDCA_CTL_REQ_POWER_STATE, 0): + return true; + default: + return false; + } +} + +static bool rt1316_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0xc000: + case 0xc093: + case 0xc09d: + case 0xc0a3: + case 0xc201: + case 0xc427 ... 0xc428: + case 0xd102: + return true; + default: + return false; + } +} + +static const struct regmap_config rt1316_sdw_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = rt1316_readable_register, + .volatile_reg = rt1316_volatile_register, + .max_register = 0x4108ffff, + .reg_defaults = rt1316_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rt1316_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt1316_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + prop->is_sdca = true; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x04; /* BITMAP: 00000100 */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + dev_dbg(&slave->dev, "%s\n", __func__); + + return 0; +} + +static int rt1316_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(dev); + + if (rt1316->hw_init) + return 0; + + if (rt1316->first_hw_init) { + regcache_cache_only(rt1316->regmap, false); + regcache_cache_bypass(rt1316->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + /* sw reset */ + regmap_write(rt1316->regmap, 0xc000, 0x02); + + /* initial settings - blind write */ + regmap_multi_reg_write(rt1316->regmap, rt1316_blind_write, + ARRAY_SIZE(rt1316_blind_write)); + + if (rt1316->first_hw_init) { + regcache_cache_bypass(rt1316->regmap, false); + regcache_mark_dirty(rt1316->regmap); + } else + rt1316->first_hw_init = true; + + /* Mark Slave initialization complete */ + rt1316->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + return 0; +} + +static int rt1316_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(&slave->dev); + + /* Update the status */ + rt1316->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + rt1316->hw_init = false; + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt1316->hw_init || rt1316->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt1316_io_init(&slave->dev, slave); +} + +static int rt1316_classd_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 rt1316_sdw_priv *rt1316 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE23, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE27, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE22, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE23, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE27, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE22, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + break; + + default: + break; + } + + return 0; +} + +static int rt1316_pde24_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 rt1316_sdw_priv *rt1316 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE24, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE24, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + break; + } + return 0; +} + +static const char * const rt1316_rx_data_ch_select[] = { + "L,R", + "L,L", + "L,R", + "L,L+R", + "R,L", + "R,R", + "R,L+R", + "L+R,L", + "L+R,R", + "L+R,L+R", +}; + +static SOC_ENUM_SINGLE_DECL(rt1316_rx_data_ch_enum, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_UDMPU21, RT1316_SDCA_CTL_UDMPU_CLUSTER, 0), 0, + rt1316_rx_data_ch_select); + +static const struct snd_kcontrol_new rt1316_snd_controls[] = { + + /* I2S Data Channel Selection */ + SOC_ENUM("RX Channel Select", rt1316_rx_data_ch_enum), + + /* XU24 Bypass Control */ + SOC_SINGLE("XU24 Bypass Switch", + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_XU24, RT1316_SDCA_CTL_BYPASS, 0), 0, 1, 0), + + /* Left/Right IV tag */ + SOC_SINGLE("Left V Tag Select", 0x3004, 0, 7, 0), + SOC_SINGLE("Left I Tag Select", 0x3004, 4, 7, 0), + SOC_SINGLE("Right V Tag Select", 0x3005, 0, 7, 0), + SOC_SINGLE("Right I Tag Select", 0x3005, 4, 7, 0), + + /* IV mixer Control */ + SOC_DOUBLE("Isense Mixer Switch", 0xc605, 2, 0, 1, 1), + SOC_DOUBLE("Vsense Mixer Switch", 0xc605, 3, 1, 1, 1), +}; + +static const struct snd_kcontrol_new rt1316_sto_dac_l = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_L), + 0, 1, 1); + +static const struct snd_kcontrol_new rt1316_sto_dac_r = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_R), + 0, 1, 1); + +static const struct snd_soc_dapm_widget rt1316_dapm_widgets[] = { + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("DP1RX", "DP1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP2TX", "DP2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SWITCH("DAC L", SND_SOC_NOPM, 0, 0, &rt1316_sto_dac_l), + SND_SOC_DAPM_SWITCH("DAC R", SND_SOC_NOPM, 0, 0, &rt1316_sto_dac_r), + + /* Output Lines */ + SND_SOC_DAPM_PGA_E("CLASS D", SND_SOC_NOPM, 0, 0, NULL, 0, + rt1316_classd_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + + SND_SOC_DAPM_SUPPLY("PDE 24", SND_SOC_NOPM, 0, 0, + rt1316_pde24_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA("I Sense", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("V Sense", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SIGGEN("I Gen"), + SND_SOC_DAPM_SIGGEN("V Gen"), +}; + +static const struct snd_soc_dapm_route rt1316_dapm_routes[] = { + { "DAC", NULL, "DP1RX" }, + { "DAC L", "Switch", "DAC" }, + { "DAC R", "Switch", "DAC" }, + { "CLASS D", NULL, "DAC L" }, + { "CLASS D", NULL, "DAC R" }, + { "SPOL", NULL, "CLASS D" }, + { "SPOR", NULL, "CLASS D" }, + + { "I Sense", NULL, "I Gen" }, + { "V Sense", NULL, "V Gen" }, + { "I Sense", NULL, "PDE 24" }, + { "V Sense", NULL, "PDE 24" }, + { "DP2TX", NULL, "I Sense" }, + { "DP2TX", NULL, "V Sense" }, +}; + +static int rt1316_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + struct sdw_stream_data *stream; + + if (!sdw_stream) + return 0; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = sdw_stream; + + /* Use tx_mask or rx_mask to configure stream tag and set dma_data */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = stream; + else + dai->capture_dma_data = stream; + + return 0; +} + +static void rt1316_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int rt1316_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1316_sdw_priv *rt1316 = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_data *stream; + int retval, port, num_channels, ch_mask; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!rt1316->sdw_slave) + return -EINVAL; + + /* SoundWire specific configuration */ + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + port = 1; + } else { + direction = SDW_DATA_DIR_TX; + port = 2; + } + + num_channels = params_channels(params); + ch_mask = (1 << num_channels) - 1; + + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = num_channels; + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + port_config.ch_mask = ch_mask; + port_config.num = port; + + retval = sdw_stream_add_slave(rt1316->sdw_slave, &stream_config, + &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + return 0; +} + +static int rt1316_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1316_sdw_priv *rt1316 = + snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt1316->sdw_slave) + return -EINVAL; + + sdw_stream_remove_slave(rt1316->sdw_slave, stream->sdw_stream); + return 0; +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static struct sdw_slave_ops rt1316_slave_ops = { + .read_prop = rt1316_read_prop, + .update_status = rt1316_update_status, +}; + +static const struct snd_soc_component_driver soc_component_sdw_rt1316 = { + .controls = rt1316_snd_controls, + .num_controls = ARRAY_SIZE(rt1316_snd_controls), + .dapm_widgets = rt1316_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt1316_dapm_widgets), + .dapm_routes = rt1316_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt1316_dapm_routes), +}; + +static const struct snd_soc_dai_ops rt1316_aif_dai_ops = { + .hw_params = rt1316_sdw_hw_params, + .hw_free = rt1316_sdw_pcm_hw_free, + .set_sdw_stream = rt1316_set_sdw_stream, + .shutdown = rt1316_sdw_shutdown, +}; + +#define RT1316_STEREO_RATES SNDRV_PCM_RATE_48000 +#define RT1316_FORMATS (SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver rt1316_sdw_dai[] = { + { + .name = "rt1316-aif", + .playback = { + .stream_name = "DP1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT1316_STEREO_RATES, + .formats = RT1316_FORMATS, + }, + .capture = { + .stream_name = "DP2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT1316_STEREO_RATES, + .formats = RT1316_FORMATS, + }, + .ops = &rt1316_aif_dai_ops, + }, +}; + +static int rt1316_sdw_init(struct device *dev, struct regmap *regmap, + struct sdw_slave *slave) +{ + struct rt1316_sdw_priv *rt1316; + int ret; + + rt1316 = devm_kzalloc(dev, sizeof(*rt1316), GFP_KERNEL); + if (!rt1316) + return -ENOMEM; + + dev_set_drvdata(dev, rt1316); + rt1316->sdw_slave = slave; + rt1316->regmap = regmap; + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt1316->hw_init = false; + rt1316->first_hw_init = false; + + ret = devm_snd_soc_register_component(dev, + &soc_component_sdw_rt1316, + rt1316_sdw_dai, + ARRAY_SIZE(rt1316_sdw_dai)); + + dev_dbg(&slave->dev, "%s\n", __func__); + + return ret; +} + +static int rt1316_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *regmap; + + /* Regmap Initialization */ + regmap = devm_regmap_init_sdw(slave, &rt1316_sdw_regmap); + if (!regmap) + return -EINVAL; + + return rt1316_sdw_init(&slave->dev, regmap, slave); +} + +static const struct sdw_device_id rt1316_id[] = { + SDW_SLAVE_ENTRY_EXT(0x025d, 0x1316, 0x3, 0x1, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt1316_id); + +static int __maybe_unused rt1316_dev_suspend(struct device *dev) +{ + struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(dev); + + if (!rt1316->hw_init) + return 0; + + regcache_cache_only(rt1316->regmap, true); + + return 0; +} + +#define RT1316_PROBE_TIMEOUT 5000 + +static int __maybe_unused rt1316_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt1316->hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT1316_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt1316->regmap, false); + regcache_sync(rt1316->regmap); + + return 0; +} + +static const struct dev_pm_ops rt1316_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt1316_dev_suspend, rt1316_dev_resume) + SET_RUNTIME_PM_OPS(rt1316_dev_suspend, rt1316_dev_resume, NULL) +}; + +static struct sdw_driver rt1316_sdw_driver = { + .driver = { + .name = "rt1316-sdca", + .owner = THIS_MODULE, + .pm = &rt1316_pm, + }, + .probe = rt1316_sdw_probe, + .ops = &rt1316_slave_ops, + .id_table = rt1316_id, +}; +module_sdw_driver(rt1316_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT1316 driver SDCA SDW"); +MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt1316-sdw.h b/sound/soc/codecs/rt1316-sdw.h new file mode 100644 index 000000000000..cbcdaa8f8cfa --- /dev/null +++ b/sound/soc/codecs/rt1316-sdw.h @@ -0,0 +1,55 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt1316-sdw.h -- RT1316 SDCA ALSA SoC audio driver header + * + * Copyright(c) 2021 Realtek Semiconductor Corp. + */ + +#ifndef __RT1316_SDW_H__ +#define __RT1316_SDW_H__ + +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/soundwire/sdw_registers.h> +#include <sound/soc.h> + +/* RT1316 SDCA Control - function number */ +#define FUNC_NUM_SMART_AMP 0x04 + +/* RT1316 SDCA entity */ +#define RT1316_SDCA_ENT_PDE23 0x31 +#define RT1316_SDCA_ENT_PDE27 0x32 +#define RT1316_SDCA_ENT_PDE22 0x33 +#define RT1316_SDCA_ENT_PDE24 0x34 +#define RT1316_SDCA_ENT_XU24 0x24 +#define RT1316_SDCA_ENT_FU21 0x03 +#define RT1316_SDCA_ENT_UDMPU21 0x02 + +/* RT1316 SDCA control */ +#define RT1316_SDCA_CTL_SAMPLE_FREQ_INDEX 0x10 +#define RT1316_SDCA_CTL_REQ_POWER_STATE 0x01 +#define RT1316_SDCA_CTL_BYPASS 0x01 +#define RT1316_SDCA_CTL_FU_MUTE 0x01 +#define RT1316_SDCA_CTL_FU_VOLUME 0x02 +#define RT1316_SDCA_CTL_UDMPU_CLUSTER 0x10 + +/* RT1316 SDCA channel */ +#define CH_L 0x01 +#define CH_R 0x02 + +struct rt1316_sdw_priv { + struct snd_soc_component *component; + struct regmap *regmap; + struct sdw_slave *sdw_slave; + enum sdw_slave_status status; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +#endif /* __RT1316_SDW_H__ */ -- 2.29.0

4 5

[RFC PATCH 0/2] soundwire: add static port mapping support
by Srinivas Kandagatla 22 Feb '21

22 Feb '21

Some of the soundwire controllers can have static functions assigned to each port. One Example : Master Port Num 1 and 2 are for PCM streams Master Port 3, 4 can do PDM streams Now Codecs that are on PDM should only assign Port 3 and 4. simillarly PCM Codecs should be assigned port 1 and 2. Other examples can include some usecase based mappings. In such cases its not correct to assign/map any free port on master. Currently there is no way to pass this static mapping from Slave to Master ports. This patch provides a way to pass mapped port number along with the existing port config structure during stream config, so that master can assign correct ports based on the provided static mapping. Srinivas Kandagatla (2): soundwire: add support for static port mapping soundwire: qcom: add support for static port mapping drivers/soundwire/bus.h | 4 ++++ drivers/soundwire/qcom.c | 11 +++++++++-- drivers/soundwire/stream.c | 4 ++++ include/linux/soundwire/sdw.h | 4 ++++ 4 files changed, 21 insertions(+), 2 deletions(-) -- 2.21.0

3 18

[PATCH v3 0/5] ALSA: control - add generic LED API
by Jaroslav Kysela 22 Feb '21

22 Feb '21

This patchset tries to resolve the diversity in the audio LED control among the ALSA drivers. A new control layer registration is introduced which allows to run additional operations on top of the elementary ALSA sound controls. A new control access group (three bits in the access flags) was introduced to carry the LED group information for the sound controls. The low-level sound drivers can just mark those controls using this access group. This information is exported to the user space and eventually the user space can create sound controls which can belong to a LED group. The actual state ('route') evaluation is really easy (the minimal value check for all channels / controls / cards). If there's more complicated logic for a given hardware, the card driver may eventually export a new read-only sound control for the LED group and do the logic itself. The new LED trigger control code is completely separated and possibly optional (there's no symbol dependency). The full code separation allows eventually to move this LED trigger control to the user space in future. Actually it replaces the already present functionality in the kernel space (HDA drivers) and allows a quick adoption for the recent hardware (SoundWire ASoC codecs). # lsmod | grep snd_ctl_led snd_ctl_led 16384 0 The sound driver implementation is really easy: 1) call snd_ctl_led_request() when control LED layer should be automatically activated / it calls module_request("snd-ctl-led") on demand / 2) mark all related kcontrols with SNDRV_CTL_ELEM_ACCESS_SPK_LED or SNDRV_CTL_ELEM_ACCESS_MIC_LED v3 changes: - reorder the controls_rwsem use to fix the remaining mutex issue card->controls_rwsem <-> snd_ctl_layer_rwsem v2 changes: - fix the locking - remove the controls_rwsem read lock in the element get (the consistency is already protected with the global snd_ctl_led_mutex and possible partial value writes are catched with the next value change notification callback) - rename state to brightness and show the brightness unsigned integer value instead the text on/off string (sync with the LED core routines) - remove snd_ctl_led_hello() function (CI warning) - make snd_ctl_led_get_by_access() function static (CI warning) - move snd_ctl_layer_rwsem lock before the registraction callback call in snd_ctl_register_layer() - optimization v1: - https://lore.kernel.org/alsa-devel/20210211111400.1131020-1-perex@perex.cz/ Original RFC: - https://lore.kernel.org/alsa-devel/20210207201157.869972-1-perex@perex.cz/ Cc: Hans de Goede <hdegoede(a)redhat.com> Cc: Perry Yuan <Perry.Yuan(a)dell.com> Jaroslav Kysela (5): ALSA: control - introduce snd_ctl_notify_one() helper ALSA: control - add layer registration routines ALSA: control - add generic LED trigger module as the new control layer ALSA: HDA - remove the custom implementation for the audio LED trigger ALSA: control - add sysfs support to the LED trigger module include/sound/control.h | 27 ++- include/uapi/sound/asound.h | 6 +- sound/core/Kconfig | 6 + sound/core/Makefile | 2 + sound/core/control.c | 179 ++++++++++++--- sound/core/control_led.c | 377 ++++++++++++++++++++++++++++++++ sound/pci/hda/Kconfig | 4 +- sound/pci/hda/hda_codec.c | 69 +----- sound/pci/hda/hda_generic.c | 162 +++----------- sound/pci/hda/hda_generic.h | 15 +- sound/pci/hda/hda_local.h | 16 +- sound/pci/hda/patch_ca0132.c | 4 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_sigmatel.c | 6 +- sound/pci/hda/thinkpad_helper.c | 2 +- 15 files changed, 614 insertions(+), 263 deletions(-) create mode 100644 sound/core/control_led.c -- 2.29.2

1 5

[PATCH 0/5] ALSA: control - add generic LED trigger code
by Jaroslav Kysela 22 Feb '21

22 Feb '21

This patchset tries to resolve the diversity in the audio LED control among the ALSA drivers. A new control layer registration is introduced which allows to run additional operations on top of the elementary ALSA sound controls. A new control access group (three bits in the access flags) was introduced to carry the LED group information for the sound controls. The low-level sound drivers can just mark those controls using this access group. This information is exported to the user space and eventually the user space can create sound controls which can belong to a LED group. The actual state ('route') evaluation is really easy (the minimal value check for all channels / controls / cards). If there's more complicated logic for a given hardware, the card driver may eventually export a new read-only sound control for the LED group and do the logic itself. The new LED trigger control code is completely separated and possibly optional (there's no symbol dependency). The full code separation allows eventually to move this LED trigger control to the user space in future. Actually it replaces the already present functionality in the kernel space (HDA drivers) and allows a quick adoption for the recent hardware (SoundWire ASoC codecs). # lsmod | grep snd_ctl_led snd_ctl_led 16384 0 The sound driver implementation is really easy: 1) call snd_ctl_led_request() when control LED layer should be automatically activated / it calls module_request("snd-ctl-led") on demand / 2) mark all related kcontrols with SNDRV_CTL_ELEM_ACCESS_SPK_LED or SNDRV_CTL_ELEM_ACCESS_MIC_LED Original RFC: https://lore.kernel.org/alsa-devel/20210207201157.869972-1-perex@perex.cz/ Cc: Hans de Goede <hdegoede(a)redhat.com> Cc: Perry Yuan <Perry.Yuan(a)dell.com> Jaroslav Kysela (5): ALSA: control - introduce snd_ctl_notify_one() helper ALSA: control - add layer registration routines ALSA: control - add generic LED trigger module as the new control layer ALSA: HDA - remove the custom implementation for the audio LED trigger ALSA: control - add sysfs support to the LED trigger module include/sound/control.h | 27 ++- include/uapi/sound/asound.h | 6 +- sound/core/Kconfig | 6 + sound/core/Makefile | 2 + sound/core/control.c | 173 ++++++++++++-- sound/core/control_led.c | 407 ++++++++++++++++++++++++++++++++ sound/pci/hda/Kconfig | 4 +- sound/pci/hda/hda_codec.c | 69 +----- sound/pci/hda/hda_generic.c | 162 ++----------- sound/pci/hda/hda_generic.h | 15 +- sound/pci/hda/hda_local.h | 16 +- sound/pci/hda/patch_ca0132.c | 4 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_sigmatel.c | 6 +- sound/pci/hda/thinkpad_helper.c | 2 +- 15 files changed, 638 insertions(+), 263 deletions(-) create mode 100644 sound/core/control_led.c -- 2.29.2

5 20

[PATCH v2 0/5] ALSA: control - add generic LED API
by Jaroslav Kysela 21 Feb '21

21 Feb '21

This patchset tries to resolve the diversity in the audio LED control among the ALSA drivers. A new control layer registration is introduced which allows to run additional operations on top of the elementary ALSA sound controls. A new control access group (three bits in the access flags) was introduced to carry the LED group information for the sound controls. The low-level sound drivers can just mark those controls using this access group. This information is exported to the user space and eventually the user space can create sound controls which can belong to a LED group. The actual state ('route') evaluation is really easy (the minimal value check for all channels / controls / cards). If there's more complicated logic for a given hardware, the card driver may eventually export a new read-only sound control for the LED group and do the logic itself. The new LED trigger control code is completely separated and possibly optional (there's no symbol dependency). The full code separation allows eventually to move this LED trigger control to the user space in future. Actually it replaces the already present functionality in the kernel space (HDA drivers) and allows a quick adoption for the recent hardware (SoundWire / ASoC codecs). # lsmod | grep snd_ctl_led snd_ctl_led 16384 0 The sound driver implementation is really easy: 1) call snd_ctl_led_request() when control LED layer should be automatically activated / it calls module_request("snd-ctl-led") on demand / 2) mark all related kcontrols with SNDRV_CTL_ELEM_ACCESS_SPK_LED or SNDRV_CTL_ELEM_ACCESS_MIC_LED v2 changes: - fix the locking - remove the controls_rwsem read lock in the element get (the consistency is already protected with the global snd_ctl_led_mutex and possible partial value writes are catched with the next value change notification callback) - rename state to brightness and show the brightness unsigned integer value instead the text on/off string (sync with the LED core routines) - remove snd_ctl_led_hello() function (CI warning) - make snd_ctl_led_get_by_access() function static (CI warning) - move snd_ctl_layer_rwsem lock before the registraction callback call in snd_ctl_register_layer() - optimization v1: - https://lore.kernel.org/alsa-devel/20210211111400.1131020-1-perex@perex.cz/ Original RFC: - https://lore.kernel.org/alsa-devel/20210207201157.869972-1-perex@perex.cz/ Cc: Hans de Goede <hdegoede(a)redhat.com> Cc: Perry Yuan <Perry.Yuan(a)dell.com> Jaroslav Kysela (5): ALSA: control - introduce snd_ctl_notify_one() helper ALSA: control - add layer registration routines ALSA: control - add generic LED trigger module as the new control layer ALSA: HDA - remove the custom implementation for the audio LED trigger ALSA: control - add sysfs support to the LED trigger module include/sound/control.h | 27 ++- include/uapi/sound/asound.h | 6 +- sound/core/Kconfig | 6 + sound/core/Makefile | 2 + sound/core/control.c | 173 ++++++++++++--- sound/core/control_led.c | 378 ++++++++++++++++++++++++++++++++ sound/pci/hda/Kconfig | 4 +- sound/pci/hda/hda_codec.c | 69 +----- sound/pci/hda/hda_generic.c | 162 +++----------- sound/pci/hda/hda_generic.h | 15 +- sound/pci/hda/hda_local.h | 16 +- sound/pci/hda/patch_ca0132.c | 4 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_sigmatel.c | 6 +- sound/pci/hda/thinkpad_helper.c | 2 +- 15 files changed, 609 insertions(+), 263 deletions(-) create mode 100644 sound/core/control_led.c -- 2.29.2

2 8

[PATCH v4 0/9] ALSA: add virtio sound driver
by Anton Yakovlev 21 Feb '21

21 Feb '21

This series implements a driver part of the virtio sound device specification v8 [1]. The driver supports PCM playback and capture substreams, jack and channel map controls. A message-based transport is used to write/read PCM frames to/from a device. As a device part was used OpenSynergy proprietary implementation. v4 changes: - fixed sparse warnings, no functional change [1] https://lists.oasis-open.org/archives/virtio-dev/202003/msg00185.html Anton Yakovlev (9): uapi: virtio_ids: add a sound device type ID from OASIS spec ALSA: virtio: add virtio sound driver ALSA: virtio: handling control messages ALSA: virtio: build PCM devices and substream hardware descriptors ALSA: virtio: handling control and I/O messages for the PCM device ALSA: virtio: PCM substream operators ALSA: virtio: introduce jack support ALSA: virtio: introduce PCM channel map support ALSA: virtio: introduce device suspend/resume support MAINTAINERS | 9 + include/uapi/linux/virtio_ids.h | 1 + include/uapi/linux/virtio_snd.h | 334 +++++++++++++++++++++ sound/Kconfig | 2 + sound/Makefile | 3 +- sound/virtio/Kconfig | 10 + sound/virtio/Makefile | 13 + sound/virtio/virtio_card.c | 462 +++++++++++++++++++++++++++++ sound/virtio/virtio_card.h | 113 ++++++++ sound/virtio/virtio_chmap.c | 219 ++++++++++++++ sound/virtio/virtio_ctl_msg.c | 310 ++++++++++++++++++++ sound/virtio/virtio_ctl_msg.h | 78 +++++ sound/virtio/virtio_jack.c | 233 +++++++++++++++ sound/virtio/virtio_pcm.c | 499 ++++++++++++++++++++++++++++++++ sound/virtio/virtio_pcm.h | 121 ++++++++ sound/virtio/virtio_pcm_msg.c | 392 +++++++++++++++++++++++++ sound/virtio/virtio_pcm_ops.c | 491 +++++++++++++++++++++++++++++++ 17 files changed, 3289 insertions(+), 1 deletion(-) create mode 100644 include/uapi/linux/virtio_snd.h create mode 100644 sound/virtio/Kconfig create mode 100644 sound/virtio/Makefile create mode 100644 sound/virtio/virtio_card.c create mode 100644 sound/virtio/virtio_card.h create mode 100644 sound/virtio/virtio_chmap.c create mode 100644 sound/virtio/virtio_ctl_msg.c create mode 100644 sound/virtio/virtio_ctl_msg.h create mode 100644 sound/virtio/virtio_jack.c create mode 100644 sound/virtio/virtio_pcm.c create mode 100644 sound/virtio/virtio_pcm.h create mode 100644 sound/virtio/virtio_pcm_msg.c create mode 100644 sound/virtio/virtio_pcm_ops.c -- 2.30.0

1 9

[PATCH v3 0/7] HID: lenovo: Mute LED handling fixes and improvements
by Hans de Goede 21 Feb '21

21 Feb '21

Hi All, Here is v3 of my series with mute LED handling fixes and improvements for the hid-lenovo driver. v3 addresses the review-remarks from Marek Behún, thank you for all the reviews Marek. See the individual patches for detailed per patch changelogs. Regards, Hans Hans de Goede (7): HID: lenovo: Use brightness_set_blocking callback for setting LEDs brightness HID: lenovo: Fix lenovo_led_set_tp10ubkbd() error handling HID: lenovo: Check hid_get_drvdata() returns non NULL in lenovo_event() HID: lenovo: Remove lenovo_led_brightness_get() HID: lenovo: Set LEDs max_brightness value HID: lenovo: Map mic-mute button to KEY_F20 instead of KEY_MICMUTE HID: lenovo: Set default_triggers for the mute and micmute LEDs drivers/hid/hid-lenovo.c | 69 ++++++++++++++++++++++------------------ 1 file changed, 38 insertions(+), 31 deletions(-) -- 2.30.1

3 13

[PATCH] sound/usb generate midi streaming substream names from jack names
by George Harker 21 Feb '21

21 Feb '21

A number of devices have named substreams which are hard to remember / decypher from <device> MIDI n names. Eg. Korg puts a pass through on one substream and iConnectivity devices name the connections. This makes it easier to connect to the correct device. Devices which handle naming through quirks are unaffected by this change. Addresses TODO comment in sound/usb/midi.c Signed-off-by: George Harker <george(a)george-graphics.co.uk> --- sound/usb/midi.c | 122 ++++++++++++++++++++++++++++++++++++++--------- sound/usb/midi.h | 2 + 2 files changed, 101 insertions(+), 23 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 0c23fa6d8..c6651a566 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -47,6 +47,7 @@ #include <linux/usb.h> #include <linux/wait.h> #include <linux/usb/audio.h> +#include <linux/usb/midi.h> #include <linux/module.h> #include <sound/core.h> @@ -77,23 +78,6 @@ MODULE_AUTHOR("Clemens Ladisch <clemens(a)ladisch.de>"); MODULE_DESCRIPTION("USB Audio/MIDI helper module"); MODULE_LICENSE("Dual BSD/GPL"); - -struct usb_ms_header_descriptor { - __u8 bLength; - __u8 bDescriptorType; - __u8 bDescriptorSubtype; - __u8 bcdMSC[2]; - __le16 wTotalLength; -} __attribute__ ((packed)); - -struct usb_ms_endpoint_descriptor { - __u8 bLength; - __u8 bDescriptorType; - __u8 bDescriptorSubtype; - __u8 bNumEmbMIDIJack; - __u8 baAssocJackID[]; -} __attribute__ ((packed)); - struct snd_usb_midi_in_endpoint; struct snd_usb_midi_out_endpoint; struct snd_usb_midi_endpoint; @@ -1756,12 +1740,68 @@ static void snd_usbmidi_get_port_info(struct snd_rawmidi *rmidi, int number, } } +static struct usb_midi_in_jack_descriptor *find_usb_in_jack_descriptor( + struct usb_host_interface *hostif, uint8_t jack_id) +{ + unsigned char *extra = hostif->extra; + int extralen = hostif->extralen; + + while (extralen > 4) { + struct usb_midi_in_jack_descriptor *injd = + (struct usb_midi_in_jack_descriptor *)extra; + + if (injd->bLength > 4 && + injd->bDescriptorType == USB_DT_CS_INTERFACE && + injd->bDescriptorSubtype == UAC_MIDI_IN_JACK && + injd->bJackID == jack_id) + return injd; + if (!extra[0]) + break; + extralen -= extra[0]; + extra += extra[0]; + } + return NULL; +} + +static struct usb_midi_out_jack_descriptor *find_usb_out_jack_descriptor( + struct usb_host_interface *hostif, uint8_t jack_id) +{ + unsigned char *extra = hostif->extra; + int extralen = hostif->extralen; + + while (extralen > 4) { + struct usb_midi_out_jack_descriptor *outjd = + (struct usb_midi_out_jack_descriptor *)extra; + + if (outjd->bLength > 4 && + outjd->bDescriptorType == USB_DT_CS_INTERFACE && + outjd->bDescriptorSubtype == UAC_MIDI_OUT_JACK && + outjd->bJackID == jack_id) + return outjd; + if (!extra[0]) + break; + extralen -= extra[0]; + extra += extra[0]; + } + return NULL; +} + static void snd_usbmidi_init_substream(struct snd_usb_midi *umidi, - int stream, int number, + int stream, int number, int jack_id, struct snd_rawmidi_substream **rsubstream) { struct port_info *port_info; const char *name_format; + struct usb_interface *intf; + struct usb_host_interface *hostif; + struct usb_midi_in_jack_descriptor *injd; + struct usb_midi_out_jack_descriptor *outjd; + uint8_t jack_name_buf[32]; + uint8_t *default_jack_name = "MIDI"; + uint8_t *jack_name = default_jack_name; + uint8_t iJack; + size_t sz; + int res; struct snd_rawmidi_substream *substream = snd_usbmidi_find_substream(umidi, stream, number); @@ -1771,11 +1811,35 @@ static void snd_usbmidi_init_substream(struct snd_usb_midi *umidi, return; } - /* TODO: read port name from jack descriptor */ + intf = umidi->iface; + if (intf && jack_id >= 0) { + hostif = intf->cur_altsetting; + iJack = 0; + if (stream != SNDRV_RAWMIDI_STREAM_OUTPUT) { + /* in jacks connect to outs */ + outjd = find_usb_out_jack_descriptor(hostif, jack_id); + if (outjd) { + sz = USB_DT_MIDI_OUT_SIZE(outjd->bNrInputPins); + iJack = *(((uint8_t *) outjd) + sz - 1); + } + } else { + /* and out jacks connect to ins */ + injd = find_usb_in_jack_descriptor(hostif, jack_id); + if (injd) + iJack = injd->iJack; + } + if (iJack != 0) { + res = usb_string(umidi->dev, iJack, jack_name_buf, 32); + if (res) + jack_name = jack_name_buf; + } + } + port_info = find_port_info(umidi, number); - name_format = port_info ? port_info->name : "%s MIDI %d"; + name_format = port_info ? port_info->name : + (jack_name != default_jack_name ? "%s %s" : "%s %s %d"); snprintf(substream->name, sizeof(substream->name), - name_format, umidi->card->shortname, number + 1); + name_format, umidi->card->shortname, jack_name, number + 1); *rsubstream = substream; } @@ -1810,6 +1874,7 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi, snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_OUTPUT, out_ports, + endpoints[i].assoc_out_jacks[j], &umidi->endpoints[i].out->ports[j].substream); ++out_ports; } @@ -1817,6 +1882,7 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi, snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_INPUT, in_ports, + endpoints[i].assoc_in_jacks[j], &umidi->endpoints[i].in->ports[j].substream); ++in_ports; } @@ -1862,7 +1928,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, struct usb_host_endpoint *hostep; struct usb_endpoint_descriptor *ep; struct usb_ms_endpoint_descriptor *ms_ep; - int i, epidx; + int i, j, epidx; intf = umidi->iface; if (!intf) @@ -1875,7 +1941,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ms_header->bDescriptorType == USB_DT_CS_INTERFACE && ms_header->bDescriptorSubtype == UAC_HEADER) dev_dbg(&umidi->dev->dev, "MIDIStreaming version %02x.%02x\n", - ms_header->bcdMSC[1], ms_header->bcdMSC[0]); + ((uint8_t *)&ms_header->bcdMSC)[1], ((uint8_t *)&ms_header->bcdMSC)[0]); else dev_warn(&umidi->dev->dev, "MIDIStreaming interface descriptor not found\n"); @@ -1911,6 +1977,10 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, endpoints[epidx].out_interval = 1; endpoints[epidx].out_cables = (1 << ms_ep->bNumEmbMIDIJack) - 1; + for (j = 0; j < ms_ep->bNumEmbMIDIJack; ++j) + endpoints[epidx].assoc_out_jacks[j] = ms_ep->baAssocJackID[j]; + for (; j < 0x10; ++j) + endpoints[epidx].assoc_out_jacks[j] = -1; dev_dbg(&umidi->dev->dev, "EP %02X: %d jack(s)\n", ep->bEndpointAddress, ms_ep->bNumEmbMIDIJack); } else { @@ -1928,6 +1998,10 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, endpoints[epidx].in_interval = 1; endpoints[epidx].in_cables = (1 << ms_ep->bNumEmbMIDIJack) - 1; + for (j = 0; j < ms_ep->bNumEmbMIDIJack; ++j) + endpoints[epidx].assoc_in_jacks[j] = ms_ep->baAssocJackID[j]; + for (; j < 0x10; ++j) + endpoints[epidx].assoc_in_jacks[j] = -1; dev_dbg(&umidi->dev->dev, "EP %02X: %d jack(s)\n", ep->bEndpointAddress, ms_ep->bNumEmbMIDIJack); } @@ -2244,11 +2318,13 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi *umidi, snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_OUTPUT, cable, + -1 /* prevent trying to find jack */, &umidi->endpoints[cable & 1].out->ports[cable].substream); if (endpoint->in_cables & (1 << cable)) snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_INPUT, cable, + -1 /* prevent trying to find jack */, &umidi->endpoints[0].in->ports[cable].substream); } return 0; diff --git a/sound/usb/midi.h b/sound/usb/midi.h index 8c38aec22..3f153195c 100644 --- a/sound/usb/midi.h +++ b/sound/usb/midi.h @@ -13,6 +13,8 @@ struct snd_usb_midi_endpoint_info { uint8_t in_interval; uint16_t out_cables; /* bitmask */ uint16_t in_cables; /* bitmask */ + int16_t assoc_in_jacks[16]; + int16_t assoc_out_jacks[16]; }; /* for QUIRK_MIDI_YAMAHA, data is NULL */ -- 2.20.1

1 0

[PATCH 1/3] HID: elan: Silence mute LED errors being logged when the device is unplugged
by Hans de Goede 20 Feb '21

20 Feb '21

Being integrated into an USB keyboard-dock the mute LED can go away at any time, leading to the following errors: [ 918.667671] elan 0003:04F3:0755.0002: Failed to set mute led brightness: -19 [ 918.667737] leds elan:red:mute: Setting an LED's brightness failed (-19) Fix this by making the following changes: 1. Don't log an error from elan_mute_led_set_brigtness() when ret == ENODEV 2. Set the LED_HW_PLUGGABLE flag on the mute LED led_classdev Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- drivers/hid/hid-elan.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/drivers/hid/hid-elan.c b/drivers/hid/hid-elan.c index dae193749d44..71740988a3f6 100644 --- a/drivers/hid/hid-elan.c +++ b/drivers/hid/hid-elan.c @@ -445,7 +445,8 @@ static int elan_mute_led_set_brigtness(struct led_classdev *led_cdev, kfree(dmabuf); if (ret != ELAN_LED_REPORT_SIZE) { - hid_err(hdev, "Failed to set mute led brightness: %d\n", ret); + if (ret != -ENODEV) + hid_err(hdev, "Failed to set mute led brightness: %d\n", ret); return ret; } @@ -462,6 +463,7 @@ static int elan_init_mute_led(struct hid_device *hdev) mute_led->brightness_get = elan_mute_led_get_brigtness; mute_led->brightness_set_blocking = elan_mute_led_set_brigtness; mute_led->max_brightness = LED_ON; + mute_led->flags = LED_HW_PLUGGABLE; mute_led->dev = &hdev->dev; return devm_led_classdev_register(&hdev->dev, mute_led); -- 2.30.1

1 2

[PATCH] leds: trigger: audio: Add an activate callback to ensure the initial brightness is set
by Hans de Goede 20 Feb '21

20 Feb '21

Some 2-in-1s with a detachable (USB) keyboard(dock) have mute-LEDs in the speaker- and/or mic-mute keys on the keyboard. Examples of this are the Lenovo Thinkpad10 tablet (with its USB kbd-dock) and the HP x2 10 series. The detachable nature of these keyboards means that the keyboard and thus the mute LEDs may show up after the user (or userspace restoring old mixer settings) has muted the speaker and/or mic. Current LED-class devices with a default_trigger of "audio-mute" or "audio-micmute" initialize the brightness member of led_classdev with ledtrig_audio_get() before registering the LED. This makes the software state after attaching the keyboard match the actual audio mute state, e.g. cat /sys/class/leds/foo/brightness will show the right value. But before this commit nothing was actually calling the led_classdev's brightness_set[_blocking] callback so the value returned by ledtrig_audio_get() was never actually being send to the hw, leading to the mute LEDs staying in their default power-on state, after attaching the keyboard, even if ledtrig_audio_get() returns a different state. This could be fixed by having the individual LED drivers call brightness_set[_blocking] themselves after registering the LED, but this really is something which should be done by a led-trigger activate callback. Add an activate callback for this, fixing the issue of the mute LEDs being out of sync after (re)attaching the keyboard. Cc: Takashi Iwai <tiwai(a)suse.de> Fixes: faa2541f5b1a ("leds: trigger: Introduce audio mute LED trigger") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- drivers/leds/trigger/ledtrig-audio.c | 37 ++++++++++++++++++++++------ 1 file changed, 29 insertions(+), 8 deletions(-) diff --git a/drivers/leds/trigger/ledtrig-audio.c b/drivers/leds/trigger/ledtrig-audio.c index f76621e88482..c6b437e6369b 100644 --- a/drivers/leds/trigger/ledtrig-audio.c +++ b/drivers/leds/trigger/ledtrig-audio.c @@ -6,10 +6,33 @@ #include <linux/kernel.h> #include <linux/leds.h> #include <linux/module.h> +#include "../leds.h" -static struct led_trigger *ledtrig_audio[NUM_AUDIO_LEDS]; static enum led_brightness audio_state[NUM_AUDIO_LEDS]; +static int ledtrig_audio_mute_activate(struct led_classdev *led_cdev) +{ + led_set_brightness_nosleep(led_cdev, audio_state[LED_AUDIO_MUTE]); + return 0; +} + +static int ledtrig_audio_micmute_activate(struct led_classdev *led_cdev) +{ + led_set_brightness_nosleep(led_cdev, audio_state[LED_AUDIO_MICMUTE]); + return 0; +} + +static struct led_trigger ledtrig_audio[NUM_AUDIO_LEDS] = { + [LED_AUDIO_MUTE] = { + .name = "audio-mute", + .activate = ledtrig_audio_mute_activate, + }, + [LED_AUDIO_MICMUTE] = { + .name = "audio-micmute", + .activate = ledtrig_audio_micmute_activate, + }, +}; + enum led_brightness ledtrig_audio_get(enum led_audio type) { return audio_state[type]; @@ -19,24 +42,22 @@ EXPORT_SYMBOL_GPL(ledtrig_audio_get); void ledtrig_audio_set(enum led_audio type, enum led_brightness state) { audio_state[type] = state; - led_trigger_event(ledtrig_audio[type], state); + led_trigger_event(&ledtrig_audio[type], state); } EXPORT_SYMBOL_GPL(ledtrig_audio_set); static int __init ledtrig_audio_init(void) { - led_trigger_register_simple("audio-mute", - &ledtrig_audio[LED_AUDIO_MUTE]); - led_trigger_register_simple("audio-micmute", - &ledtrig_audio[LED_AUDIO_MICMUTE]); + led_trigger_register(&ledtrig_audio[LED_AUDIO_MUTE]); + led_trigger_register(&ledtrig_audio[LED_AUDIO_MICMUTE]); return 0; } module_init(ledtrig_audio_init); static void __exit ledtrig_audio_exit(void) { - led_trigger_unregister_simple(ledtrig_audio[LED_AUDIO_MUTE]); - led_trigger_unregister_simple(ledtrig_audio[LED_AUDIO_MICMUTE]); + led_trigger_unregister(&ledtrig_audio[LED_AUDIO_MUTE]); + led_trigger_unregister(&ledtrig_audio[LED_AUDIO_MICMUTE]); } module_exit(ledtrig_audio_exit); -- 2.30.1

1 0

What is the semantics of snd_pcm_hw_params_any?
by GitHub issues - opened 20 Feb '21

20 Feb '21

alsa-project/alsa-lib issue #119 was opened from jpalus: It is not really a bug report but rather a request for clarification in attempt to figure out who is to blame for incorrect pulseaudio behavior. I've got two different cards -- one internal and one USB -- which behave quite differently for this simple high level code: ``` snd_pcm_open() snd_pcm_hw_params_any() iterate snd_pcm_hw_params_test_rate() to check for supported rates snd_pcm_hw_params_set_rate_near(44100) snd_pcm_hw_params() snd_pcm_hw_params_any() iterate snd_pcm_hw_params_test_rate() to check for supported rates ``` In essence supported sample rates are queried twice, always with snd_pcm_hw_params_any() before them, but separated by snd_pcm_hw_params_set_rate_near(). Now with internal card both iterations report 7 supported rates while for USB first iteration reports 4 rates and second only one (44100). Is that expected or is snd_pcm_hw_params_any() supposed to provide configuration which would result in all rates reported again? Issue URL : https://github.com/alsa-project/alsa-lib/issues/119 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: Intel: Skylake: Fix missing check in skl_pcm_trigger
by Dinghao Liu 19 Feb '21

19 Feb '21

When cmd == SNDRV_PCM_TRIGGER_STOP, we should also check the return value of skl_decoupled_trigger() just like what we have done in case SNDRV_PCM_TRIGGER_PAUSE_RELEASE. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/soc/intel/skylake/skl-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index b1ca64d2f7ea..a5b1f333a500 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -516,6 +516,9 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, return ret; ret = skl_decoupled_trigger(substream, cmd); + if (ret < 0) + return ret; + if ((cmd == SNDRV_PCM_TRIGGER_SUSPEND) && !w->ignore_suspend) { /* save the dpib and lpib positions */ stream->dpib = readl(bus->remap_addr + -- 2.17.1

3 2

[PATCH] ASoC: imx-hdmi: no need to set .owner when using module_platform_driver
by Tian Tao 19 Feb '21

19 Feb '21

the module_platform_driver will call platform_driver_register. and It will set the .owner to THIS_MODULE Signed-off-by: Tian Tao <tiantao6(a)hisilicon.com> --- sound/soc/fsl/imx-hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index dbbb7618..cd0235a 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -223,7 +223,6 @@ MODULE_DEVICE_TABLE(of, imx_hdmi_dt_ids); static struct platform_driver imx_hdmi_driver = { .driver = { .name = "imx-hdmi", - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, .of_match_table = imx_hdmi_dt_ids, }, -- 2.7.4

2 1

[PATCH 0/2] dmic fix to allow userspace to choose ch count
by vamshi.krishna.gopal＠intel.com 18 Feb '21

18 Feb '21

From: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.corp-partner.google.com> hello, git.kernel.org/pub/scm/linux/kernel/git/vkoul/sound.git/commit/?h=intel2&id… git.kernel.org/pub/scm/linux/kernel/git/vkoul/sound.git/commit/?h=intel2&id… These patch series is about removing channel count selection based on mach/pdata and assign min and max channel only based on hw params. This allows userspcae to choose channel count. This patch series was Acked by Vinod and taken into his tree. But seems its missed to be posted. hence i've rebased the same commits and re sending. Naveen Manohar (2): ASoC: Intel: kbl: Remove option of choosing CH count based on pdata ASoC: Intel: kbl: update dmic fixup params sound/soc/intel/boards/kbl_rt5663_max98927.c | 44 +++++++------------- 1 file changed, 14 insertions(+), 30 deletions(-) -- 2.17.1

1 2

Re: [PATCH] ASoC: rt1316: Add RT1316 SDCA vendor-specific driver
by Mark Brown 18 Feb '21

18 Feb '21

On Tue, Feb 09, 2021 at 01:34:24AM +0000, Shuming [范書銘] wrote: > > Right, this was my expectation for how this would work - dynamic control at > > runtime so you can do stuff like that rather than a more static setup like slot > > assignment is normally.. > Actually, the "RX Channel Select" control could select the channel. > This control could select the different channels to output dynamically. > However, could I keep the rt1316_sdw_set_tdm_slot() here? > There will be more options to set this configuration. If there's a control to override it at runtime then duplicating that with set_tdm_slot() seems like it's going to be a route to confusion if people change it?

2 1

[PATCH] ASoC: soc-pcm: fix hw param limits calculation for multi-DAI
by Kai Vehmanen 18 Feb '21

18 Feb '21

In case DPCM runtime has multiple CPU DAIs, dpcm_init_runtime_hw() is called multiple times, once for each CPU DAI. This will will lead to ignoring hw limits of all but the first DAI. Fix this by moving soc_pcm_hw_init() up by one level to dpcm_init_runtime_hw(). Fixes: 140f553d1298 ("ASoC: soc-pcm: fix hwparams min/max init for dpcm") Suggested-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/soc-pcm.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 37094aeff440..14d85ca1e435 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1531,10 +1531,12 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, { struct snd_pcm_hardware *hw = &runtime->hw; - soc_pcm_hw_init(hw); soc_pcm_hw_update_rate(hw, stream); soc_pcm_hw_update_chan(hw, stream); - soc_pcm_hw_update_format(hw, stream); + if (runtime->hw.formats) + runtime->hw.formats &= stream->formats; + else + runtime->hw.formats = stream->formats; } static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, @@ -1662,10 +1664,13 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_hardware *hw = &runtime->hw; struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai; int i; + soc_pcm_hw_init(hw); + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { /* * Skip CPUs which don't support the current stream base-commit: 140f553d1298e0ddbe21983efe5789fe8b484273 -- 2.29.2

2 1

[PATCH] ASoC: rt5682: remove jack detect delay
by Curtis Malainey 17 Feb '21

17 Feb '21

There is a 250ms delay on the jack detect interrupt currently, this delay is observable to users who are using inline controls. It can also mask multiple presses which is a negative experience. Cc: Bard liao <yung-chuan.liao(a)linux.intel.com> Cc: Shuming Fan <shumingf(a)realtek.com> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5682-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 93c1603b42f1..b15c3e7d1f59 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -78,7 +78,7 @@ static irqreturn_t rt5682_irq(int irq, void *data) struct rt5682_priv *rt5682 = data; mod_delayed_work(system_power_efficient_wq, - &rt5682->jack_detect_work, msecs_to_jiffies(250)); + &rt5682->jack_detect_work, 0); return IRQ_HANDLED; } -- 2.30.0.478.g8a0d178c01-goog

2 1

ucm2: add support to for Qualcomm RB5 Platform
by GitHub pull_request - opened 17 Feb '21

17 Feb '21

alsa-project/alsa-ucm-conf pull request #84 was opened from lumag: Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/84 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/84.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC updates for v5.12
by Mark Brown 17 Feb '21

17 Feb '21

The following changes since commit 92bf22614b21a2706f4993b278017e437f7785b3: Linux 5.11-rc7 (2021-02-07 13:57:38 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.12 for you to fetch changes up to 0969db0d8d15caee41cd817154670c38d9ed7f61: Merge remote-tracking branch 'asoc/for-5.12' into asoc-linus (2021-02-17 18:52:26 +0000) ---------------------------------------------------------------- ASoC: Updates for v5.12 Another quiet release in terms of features, though several of the drivers got quite a bit of work and there were a lot of general changes resulting from Morimoto-san's ongoing cleanup work. - As ever, lots of hard work by Morimoto-san cleaning up the code and making it more consistent. - Many improvements in the Intel drivers including a wide range of quirks and bug fixes. - A KUnit testsuite for the topology code. - Support for Ingenic JZ4760(B), Intel AlderLake-P, DT configured nVidia cards, Qualcomm lpass-rx-macro and lpass-tx-macro - Removal of obsolete SIRF prima/atlas, Txx9 and ZTE zx drivers. ---------------------------------------------------------------- Adam Ford (1): ASoC: wm8962: Add optional mclk device tree binding Amadeusz Sławiński (6): ASoC: topology: KUnit: Add KUnit tests passing various arguments to snd_soc_tplg_component_load ASoC: topology: KUnit: Add KUnit tests passing empty topology with variants to snd_soc_tplg_component_load ASoC: topology: KUnit: Add KUnit tests passing topology with PCM to snd_soc_tplg_component_load ASoC: topology: Ensure that needed parameters are set ASoC: topology: Check if ops is set before dereference ASoC: topology: KUnit: Convert from cpu to data format Andy Shevchenko (2): catpt: Switch to use list_entry_is_head() helper ASoC: sprd: Switch to use list_entry_is_head() helper Arnd Bergmann (4): ASoC: mediatek: add MTK_PMIC_WRAP dependency ASoC: fsl_aud2htx: select SND_SOC_IMX_PCM_DMA ASoC: remove sirf prima/atlas drivers ASoC: remove zte zx drivers Bard Liao (8): ASoC: SOF: Intel: hda: use snd_sof_dsp_core_power_up/down API ASoC: SOF: Intel: hda-loader: keep init cores alive ASoC: SOF: update dsp core power status in common APIs ASoC: SOF: Filter out unneeded core power up/downs ASoC: SOF: intel: hda-loader: use snd_sof_dsp_core_power_down/up APIs ASoC: rt5682: do nothing in rt5682_suspend/resume in sdw mode ASoC: Intel: boards: max98373: get dapm from cpu_dai ASoC: intel: sof_sdw: add trace for dai links Bjorn Helgaas (1): ASoC: q6asm: fix 'physical' typos Charles Keepax (1): ASoC: wm_adsp: Remove unused control callback structure Chris Chiu (4): ASoC: rt5645: Introduce mapping for ACPI-defined GPIO ASoC: rt5645: Add ACPI-defined GPIO for ECS EF20 series ASoC: rt5645: add inv_hp_det flag ASoC: rt5645: Enable internal microphone and JD on ECS EF20 Christophe Branchereau (1): ASoC: codec: Add driver for JZ4760 internal codec Christophe JAILLET (1): ASoC: Intel: common: Fix some typos Curtis Malainey (2): ASoC: SOF: add missing pm debug ASoC: SOF: fix string format for errors Dan Carpenter (2): ASoC: soc-pcm: Fix an uninitialized error code ASoC: cs42l56: fix up error handling in probe Dmitry Osipenko (3): ASoC: tegra: ahub: Add missing resets ASoC: tegra: ahub: Use clk_bulk helpers ASoC: tegra: ahub: Reset hardware properly Fabio Estevam (7): ASoC: fsl_ssi: Use of_device_get_match_data() ASoC: fsl_micfil: Use of_device_get_match_data() ASoC: fsl_xcvr: Remove unused of_id variable ASoC: fsl_asrc: Remove of_device_get_match_data() error check ASoC: fsl_esai: Remove of_device_get_match_data() error check ASoC: fsl_spdif: Remove of_device_get_match_data() error check ASoC: fsl_esai: Remove unused 'imx' field Fred Oh (1): ASoC: SOF: ext_manifest: use explicit number for elem_type Geert Uytterhoeven (2): ASoC: tegra: SND_SOC_TEGRA_AUDIO_GRAPH_CARD should depend on SND_SOC_TEGRA ASoC: dt-bindings: renesas, rsnd: Group tuples in playback/capture properties Guennadi Liakhovetski (3): ASoC: SOF: remove unused functions ASoC: SOF: HDA: (cosmetic) simplify hda_dsp_d0i3_work() ASoC: SOF: Intel: HDA: don't keep a temporary variable Hans de Goede (14): ASoC: Intel: bytcht_es8316: Remove comment about SSP0 being untested ASoC: Intel: bytcr_rt5640: Add new BYT_RT5640_NO_SPEAKERS quirk-flag ASoC: Intel: bytcr_rt5640: Add new BYT_RT5640_NO_INTERNAL_MIC_MAP input-mapping ASoC: Intel: bytcr_rt5640: Add quirk for the Mele PCG03 Mini PC ASoC: Intel: cht_bsw_nau8824: Move snd_soc_dai_set_tdm_slot call to cht_codec_fixup ASoC: es8316: Fix possible NULL pointer deref in es8316_disable_jack_detect() mfd: arizona: Add MODULE_SOFTDEP("pre: arizona_ldo1") mfd: arizona: Replace arizona_of_get_type() with device_get_match_data() mfd: arizona: Add support for ACPI enumeration of WM5102 connected over SPI ASoC: Intel: Add DMI quirk table to soc_intel_is_byt_cr() ASoC: Intel: bytcr_rt5640: Add quirk for the Estar Beauty HD MID 7316R tablet ASoC: Intel: bytcr_rt5640: Add quirk for the Voyo Winpad A15 tablet ASoC: Intel: bytcr_rt5651: Add quirk for the Jumper EZpad 7 tablet ASoC: Intel: bytcr_rt5640: Add quirk for the Acer One S1002 tablet Hui Wang (1): ASoC: SOF: debug: Fix a potential issue on string buffer termination Jack Yu (1): ASoC: rt1015: modify calibration sequence for better performance Jiaxin Yu (1): ASoC: mediatek: mt8183-mt6358: set playback and capture constraints Judy Hsiao (1): ASoC: max98373: Fixes a typo in max98373_feedback_get Jun Nie (1): ASoC: qcom: lpass: Fix i2s ctl register bit map Kai Vehmanen (7): ASoC: SOF: Intel: initial support to AlderLake-P ASoC: SOF: add a pointer to download repo in case FW request fails ASoC: Intel: remove duplicate MODULE_LICENSE/DESCRIPTION tags ASoC: SOF: fix runtime pm usage mismatch after probe errors ASoC: SOF: relax ABI checks and avoid unnecessary warnings ASoC: soc-pcm: fix hwparams min/max init for dpcm ASoC: soc-pcm: fix hw param limits calculation for multi-DAI Keyon Jie (5): ASoC: SOF: add mutex to protect the dsp_power_state access ASoC: SOF: add .shutdown() callback to snd_sof_dsp_ops ASoC: SOF: add snd_sof_device_shutdown() helper for shutdown ASoC: SOF: sof-pci-dev: add .shutdown() callback ASoC: SOF: Intel: tgl: do thorough remove at .shutdown() callback Kuninori Morimoto (56): ASoC: soc-pcm: move dpcm_set_fe_update_state() ASoC: soc-pcm: add dpcm_set_be_update_state() ASoC: soc-pcm: add soc_pcm_set_dai_params() ASoC: soc-pcm: cleanup soc_pcm_apply_symmetry() ASoC: soc-pcm: revert soc_pcm_apply_symmetry() ASoC: sync parameter naming : rate / sample_bits ASoC: adi: sync parameter naming (rate/sample_bits) ASoC: atmel: sync parameter naming (rate/sample_bits) ASoC: au1x: sync parameter naming (rate/sample_bits) ASoC: bcm: sync parameter naming (rate/sample_bits) ASoC: cirrus: sync parameter naming (rate/sample_bits) ASoC: tegra: sync parameter naming (rate/sample_bits) ASoC: rockchip: sync parameter naming (rate/sample_bits) ASoC: samsung: sync parameter naming (rate/sample_bits) ASoC: sh: sync parameter naming (rate/sample_bits) ASoC: ti: sync parameter naming (rate/sample_bits) ASoC: pxa: sync parameter naming (rate/sample_bits) ASoC: mediatek: sync parameter naming (rate/sample_bits) ASoC: fsl: sync parameter naming (rate/sample_bits) ASoC: wm*: sync parameter naming (rate/sample_bits) ASoC: tlv*: sync parameter naming (rate/sample_bits) ASoC: rt*: sync parameter naming (rate/sample_bits) ASoC: nau*: sync parameter naming (rate/sample_bits) ASoC: tas*: sync parameter naming (rate/sample_bits) ASoC: da*: sync parameter naming (rate/sample_bits) ASoC: es*: sync parameter naming (rate/sample_bits) ASoC: max*: sync parameter naming (rate/sample_bits) ASoC: cs*: sync parameter naming (rate/sample_bits) ASoC: tscs*: sync parameter naming (rate/sample_bits) ASoC: ak*: sync parameter naming (rate/sample_bits) ASoC: adau*: sync parameter naming (rate/sample_bits) ASoC: jz4740: sync parameter naming (rate/sample_bits) ASoC: sunxi: sync parameter naming (rate/sample_bits) ASoC: zl38060: sync parameter naming (rate/sample_bits) ASoC: ssm2602: sync parameter naming (rate/sample_bits) ASoC: sgtl5000: sync parameter naming (rate/sample_bits) ASoC: mc13783: sync parameter naming (rate/sample_bits) ASoC: mt6660: sync parameter naming (rate/sample_bits) ASoC: lochnagar-sc: sync parameter naming (rate/sample_bits) ASoC: ab8500: sync parameter naming (rate/sample_bits) ASoC: alc5632: sync parameter naming (rate/sample_bits) ASoC: cx2072x: sync parameter naming (rate/sample_bits) ASoC: inno_rk3036: sync parameter naming (rate/sample_bits) ASoC: lm49453: sync parameter naming (rate/sample_bits) ASoC: ml26124: sync parameter naming (rate/sample_bits) ASoC: soc-dai.h: remove symmetric_rates/samplebits ASoC: soc-pcm: cleanup soc_pcm_apply_symmetry() ASoC: soc-pcm: cleanup soc_pcm_params_symmetry() ASoC: soc-pcm: tidyup pcm setting ASoC: soc-pcm: add soc_get_playback_capture() and simplify soc_new_pcm() ASoC: soc-pcm: add soc_create_pcm() and simplify soc_new_pcm() ASoC: soc-pcm: use snd_pcm_hardware at dpcm_runtime_merge_xxx() ASoC: soc-pcm: fixup snd_pcm_limit_hw_rates() timing ASoC: soc-pcm: add soc_pcm_hw_update_rate() ASoC: soc-pcm: add soc_pcm_hw_update_chan() ASoC: soc-pcm: add soc_pcm_hw_update_format() Lars-Peter Clausen (4): ASoC: meson: Use managed DMA buffer allocation ASoC: atmel-pdc: Use managed DMA buffer allocation ASoC: bcm: cygnus: Use managed DMA buffer allocation ASoC: kirkwood: Use managed DMA buffer allocation Libin Yang (3): ASoC: SOF: allow soundwire use desc->default_fw_filename ASoC: Intel: tgl: remove sof_fw_filename set for tgl_3_in_1_default ASoC: Intel: adl: remove sof_fw_filename setting in ADL snd_soc_acpi_mach Mark Brown (35): Merge existing fixes from asoc/for-5.11 Merge series "ASoC: rt1015: apply some refactors" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge series "Enable DMA mode on Intel Keem Bay platform" from Michael Sit Wei Hong <michael.wei.hong.sit(a)intel.com>: Merge series "ASoC: rt5645: Enable internal mic and headset on ECS EF20" from Chris Chiu <chiu(a)endlessos.org>: Merge series "ASoC: Intel: bytcr_rt5640: Add support for devices with only a headphone jack (no speakers/mic)" from Hans de Goede <hdegoede(a)redhat.com>: Merge series "ASoC: soc-pcm: cleanup code" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge v5.11-rc3 Merge series "ASoC: qcom: Minor code cleanups for lpass-cpu" from Stephen Boyd <swboyd(a)chromium.org>: Merge series "ASoC/SoundWire: fix timeout values" from Bard Liao <yung-chuan.liao(a)linux.intel.com>: Merge series "ASoC: Simplify with dma_set_mask_and_coherent()" from Takashi Iwai <tiwai(a)suse.de>: Merge series "Tegra210 audio graph card" from Sameer Pujar <spujar(a)nvidia.com>: Merge series "ASoC: remove obsolete drivers" from Arnd Bergmann <arnd(a)kernel.org> Arnd Bergmann <arnd(a)arndb.de>: Merge series "ASoC: mediatek: mt8192-mt6359: support DP audio" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge series "Add KUNIT tests for ASoC topology" from Amadeusz Sławiński<amadeuszx.slawinski(a)linux.intel.com>: Merge series "ASoC: sync parameter naming : rate / sample_bits" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "Add sanity checks for topology API calls" from Amadeusz Sławiński<amadeuszx.slawinski(a)linux.intel.com>: Merge series "ASoC: SOF: Intel: hda: Enable DMI L1 for power savings" from Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com>: Merge series "ASoC: SOF: use common code for DSP core pm" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge series "Tegra186 and Tegra194 audio graph card" from Sameer Pujar <spujar(a)nvidia.com>: Merge series "ASoC: soc-pcm: cleanup soc_new_pcm() and bugfix" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: mediatek: mt8192: apply some cleanup" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge series "ASoC: SOF/Intel/SoundWire: add missing quirks and DMIC support" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge branch 'asoc-5.11' into asoc-5.12 Merge series "ASoC: stm32: i2s: add master clock provider" from Olivier Moysan <olivier.moysan(a)foss.st.com>: Merge series "Rename audio graph export functions" from Sameer Pujar <spujar(a)nvidia.com>: Merge series "Add HDMI support for Intel KeemBay I2S" from Sia Jee Heng <jee.heng.sia(a)intel.com>: Merge tag 'ib-mfd-asoc-v5.12' of git://git.kernel.org/pub/scm/linux/kernel/git/lee/mfd into asoc-5.12 Merge series "MFD/ASoC: Add support for Intel Bay Trail boards with WM5102 codec" from Hans de Goede <hdegoede(a)redhat.com>: Merge series "ASoC: SOF/Intel/SoundWire: add missing quirks and DMIC support" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: SOF: cleanups" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: dapm/pins: handle component prefix" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "Use clocks property in a device node" from Sameer Pujar <spujar(a)nvidia.com>: Merge series "ASoC: Intel: bytcr_rt5640: Add quirks for 4 more tablet / 2-in-1 models" from Hans de Goede <hdegoede(a)redhat.com>: Merge remote-tracking branch 'asoc/for-5.11' into asoc-linus Merge remote-tracking branch 'asoc/for-5.12' into asoc-linus Michael Sit Wei Hong (3): ASoC: Intel: KMB: Enable DMA transfer mode ASoC: intel, keembay-i2s: Add info for device to use DMA ASoC: intel, keembay-i2s: Fix dt binding errors Oder Chiou (1): ASoC: rt5645: Remove the redundant delay time Olivier Moysan (2): ASoC: dt-bindings: add mclk provider support to stm32 i2s ASoC: stm32: i2s: add master clock provider Paul Cercueil (2): ASoC: codec/ingenic: Depend on MACH_INGENIC ASoC: Add compatible strings for JZ4760(B) SoC Pierre-Louis Bossart (17): ASoC: codecs: soundwire: increase resume timeout ASoC: Intel: sof_sdw: add missing TGL_HDMI quirk for Dell SKU 0A5E ASoC: Intel: sof_sdw: add missing TGL_HDMI quirk for Dell SKU 0A32 ASoC: Intel: sof_sdw: add missing TGL_HDMI quirk for Dell SKU 0A3E ASoC: rt700-sdw: use cancel_work_sync() in .remove as well as .suspend ASoC: rt711-sdw: use cancel_work_sync() for .remove ASoC: rt5682-sdw: cancel_work_sync() in .remove and .suspend ASoC: Intel: bytcr_wm5102: Add machine driver for BYT/WM5102 ASoC: SOF: sof-pci-dev: add missing Up-Extreme quirk ASoC: Intel: sof_sdw: reorganize quirks by generation ASoC: Intel: sof-sdw: indent and add quirks consistently ASoC: Intel: sof_sdw: add quirk for HP Spectre x360 convertible ASoC: Intel: sof_sdw: add mic:dmic and cfg-mics component strings ASoC: Intel: soc-acpi: add ACPI matching table for HP Spectre x360 ASoC: SOF: Intel: SoundWire: refine ACPI match ASoC: SOF: Intel: detect DMIC number in SoundWire mixed config ASoC: SOF: Intel: hda: add dev_dbg() when DMIC number is overridden Rander Wang (2): ASoC: SOF: add be_hw_params_fixup() for ALH ASoC: Intel: sof_sdw: detect DMIC number based on mach params Ranjani Sridharan (4): ASoC: SOF: Intel: hda: enable DMI L1 for D0i3-compatible streams ASoC: SOF: Intel: hda: Enable DMI L1 for trace ASoC: SOF: Intel: hda: cancel D0i3 work during runtime suspend ASoC: SOF: topology: Prevent NULL pointer dereference with no TLV Richard Fitzgerald (4): ASoC: wm_adsp: Improve handling of raw byte streams ASoC: wm_adsp: Only use __be32 for big-endian data ASoC: wm_adsp: Use snd_ctl_elem_type_t for control types ASoC: wm_adsp: Fix uninitialized variable warnings Rikard Falkeborn (1): ASoC: fsl: constify static snd_soc_dai_ops structs Sameer Pujar (12): ASoC: audio-graph-card: Drop remote-endpoint as required property ASoC: dt-bindings: tegra: Add graph bindings ASoC: dt-bindings: tegra: Add json-schema for Tegra audio graph card ASoC: tegra: Add audio graph based card driver ASoC: dt-bindings: rt5659: Update binding doc ASoC: dt-bindings: tegra: Add iommus property to Tegra graph card ASoC: audio-graph-card: Add clocks property to endpoint node ASoC: rt5659: Add Kconfig prompt ASoC: audio-graph: Rename functions needed for export ASoC: tegra: Add driver remove() callback ASoC: simple-card-utils: Fix device module clock ASoC: audio-graph-card: Drop device node clocks property Sathyanarayana Nujella (1): ASoC: rt5682: Fix panic in rt5682_jack_detect_handler happening during system shutdown Sebastian Reichel (1): ASoC: cpcap: fix microphone timeslot mask Shengjiu Wang (2): ASoC: dmaengine_pcm: add peripheral configuration ASoC: soc-pcm: change error message to debug message Shuming Fan (4): ASoC: rt5682: fix getting the wrong device id when the suspend_stress_test ASoC: rt5682: enable fast discharge for headset unplugging ASoC: rt5682: remove connection with LDO2 in DAPM graph ASoC: dapm: use component prefix when checking widget names Sia Jee Heng (3): ASoC: codec: hdmi-codec: Support IEC958 encoded PCM format ASoC: Intel: KMB: Support IEC958 encoded PCM format ASoC: Intel, Keembay-i2s: Add hdmi-i2s compatible string Souptick Joarder (1): ASoC: soc-pcm: return correct -ERRNO in failure path Srinivas Kandagatla (11): ASoC: soc-component: add snd_soc_component_read/write_field() ASoC: codecs: lpass-wsa-macro: make use of snd_soc_component_read_field() ASoC: soc-component: fix undefined reference to __ffssi2 ASoC: codecs: add missing max_register in regmap config ASoC: qcom: dt-bindings: add bindings for lpass rx macro codec ASoC: codecs: lpass-rx-macro: add support for lpass rx macro ASoC: codecs: lpass-rx-macro: add dapm widgets and route ASoC: codecs: lpass-rx-macro: add iir widgets ASoC: qcom: dt-bindings: add bindings for lpass tx macro codec ASoC: codecs: lpass-tx-macro: add support for lpass tx macro ASoC: codecs: lpass-tx-macro: add dapm widgets and route Srinivasa Rao Mandadapu (2): ASoC: qcom: lpass-cpu: Remove bit clock state check ASoC: qcom: Fix typo error in HDMI regmap config callbacks Stephan Gerhold (1): ASoC: qcom: qdsp6: Move frontend AIFs to q6asm-dai Stephen Boyd (5): ASoC: qcom: Remove useless debug print ASoC: qcom: Add some names to regmap configs ASoC: qcom: Stop casting away __iomem for error pointers ASoC: qcom: Remove duplicate error messages on ioremap ASoC: da7218: Drop CONFIG_OF ifdef Takashi Iwai (4): ASoC: intel: skl: Simplify with dma_set_mask_and_coherent() ASoC: SOF: intel: Simplify with dma_set_mask_and_coherent() ASoC: tegra: Simplify with dma_set_mask_and_coherent() ASoC: siu: Fix build error by a wrong const prefix Tang Bin (1): ASoC: fsl_spdif: Utilize the defined parameter to clear code Thomas Bogendoerfer (1): ASoC: txx9: Remove driver Tony Lindgren (1): ASoC: cpcap: Implement set_tdm_slot for voice call support Tzung-Bi Shih (16): ASoC: rt1015: sort header inclusions ASoC: rt1015: save boost_mode only if valid ASoC: rt1015: return error if any when setting bypass_boost ASoC: rt1015: refactor retry loop and rt1015_priv allocation ASoC: rt1015: remove unneeded variables in rt1015_priv ASoC: rt1015: re-calibrate again when resuming ASoC: rt1015: remove bclk_ratio ASoC: mediatek: mt8192-mt6359: move headset_jack to card specific data ASoC: mediatek: mt8192-mt6359: simplify mt8192_rt5682_init ASoC: mediatek: mt8192: change mclk_multiple of TDM from 128 to 512 ASoC: dt-bindings: mt8192-mt6359: add hdmi-codec property ASoC: mediatek: mt8192-mt6359: support audio over DP ASoC: mt6359: reduce log verbosity for optional DT properties ASoC: mediatek: mt8192-mt6359: use asoc_substream_to_rtd() ASoC: mediatek: mt8192: use asoc_substream_to_rtd() ASoC: mediatek: mt8192-mt6359: simplify ops for Capture1 DAI link Xu Wang (2): ASoC: es8328: Remove redundant null check before clk_disable_unprepare ASoC: adau17x1: Remove redundant null check before clk_disable_unprepare Yang Li (1): ASoC: Intel: catpt: remove unneeded semicolon Yu-Hsuan Hsu (2): cros_ec_commands: Add EC_CODEC_I2S_RX_RESET ASoC: cros_ec_codec: Reset I2S RX when probing 朱灿灿 (1): ASoC: soc-pcm: disconnect BEs if the FE is not ready .../bindings/sound/audio-graph-port.yaml | 3 - .../devicetree/bindings/sound/ingenic,codec.yaml | 11 +- .../bindings/sound/intel,keembay-i2s.yaml | 13 + .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 5 + .../sound/nvidia,tegra-audio-graph-card.yaml | 190 ++ .../bindings/sound/nvidia,tegra186-dspk.yaml | 18 +- .../bindings/sound/nvidia,tegra210-admaif.yaml | 13 +- .../bindings/sound/nvidia,tegra210-ahub.yaml | 13 +- .../bindings/sound/nvidia,tegra210-dmic.yaml | 18 +- .../bindings/sound/nvidia,tegra210-i2s.yaml | 18 +- .../bindings/sound/qcom,lpass-rx-macro.yaml | 62 + .../bindings/sound/qcom,lpass-tx-macro.yaml | 67 + .../devicetree/bindings/sound/renesas,rsnd.yaml | 6 +- Documentation/devicetree/bindings/sound/rt5659.txt | 11 + .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 - .../devicetree/bindings/sound/sirf-usp.txt | 27 - .../devicetree/bindings/sound/st,stm32-i2s.yaml | 4 + Documentation/devicetree/bindings/sound/wm8962.txt | 4 + .../devicetree/bindings/sound/zte,tdm.txt | 30 - .../devicetree/bindings/sound/zte,zx-aud96p22.txt | 24 - .../devicetree/bindings/sound/zte,zx-i2s.txt | 45 - .../devicetree/bindings/sound/zte,zx-spdif.txt | 27 - drivers/mfd/arizona-core.c | 11 - drivers/mfd/arizona-i2c.c | 11 +- drivers/mfd/arizona-spi.c | 138 +- drivers/mfd/arizona.h | 9 - include/linux/platform_data/cros_ec_commands.h | 1 + include/sound/dmaengine_pcm.h | 5 + include/sound/graph_card.h | 6 +- include/sound/hdmi-codec.h | 5 + include/sound/rt5645.h | 2 + include/sound/soc-component.h | 6 + include/sound/soc-dai.h | 4 +- include/sound/soc.h | 4 +- include/sound/sof/ext_manifest.h | 6 +- sound/core/pcm_dmaengine.c | 2 + sound/soc/Kconfig | 20 +- sound/soc/Makefile | 8 +- sound/soc/adi/axi-i2s.c | 2 +- sound/soc/amd/acp3x-rt5682-max9836.c | 4 +- sound/soc/atmel/atmel-i2s.c | 2 +- sound/soc/atmel/atmel-pcm-pdc.c | 78 +- sound/soc/atmel/mchp-i2s-mcc.c | 4 +- sound/soc/au1x/i2sc.c | 2 +- sound/soc/bcm/bcm2835-i2s.c | 4 +- sound/soc/bcm/bcm63xx-i2s-whistler.c | 2 +- sound/soc/bcm/cygnus-pcm.c | 107 +- sound/soc/cirrus/ep93xx-i2s.c | 2 +- sound/soc/codecs/Kconfig | 31 +- sound/soc/codecs/Makefile | 10 +- sound/soc/codecs/ab8500-codec.c | 4 +- sound/soc/codecs/adau1372.c | 2 +- sound/soc/codecs/adau1373.c | 6 +- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/adau17x1.c | 3 +- sound/soc/codecs/ak4554.c | 2 +- sound/soc/codecs/ak4613.c | 2 +- sound/soc/codecs/ak4641.c | 4 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/cpcap.c | 139 +- sound/soc/codecs/cros_ec_codec.c | 12 + sound/soc/codecs/cs35l32.c | 2 +- sound/soc/codecs/cs35l33.c | 2 +- sound/soc/codecs/cs35l34.c | 2 +- sound/soc/codecs/cs35l35.c | 2 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs4234.c | 2 +- sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/cs42l56.c | 3 +- sound/soc/codecs/cs42l73.c | 6 +- sound/soc/codecs/cs43130.c | 6 +- sound/soc/codecs/cs4341.c | 2 +- sound/soc/codecs/cs4349.c | 2 +- sound/soc/codecs/cs47l15.c | 12 +- sound/soc/codecs/cs47l24.c | 12 +- sound/soc/codecs/cs47l35.c | 12 +- sound/soc/codecs/cs47l85.c | 16 +- sound/soc/codecs/cs47l90.c | 16 +- sound/soc/codecs/cs47l92.c | 12 +- sound/soc/codecs/cs53l30.c | 2 +- sound/soc/codecs/cx2072x.c | 2 +- sound/soc/codecs/da7210.c | 2 +- sound/soc/codecs/da7213.c | 2 +- sound/soc/codecs/da7218.c | 8 +- sound/soc/codecs/da7219.c | 4 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/es8316.c | 5 +- sound/soc/codecs/es8328.c | 5 +- sound/soc/codecs/hdmi-codec.c | 4 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/jz4740.c | 2 +- sound/soc/codecs/jz4760.c | 889 +++++ sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/lochnagar-sc.c | 12 +- sound/soc/codecs/lpass-rx-macro.c | 3599 ++++++++++++++++++++ sound/soc/codecs/lpass-tx-macro.c | 1862 ++++++++++ sound/soc/codecs/lpass-wsa-macro.c | 43 +- sound/soc/codecs/max98373-sdw.c | 4 +- sound/soc/codecs/max98373.c | 2 +- sound/soc/codecs/max9860.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/mc13783.c | 2 +- sound/soc/codecs/ml26124.c | 2 +- sound/soc/codecs/mt6359.c | 18 +- sound/soc/codecs/mt6660.c | 4 +- sound/soc/codecs/nau8810.c | 2 +- sound/soc/codecs/nau8822.c | 2 +- sound/soc/codecs/rt1015.c | 122 +- sound/soc/codecs/rt1015.h | 5 - sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt274.c | 2 +- sound/soc/codecs/rt286.c | 4 +- sound/soc/codecs/rt298.c | 4 +- sound/soc/codecs/rt5645.c | 78 +- sound/soc/codecs/rt5670.c | 4 +- sound/soc/codecs/rt5682-i2c.c | 3 + sound/soc/codecs/rt5682-sdw.c | 25 +- sound/soc/codecs/rt5682.c | 16 +- sound/soc/codecs/rt5682.h | 2 +- sound/soc/codecs/rt700-sdw.c | 6 +- sound/soc/codecs/rt711-sdw.c | 6 +- sound/soc/codecs/rt715-sdw.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sirf-audio-codec.c | 575 ---- sound/soc/codecs/ssm2602.c | 4 +- sound/soc/codecs/tas2764.c | 2 +- sound/soc/codecs/tas2770.c | 2 +- sound/soc/codecs/tlv320adcx140.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 4 +- sound/soc/codecs/tlv320aic32x4.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tscs42xx.c | 4 +- sound/soc/codecs/tscs454.c | 12 +- sound/soc/codecs/wm5102.c | 12 +- sound/soc/codecs/wm5110.c | 12 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 4 +- sound/soc/codecs/wm8997.c | 8 +- sound/soc/codecs/wm8998.c | 12 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/wm_adsp.c | 108 +- sound/soc/codecs/wmfw.h | 6 +- sound/soc/codecs/wsa881x.c | 1 + sound/soc/codecs/zl38060.c | 4 +- sound/soc/codecs/zx_aud96p22.c | 401 --- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_asrc.c | 5 - sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 13 +- sound/soc/fsl/fsl_micfil.c | 9 +- sound/soc/fsl/fsl_sai.c | 8 +- sound/soc/fsl/fsl_spdif.c | 14 +- sound/soc/fsl/fsl_ssi.c | 12 +- sound/soc/fsl/fsl_xcvr.c | 7 +- sound/soc/generic/audio-graph-card.c | 17 +- sound/soc/generic/simple-card-utils.c | 13 +- sound/soc/intel/Kconfig | 2 + sound/soc/intel/boards/Kconfig | 12 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcht_es8316.c | 3 - sound/soc/intel/boards/bytcr_rt5640.c | 89 +- sound/soc/intel/boards/bytcr_rt5651.c | 13 + sound/soc/intel/boards/bytcr_wm5102.c | 465 +++ sound/soc/intel/boards/cht_bsw_nau8824.c | 15 +- sound/soc/intel/boards/sof_maxim_common.c | 5 +- sound/soc/intel/boards/sof_rt5682.c | 7 - sound/soc/intel/boards/sof_sdw.c | 131 +- sound/soc/intel/catpt/pcm.c | 14 +- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 1 - sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 3 - sound/soc/intel/common/soc-acpi-intel-byt-match.c | 19 +- sound/soc/intel/common/soc-acpi-intel-cfl-match.c | 3 - sound/soc/intel/common/soc-acpi-intel-cht-match.c | 3 - sound/soc/intel/common/soc-acpi-intel-cml-match.c | 3 - sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 3 - sound/soc/intel/common/soc-acpi-intel-ehl-match.c | 5 +- sound/soc/intel/common/soc-acpi-intel-glk-match.c | 3 - .../intel/common/soc-acpi-intel-hsw-bdw-match.c | 3 - sound/soc/intel/common/soc-acpi-intel-icl-match.c | 3 - sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 3 - sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 3 - sound/soc/intel/common/soc-acpi-intel-skl-match.c | 3 - sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 32 +- sound/soc/intel/common/soc-intel-quirks.h | 25 + sound/soc/intel/keembay/kmb_platform.c | 230 +- sound/soc/intel/keembay/kmb_platform.h | 10 + sound/soc/intel/skylake/skl.c | 8 +- sound/soc/jz4740/jz4740-i2s.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 79 +- sound/soc/mediatek/Kconfig | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 10 +- sound/soc/mediatek/mt6797/mt6797-dai-pcm.c | 8 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 6 - sound/soc/mediatek/mt8183/mt8183-dai-pcm.c | 8 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 47 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 4 +- sound/soc/mediatek/mt8192/mt8192-dai-pcm.c | 8 +- sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 2 +- .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 84 +- sound/soc/meson/aiu-fifo-i2s.c | 1 - sound/soc/meson/aiu-fifo-spdif.c | 1 - sound/soc/meson/aiu-fifo.c | 18 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/qcom/lpass-apq8016.c | 2 +- sound/soc/qcom/lpass-cpu.c | 47 +- sound/soc/qcom/lpass-lpaif-reg.h | 3 - sound/soc/qcom/lpass-sc7180.c | 2 +- sound/soc/qcom/lpass.h | 1 - sound/soc/qcom/qdsp6/q6asm-dai.c | 21 + sound/soc/qcom/qdsp6/q6asm.c | 2 +- sound/soc/qcom/qdsp6/q6routing.c | 18 - sound/soc/rockchip/rockchip_i2s.c | 4 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/pcm.c | 2 +- sound/soc/sh/rcar/core.c | 4 +- sound/soc/sh/siu.h | 2 +- sound/soc/sh/siu_pcm.c | 2 +- sound/soc/sirf/Kconfig | 21 - sound/soc/sirf/Makefile | 8 - sound/soc/sirf/sirf-audio-port.c | 86 - sound/soc/sirf/sirf-audio.c | 160 - sound/soc/sirf/sirf-usp.c | 435 --- sound/soc/sirf/sirf-usp.h | 292 -- sound/soc/soc-component.c | 54 + sound/soc/soc-dapm.c | 13 +- sound/soc/soc-pcm.c | 465 ++- sound/soc/soc-topology-test.c | 843 +++++ sound/soc/soc-topology.c | 30 +- sound/soc/sof/core.c | 19 + sound/soc/sof/debug.c | 2 +- sound/soc/sof/intel/hda-compress.c | 4 +- sound/soc/sof/intel/hda-dsp.c | 24 +- sound/soc/sof/intel/hda-loader.c | 17 +- sound/soc/sof/intel/hda-pcm.c | 18 +- sound/soc/sof/intel/hda-stream.c | 38 +- sound/soc/sof/intel/hda-trace.c | 8 +- sound/soc/sof/intel/hda.c | 277 +- sound/soc/sof/intel/hda.h | 7 +- sound/soc/sof/intel/tgl.c | 3 +- sound/soc/sof/ipc.c | 4 +- sound/soc/sof/loader.c | 8 +- sound/soc/sof/ops.h | 43 +- sound/soc/sof/pcm.c | 7 +- sound/soc/sof/pm.c | 1 - sound/soc/sof/sof-pci-dev.c | 22 +- sound/soc/sof/sof-priv.h | 8 +- sound/soc/sof/topology.c | 14 +- sound/soc/sprd/sprd-mcdt.c | 10 +- sound/soc/stm/stm32_i2s.c | 310 +- sound/soc/sunxi/sun4i-i2s.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 12 +- sound/soc/tegra/Kconfig | 42 +- sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra186_dspk.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra210_dmic.c | 2 +- sound/soc/tegra/tegra210_i2s.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 64 +- sound/soc/tegra/tegra30_ahub.h | 5 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/tegra/tegra_audio_graph_card.c | 252 ++ sound/soc/tegra/tegra_pcm.c | 6 +- sound/soc/ti/davinci-mcasp.c | 2 +- sound/soc/txx9/Kconfig | 30 - sound/soc/txx9/Makefile | 12 - sound/soc/txx9/txx9aclc-ac97.c | 230 -- sound/soc/txx9/txx9aclc-generic.c | 88 - sound/soc/txx9/txx9aclc.c | 422 --- sound/soc/txx9/txx9aclc.h | 71 - sound/soc/zte/Kconfig | 26 - sound/soc/zte/Makefile | 4 - sound/soc/zte/zx-i2s.c | 452 --- sound/soc/zte/zx-spdif.c | 363 -- sound/soc/zte/zx-tdm.c | 458 --- 291 files changed, 10822 insertions(+), 5986 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt delete mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,tdm.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,zx-i2s.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,zx-spdif.txt create mode 100644 sound/soc/codecs/jz4760.c create mode 100644 sound/soc/codecs/lpass-rx-macro.c create mode 100644 sound/soc/codecs/lpass-tx-macro.c delete mode 100644 sound/soc/codecs/sirf-audio-codec.c delete mode 100644 sound/soc/codecs/zx_aud96p22.c create mode 100644 sound/soc/intel/boards/bytcr_wm5102.c delete mode 100644 sound/soc/sirf/Kconfig delete mode 100644 sound/soc/sirf/Makefile delete mode 100644 sound/soc/sirf/sirf-audio-port.c delete mode 100644 sound/soc/sirf/sirf-audio.c delete mode 100644 sound/soc/sirf/sirf-usp.c delete mode 100644 sound/soc/sirf/sirf-usp.h create mode 100644 sound/soc/soc-topology-test.c create mode 100644 sound/soc/tegra/tegra_audio_graph_card.c delete mode 100644 sound/soc/txx9/Kconfig delete mode 100644 sound/soc/txx9/Makefile delete mode 100644 sound/soc/txx9/txx9aclc-ac97.c delete mode 100644 sound/soc/txx9/txx9aclc-generic.c delete mode 100644 sound/soc/txx9/txx9aclc.c delete mode 100644 sound/soc/txx9/txx9aclc.h delete mode 100644 sound/soc/zte/Kconfig delete mode 100644 sound/soc/zte/Makefile delete mode 100644 sound/soc/zte/zx-i2s.c delete mode 100644 sound/soc/zte/zx-spdif.c delete mode 100644 sound/soc/zte/zx-tdm.c

2 1

[PATCH v2] ASoC: soc-pcm: fix hw param limits calculation for multi-DAI
by Kai Vehmanen 17 Feb '21

17 Feb '21

In case DPCM runtime has multiple CPU DAIs, dpcm_init_runtime_hw() is called multiple times, once for each CPU DAI. This will lead to ignoring hw limits of all but the last DAI. Fix this by moving soc_pcm_hw_init() up by one level to dpcm_init_runtime_hw(). Fixes: 140f553d1298 ("ASoC: soc-pcm: fix hwparams min/max init for dpcm") Suggested-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/soc-pcm.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) v2 changes: - fixed incorrect text in commit message, no functional change diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 37094aeff440..14d85ca1e435 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1531,10 +1531,12 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, { struct snd_pcm_hardware *hw = &runtime->hw; - soc_pcm_hw_init(hw); soc_pcm_hw_update_rate(hw, stream); soc_pcm_hw_update_chan(hw, stream); - soc_pcm_hw_update_format(hw, stream); + if (runtime->hw.formats) + runtime->hw.formats &= stream->formats; + else + runtime->hw.formats = stream->formats; } static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, @@ -1662,10 +1664,13 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_hardware *hw = &runtime->hw; struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai; int i; + soc_pcm_hw_init(hw); + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { /* * Skip CPUs which don't support the current stream base-commit: 140f553d1298e0ddbe21983efe5789fe8b484273 -- 2.29.2

2 1

[PATCH 0/4] ASoC: Intel: bytcr_rt5640: Add quirks for 4 more tablet / 2-in-1 models
by Hans de Goede 17 Feb '21

17 Feb '21

Hi All, Here is a patch series adding quirks with device-specific settings for 4 more tablet / 2-in-1 models. Regards, Hans Hans de Goede (4): ASoC: Intel: bytcr_rt5640: Add quirk for the Estar Beauty HD MID 7316R tablet ASoC: Intel: bytcr_rt5640: Add quirk for the Voyo Winpad A15 tablet ASoC: Intel: bytcr_rt5651: Add quirk for the Jumper EZpad 7 tablet ASoC: Intel: bytcr_rt5640: Add quirk for the Acer One S1002 tablet sound/soc/intel/boards/bytcr_rt5640.c | 37 +++++++++++++++++++++++++++ sound/soc/intel/boards/bytcr_rt5651.c | 13 ++++++++++ 2 files changed, 50 insertions(+) -- 2.30.1

3 6

[PATCH v4 1/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 17 Feb '21

17 Feb '21

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. CPU0 CPU1 dpcm_be_dai_trigger(): dpcm_be_disconnect(): for_each_dpcm_be(fe, stream, dpcm) { spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); struct snd_soc_pcm_runtime *be = dpcm->be; <-- Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_be_dai_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ee51dc7fd893..718f6b3a309a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2074,12 +2074,17 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; + unsigned long flags; int ret = 0; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -2102,7 +2107,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2112,7 +2117,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2122,7 +2127,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2131,12 +2136,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2144,12 +2149,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2157,17 +2162,20 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret) + break; } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); return ret; } @@ -2905,10 +2913,9 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; int ret = 1; - unsigned long flags; int i; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); + lockdep_assert_held(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -2922,17 +2929,12 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, } } } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to do this BE DAI */ return ret; } -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { @@ -2943,6 +2945,23 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + unsigned long flags; + int ret; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = dpcm_can_be_free_stop(fe, be, stream); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -2952,6 +2971,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { + unsigned long flags; + int ret; + const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, SND_SOC_DPCM_STATE_PAUSED, @@ -2959,6 +2981,10 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, SND_SOC_DPCM_STATE_PREPARE, }; - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); -- 2.21.0

2 1

[PATCH v4 0/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 17 Feb '21

17 Feb '21

On 2021-01-06 at 12:41, Mark Brown wrote: >On Mon, Dec 28, 2020 at 08:23:16PM +0900, Gyeongtaek Lee wrote: >> If stop by underrun and DPCM BE disconnection is run simultaneously, >> data abort can be occurred by the sequence below. > >This doesn't apply against current code, please check and resend. Hi, Mark. I resend the patch after rebasing it on the latest of stable linux. Best regards, Gyeongtaek Lee Gyeongtaek Lee (1): ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger() sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) base-commit: f40ddce88593482919761f74910f42f4b84c004b -- 2.21.0

1 0

[PATCH 0/3] ASoC: soc-pcm: add soc_pcm_hw_update_xxx()
by Kuninori Morimoto 16 Feb '21

16 Feb '21

Hi Mark These adds soc_pcm_hw_update_xxx() and cleanup soc-pcm.c Kuninori Morimoto (3): ASoC: soc-pcm: add soc_pcm_hw_update_rate() ASoC: soc-pcm: add soc_pcm_hw_update_chan() ASoC: soc-pcm: add soc_pcm_hw_update_format() sound/soc/soc-pcm.c | 112 ++++++++++++++++++++++---------------------- 1 file changed, 56 insertions(+), 56 deletions(-) -- 2.25.1

4 14

[PATCH] ASoC: soc-pcm: fix hwparams min/max init for dpcm
by Kai Vehmanen 15 Feb '21

15 Feb '21

When runtime is initialized with dpcm_init_runtime_hw(), some of the min/max calculations assume that defaults are set. For example calculation of channel min/max values may be done using zero-initialized data and soc_pcm_hw_update_chan() will always return max-channels of 0 in this case. This will result in failure to open the PCM at all. Fix the issue by calling soc_pcm_hw_init() before calling any soc_pcm_hw_update_*() functions. Remove the conditional code on runtime->hw.formats as this field is anyways set in soc_pcm_hw_init(). Fixes: 6cb56a4549e9 ("ASoC: soc-pcm: add soc_pcm_hw_update_chan()") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/soc-pcm.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) Notes: - This fixes regressions caught by SOF CI. Matching SOF CI test run with this patch applied is at: https://github.com/thesofproject/linux/pull/2756 diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b7c90da1820e..37094aeff440 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1531,12 +1531,10 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, { struct snd_pcm_hardware *hw = &runtime->hw; + soc_pcm_hw_init(hw); soc_pcm_hw_update_rate(hw, stream); soc_pcm_hw_update_chan(hw, stream); - if (runtime->hw.formats) - runtime->hw.formats &= stream->formats; - else - runtime->hw.formats = stream->formats; + soc_pcm_hw_update_format(hw, stream); } static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, base-commit: debc71f26cdbd45798c63b0dcdabdea93d2f6870 -- 2.29.2

2 1

[PATCH] ALSA: hda/realtek: Quirk for HP Spectre x360 14 amp setup
by Takashi Iwai 15 Feb '21

15 Feb '21

HP Spectre x360 14 model (PCI SSID 103c:87f7) seems requiring a unique setup for its external amp: the GPIO0 needs to be toggled on and off shortly at each device initialization via runtime PM. This patch implements that workaround as well as the model option string, so that users with other devices may try the same workaround more easily. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210633 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 52907506e16e..1927605f0f7e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4291,6 +4291,28 @@ static void alc280_fixup_hp_gpio4(struct hda_codec *codec, } } +/* HP Spectre x360 14 model needs a unique workaround for enabling the amp; + * it needs to toggle the GPIO0 once on and off at each time (bko#210633) + */ +static void alc245_fixup_hp_x360_amp(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->gpio_mask |= 0x01; + spec->gpio_dir |= 0x01; + break; + case HDA_FIXUP_ACT_INIT: + /* need to toggle GPIO to enable the amp */ + alc_update_gpio_data(codec, 0x01, true); + msleep(100); + alc_update_gpio_data(codec, 0x01, false); + break; + } +} + static void alc_update_coef_led(struct hda_codec *codec, struct alc_coef_led *led, bool polarity, bool on) @@ -6277,6 +6299,7 @@ enum { ALC280_FIXUP_HP_DOCK_PINS, ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, ALC280_FIXUP_HP_9480M, + ALC245_FIXUP_HP_X360_AMP, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13, @@ -6982,6 +7005,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc280_fixup_hp_9480m, }, + [ALC245_FIXUP_HP_X360_AMP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc245_fixup_hp_x360_amp, + }, [ALC288_FIXUP_DELL_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_dell_alc288, @@ -7996,6 +8023,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -8368,6 +8396,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"}, {.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"}, {.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"}, + {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.26.2

1 0

[PATCH v3 3/3] ASoC: rt715:add micmute led state control supports
by Perry Yuan 14 Feb '21

14 Feb '21

From: Perry Yuan <perry_yuan(a)dell.com> Some new Dell system is going to support audio internal micphone privacy setting from hardware level with micmute led state changing When micmute hotkey pressed by user, soft mute will need to be enabled firstly in case of pop noise, and codec driver need to react to mic mute event to EC(embedded controller) notifying that SW mute is completed Then EC will do the hardware mute physically within the timeout reached This patch allow codec rt715 driver to ack EC when micmute key pressed through this micphone led control interface like hda_generic provided ACPI method defined in dell-privacy micmute led trigger will be called for notifying the EC that software mute has been completed Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> -------- v2 -> v3 * simplify the patch to reuse some val value * add more detail to the commit info v1 -> v2: * fix some format issue -------- --- sound/soc/codecs/rt715-sdca.c | 16 ++++++++++++++++ sound/soc/codecs/rt715-sdca.h | 1 + sound/soc/codecs/rt715.c | 14 ++++++++++++++ sound/soc/codecs/rt715.h | 1 + 4 files changed, 32 insertions(+) diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index b43ac8559e45..861a0d2a8957 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -12,6 +12,7 @@ #include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> +#include <linux/leds.h> #include <linux/pm_runtime.h> #include <linux/pm.h> #include <linux/soundwire/sdw.h> @@ -244,6 +245,7 @@ static int rt715_sdca_get_volsw(struct snd_kcontrol *kcontrol, unsigned int max = mc->max; int val; + pr_err("++++++rt715_sdca_get_volsw++\n"); val = snd_soc_component_read(component, mc->reg); if (val < 0) return -EINVAL; @@ -261,6 +263,7 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_sdca_priv *rt715 = snd_soc_component_get_drvdata(component); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int val, val2, loop_cnt = 2, i; @@ -268,6 +271,7 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, unsigned int reg2 = mc->rreg; unsigned int reg = mc->reg; unsigned int max = mc->max; + unsigned int val0, val1; int err; val = ucontrol->value.integer.value[0]; @@ -287,6 +291,18 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, return err; } +#if IS_ENABLED(CONFIG_DELL_PRIVACY) + /* dell privacy LED trigger state changed by muted/unmute switch */ + if (mc->invert) { + if (ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]) { + rt715->micmute_led = LED_OFF; + } else { + rt715->micmute_led = LED_ON; + } + ledtrig_audio_set(LED_AUDIO_MICMUTE, rt715->micmute_led); + } +#endif + return 0; } diff --git a/sound/soc/codecs/rt715-sdca.h b/sound/soc/codecs/rt715-sdca.h index 840c237895dd..f8988ab88f80 100644 --- a/sound/soc/codecs/rt715-sdca.h +++ b/sound/soc/codecs/rt715-sdca.h @@ -31,6 +31,7 @@ struct rt715_sdca_priv { int l_is_unmute; int r_is_unmute; int hw_sdw_ver; + bool micmute_led; }; struct rt715_sdw_stream_data { diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index cdcba70146da..b4e480744c94 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -13,6 +13,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/i2c.h> +#include <linux/leds.h> #include <linux/pm_runtime.h> #include <linux/pm.h> #include <linux/soundwire/sdw.h> @@ -88,6 +89,7 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, RT715_SET_GAIN_MIX_ADC2_L}; unsigned int addr_h, addr_l, val_h, val_ll, val_lr; unsigned int read_ll, read_rl, i, j, loop_cnt; + unsigned int val0, val1; if (strstr(ucontrol->id.name, "Main Capture Switch") || strstr(ucontrol->id.name, "Main Capture Volume")) @@ -95,6 +97,18 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, else loop_cnt = 1; +#if IS_ENABLED(CONFIG_DELL_PRIVACY) + /* Micmute LED state changed by muted/unmute switch */ + if (mc->invert) { + if (ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]) { + rt715->micmute_led = LED_OFF; + } else { + rt715->micmute_led = LED_ON; + } + ledtrig_audio_set(LED_AUDIO_MICMUTE, rt715->micmute_led); + } +#endif + for (j = 0; j < loop_cnt; j++) { /* Can't use update bit function, so read the original value first */ if (loop_cnt == 1) { diff --git a/sound/soc/codecs/rt715.h b/sound/soc/codecs/rt715.h index 009a8266f606..57c9af041181 100644 --- a/sound/soc/codecs/rt715.h +++ b/sound/soc/codecs/rt715.h @@ -22,6 +22,7 @@ struct rt715_priv { struct sdw_bus_params params; bool hw_init; bool first_hw_init; + bool micmute_led; }; struct sdw_stream_data { -- 2.25.1

8 13

amixer: Expand on channel docs in man page
by GitHub pull_request - opened 14 Feb '21

14 Feb '21

alsa-project/alsa-utils pull request #78 was opened from mcampbellizo: Add missing channel params to the amixer man page. Also call out that the channel param must come before the value to take effect. signed-off-by: Matthew Campbell <mcampbell(a)izotope.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/78 Patch URL : https://github.com/alsa-project/alsa-utils/pull/78.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: usb-audio: Add implicit fb quirk for BOSS GP-10
by Takashi Iwai 14 Feb '21

14 Feb '21

BOSS GP-10 with 0582:0185 requires the similar quirk to make the implicit feedback working like other BOSS devices. Reported-by: Keith Milner <kamilner(a)superlative.org> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/implicit.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 521cc846d9d9..bba54430e6d0 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -73,6 +73,7 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { /* No quirk for playback but with capture quirk (see below) */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x0130), /* BOSS BR-80 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x0171), /* BOSS RC-505 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0185), /* BOSS GP-10 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x0189), /* BOSS GT-100v2 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d6), /* BOSS GT-1 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d8), /* BOSS Katana */ @@ -86,6 +87,7 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { static const struct snd_usb_implicit_fb_match capture_implicit_fb_quirks[] = { IMPLICIT_FB_FIXED_DEV(0x0582, 0x0130, 0x0d, 0x01), /* BOSS BR-80 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0171, 0x0d, 0x01), /* BOSS RC-505 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0185, 0x0d, 0x01), /* BOSS GP-10 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0189, 0x0d, 0x01), /* BOSS GT-100v2 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d6, 0x0d, 0x01), /* BOSS GT-1 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d8, 0x0d, 0x01), /* BOSS Katana */ -- 2.26.2

1 0

[PATCH v3 2/3] x86/platform/dell-privacy-wmi: add document for dell privacy driver
by Perry Yuan 14 Feb '21

14 Feb '21

From: Perry Yuan <perry_yuan(a)dell.com> Describe the Dell Privacy feature capabilities and devices state class exposed by BIOS Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> --- .../testing/sysfs-platform-dell-privacy-wmi | 31 +++++++++++++++++++ 1 file changed, 31 insertions(+) create mode 100644 Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi diff --git a/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi b/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi new file mode 100644 index 000000000000..3dbc2d25b60e --- /dev/null +++ b/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi @@ -0,0 +1,31 @@ +What: /sys/bus/wmi/devices/6932965F-1671-4CEB-B988-D3AB0A901919/devices_supported +Date: Jan 2021 +KernelVersion: 5.11 +Contact: "perry_yuan(a)dell.com>" +Description: + Display which dell hardware level privacy devices are supported + “Dell Privacy” is a set of HW, FW, and SW features to enhance + Dell’s commitment to platform privacy for MIC, Camera, and + ePrivacy screens. + The supported privacy haredware privacy devices are: + * 0x0 - None, + * 0x1 - Microphone, + * 0x2 - Camera, + * 0x4 - ePrivacy Screen + +What: /sys/bus/wmi/devices/6932965F-1671-4CEB-B988-D3AB0A901919/current_state +Date: Jan 2021 +KernelVersion: 5.11 +Contact: "perry_yuan(a)dell.com>" +Description: + Allow user space to check current dell privacy device state. + Describes the Device State class exposed by BIOS which can be + consumed by various applications interested in knowing the Privacy + feature capabilities + There are three Bits for available states: + * 0 -> Off + * 1 -> On + * Bit0 -> Microphone + * Bit1 -> Camera + * Bit2 -> ePrivacyScreen + -- 2.25.1

3 3

mixer controls not updating when changed in another process for ASoC aux devices
by Matt Campbell 13 Feb '21

13 Feb '21

I am using the snd-soc-simple-card driver to glue together a DAI, Codec, and a few aux devices that have custom drivers for pre-amp control. When I have two instances of alsamixer open and change the output control (a control that is part of the main Codec driver) that change immediately shows in the second instance of alsamixer. However, when changing any of the controls that come from an aux device, these changes do not show up in the second instance. Exiting and re-launching the second alsamixer will then show the changes. I would expect these controls to behave the same as the main codec master output level with immediate update in all instances. I'm speculating that the aux devices might be treated differently in a way that is causing the unexpected behavior. I haven't been able to identify anything through a few hours of code inspection, but I'm far from an expert on the internals of ASoC. I'm hoping someone on this list might be able to confirm or deny that suspicion and possibly help me get to the bottom of this. A few other possibly relevant details: * This happens both with snd-soc-simple-driver and a different custom in-house platform driver that uses aux devs. * I am on an older version of the kernel (4.14.2). I scanned logs between the head of master and that version for ASoC changes and nothing popped out to me as relevant. * The aux dev causing this is an in-house driver for the THAT 5xx3 series of pre-amplifier controller that we intend to upstream. I'm happy to share code from that if it helps. Best, ~Matt -- Matthew Campbell Principal Engineer mcampbell(a)izotope.com iZotope, Inc. www.izotope.com

1 0

UCM profile for the Chromebook Yoga C630. Please add.
by GitHub issues - edited 12 Feb '21

12 Feb '21

alsa-project/alsa-ucm-conf issue #82 was edited from neil-vonholtum: Here is the HiFi.conf file contents: SectionVerb { Value { OutputDspName "speaker_eq" FullySpecifiedUCM "1" } EnableSequence [ cdev "hw:kblda7219max" cset "name='codec1_out mo media0_in mi Switch' off" cset "name='codec0_out mo media0_in mi Switch' off" cset "name='Playback Digital Volume' 111" cset "name='Playback Digital Switch' 1" cset "name='Out DACL Mux' DAIL" cset "name='Out DACR Mux' DAIR" cset "name='Mixer Out FilterL DACL Switch' 1" cset "name='Mixer Out FilterR DACR Switch' 1" cset "name='ST Mixer Out FilterL Out FilterL Switch' 1" cset "name='ST Mixer Out FilterR Out FilterR Switch' 1" cset "name='Headphone Jack Switch' off" cset "name='Headset Mic Switch' off" cset "name='Mic Volume' 5" cset "name='Capture Digital Volume' 111" cset "name='Capture Digital Switch' 1" cset "name='Mixin Volume' 10" cset "name='Mixin Switch' 1" cset "name='Mixer In Mic Switch' 1" cset "name='Out DAIL Mux' ADC" cset "name='Out DAIR Mux' ADC" cset "name='media0_out mo codec0_in mi Switch' off" cset "name='media0_out mo dmic01_hifi_in mi Switch' off" cset "name='Pin5-Port0 Mux' 1" cset "name='Pin6-Port0 Mux' 2" cset "name='Pin7-Port0 Mux' 3" cset "name='Gain Ramp Rate' 1" ] DisableSequence [ ] } SectionDevice."Speaker".0 { Value { PlaybackPCM "hw:kblda7219max,0" } EnableSequence [ cdev "hw:kblda7219max" cset "name='codec0_out mo media0_in mi Switch' on" cset "name='Spk Switch' on" ] DisableSequence [ cdev "hw:kblda7219max" cset "name='codec0_out mo media0_in mi Switch' off" cset "name='Spk Switch' off" ] } SectionDevice."Headphone".0 { Value { PlaybackPCM "hw:kblda7219max,0" MixerName "Headphone" JackName "kblda7219max Headset Jack" JackType "gpio" JackSwitch "2" OutputDspName "" } EnableSequence [ cdev "hw:kblda7219max" cset "name='codec1_out mo media0_in mi Switch' on" cset "name='Headphone Jack Switch' on" cset "name='Headphone Switch' 1" ] DisableSequence [ cdev "hw:kblda7219max" cset "name='codec1_out mo media0_in mi Switch' off" cset "name='Headphone Jack Switch' off" cset "name='Headphone Switch' 0" ] } SectionDevice."Line Out".0 { Value { PlaybackPCM "hw:kblda7219max,0" MixerName "Headphone" JackName "kblda7219max Headset Jack" JackType "gpio" JackSwitch "6" OutputDspName "" } EnableSequence [ cdev "hw:kblda7219max" cset "name='codec1_out mo media0_in mi Switch' on" cset "name='Headphone Jack Switch' on" cset "name='Headphone Switch' 1" ] DisableSequence [ cdev "hw:kblda7219max" cset "name='codec1_out mo media0_in mi Switch' off" cset "name='Headphone Jack Switch' off" cset "name='Headphone Switch' 0" ] } SectionDevice."Internal Mic".0 { Value { CapturePCM "hw:kblda7219max,3" CaptureChannelMap "0 1 -1 -1 -1 -1 -1 -1 -1 -1 -1" MaxSoftwareGain "2000" } EnableSequence [ cdev "hw:kblda7219max" cset "name='media0_out mo dmic01_hifi_in mi Switch' on" ] DisableSequence [ cdev "hw:kblda7219max" cset "name='media0_out mo dmic01_hifi_in mi Switch' off" ] } SectionDevice."Mic".0 { Value { CapturePCM "hw:kblda7219max,1" MixerName "Headset Mic" JackName "kblda7219max Headset Jack" JackType "gpio" } EnableSequence [ cdev "hw:kblda7219max" cset "name='Headset Mic Switch' on" cset "name='media0_out mo codec0_in mi Switch' on" cset "name='Mic Switch' on" ] DisableSequence [ cdev "hw:kblda7219max" cset "name='Headset Mic Switch' off" cset "name='media0_out mo codec0_in mi Switch' off" cset "name='Mic Switch' off" ] } SectionDevice."HDMI1".0 { Value { PlaybackPCM "hw:kblda7219max,4" JackName "kblda7219max HDMI/DP, pcm=4 Jack" JackType "gpio" OutputDspName "" } EnableSequence [ ] DisableSequence [ ] } SectionDevice."HDMI2".0 { Value { PlaybackPCM "hw:kblda7219max,5" JackName "kblda7219max HDMI/DP, pcm=5 Jack" JackType "gpio" OutputDspName "" } EnableSequence [ ] DisableSequence [ ] } SectionDevice."HDMI3".0 { Value { PlaybackPCM "hw:kblda7219max,6" JackName "kblda7219max HDMI/DP, pcm=6 Jack" JackType "gpio" OutputDspName "" } EnableSequence [ ] DisableSequence [ ] } Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/82 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Bass speaker is off on yoga slim 7i 15 (Yoga Slim 7-15ITL05)
by GitHub issues - edited 12 Feb '21

12 Feb '21

alsa-project/alsa-ucm-conf issue #83 was edited from juliusvonkohout: Alsainfo: http://alsa-project.org/db/?f=673e04313fa931831176ed3f50a9af1c9206395e I tested with fedora 33, 34 and Ubuntu 21.04. Only the high frequency speakers at the laptop bottom are working, while the speakers above the keyboard are off. Here is a link to the laptop https://www.lenovo.com/de/de/laptops/yoga/yoga-slim-series/Yoga-Slim-7-15II… In Microsoft Windows 10 everything is fine. In Ubuntu 21.04 i can at least change the bass speaker volume in alsamixer, but it is changing hardware master volume instead of hardware bass speaker volume. In Feodra 33 and 34 i can not even change the bass speaker volume in alsamixer ![audiobug](https://user-images.githubusercontent.com/45896133/107797084-d3090500-6d5a-11eb-86fb-229800b546b0.png) You can immiediatly hear, that the bass speaker above the keyboard is off [audio.zip](https://github.com/alsa-project/alsa-ucm-conf/files/5973232/audi… There are related bugs for ALC287 The same bug on other yoga laptops https://bugzilla.kernel.org/show_bug.cgi?id=205755 Patches for the issue on other yoga Laptops https://patchwork.kernel.org/project/alsa-devel/patch/20201205051130.8122-1… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/83 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Bass speaker is off on yoga slim 7i 15 (7-15ITL)
by GitHub issues - opened 12 Feb '21

12 Feb '21

alsa-project/alsa-ucm-conf issue #83 was opened from juliusvonkohout: Alsainfo: http://alsa-project.org/db/?f=673e04313fa931831176ed3f50a9af1c9206395e I tested with fedora 33, 34 and Ubuntu 21.04. Only the high frequency speakers at the laptop bottom are working, while the speakers above the keyboard are off. Here is a link to the laptop https://www.lenovo.com/de/de/laptops/yoga/yoga-slim-series/Yoga-Slim-7-15II… In Microsoft Windows 10 everything is fine. In Ubuntu 21.04 i can at least change the bass speaker volume in alsamixer, but it is changing hardware master volume instead of hardware bass speaker volume. In Feodra 33 and 34 i can not even change the bass speaker volume in alsamixer ![audiobug](https://user-images.githubusercontent.com/45896133/107797084-d3090500-6d5a-11eb-86fb-229800b546b0.png) You can immiediatly hear, that the bass speaker above the keyboard is off [audio.zip](https://github.com/alsa-project/alsa-ucm-conf/files/5973232/audi… There are related bugs for ALC287 The same bug on other yoga laptops https://bugzilla.kernel.org/show_bug.cgi?id=205755 Patches for the issue on other yoga Laptops https://patchwork.kernel.org/project/alsa-devel/patch/20201205051130.8122-1… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/83 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda: Add another CometLake-H PCI ID
by Kai Vehmanen 12 Feb '21

12 Feb '21

Add one more HD Audio PCI ID for CometLake-H PCH. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 801c5dd25707..7ba665e9272c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2475,6 +2475,8 @@ static const struct pci_device_id azx_ids[] = { /* CometLake-H */ { PCI_DEVICE(0x8086, 0x06C8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0xf1c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* CometLake-S */ { PCI_DEVICE(0x8086, 0xa3f0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, base-commit: 03cc2c42871ecf90017cace94bfd1ca5ecaf6e48 -- 2.29.2

3 2

[PATCH] ASoC: SOF: relax ABI checks and avoid unnecessary warnings
by Kai Vehmanen 12 Feb '21

12 Feb '21

With recent SOF 1.7 pre-releases, kernel has been emitting following warnings at probe: [10006.645216] sof-audio-pci 0000:00:1f.3: warn: FW ABI is more recent than kernel [10006.652137] sof-audio-pci 0000:00:1f.3: warn: topology ABI is more recent than kernel The warnings are emitted due to increase of the patch-level in firmware mainline (to 3.17.1). But the patch level should not be considered even in the strict ABI check, so modify the kernel side logic that makes the check and only consider the major.minor components. BugLink: https://github.com/thesofproject/linux/issues/2647 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/ipc.c | 2 +- sound/soc/sof/topology.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index 1bc3d6282f16..c2d07b783f60 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -798,7 +798,7 @@ int snd_sof_ipc_valid(struct snd_sof_dev *sdev) return -EINVAL; } - if (v->abi_version > SOF_ABI_VERSION) { + if (SOF_ABI_VERSION_MINOR(v->abi_version) > SOF_ABI_MINOR) { if (!IS_ENABLED(CONFIG_SND_SOC_SOF_STRICT_ABI_CHECKS)) { dev_warn(sdev->dev, "warn: FW ABI is more recent than kernel\n"); } else { diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index d6e1f33eb1e9..10f99620eb31 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -3658,7 +3658,7 @@ static int sof_manifest(struct snd_soc_component *scomp, int index, return -EINVAL; } - if (abi_version > SOF_ABI_VERSION) { + if (SOF_ABI_VERSION_MINOR(abi_version) > SOF_ABI_MINOR) { if (!IS_ENABLED(CONFIG_SND_SOC_SOF_STRICT_ABI_CHECKS)) { dev_warn(scomp->dev, "warn: topology ABI is more recent than kernel\n"); } else { base-commit: ec9d68508ff65df1dc24cf8100eb40ddd196c2fd -- 2.29.2

3 3

UCM profile for the Chromebook Yoga C630. Please add.
by GitHub issues - opened 12 Feb '21

12 Feb '21

alsa-project/alsa-ucm-conf issue #82 was opened from neil-vonholtum: SectionVerb { Value { OutputDspName "speaker_eq" FullySpecifiedUCM "1" } EnableSequence [ cdev "hw:kblda7219max" cset "name='codec1_out mo media0_in mi Switch' off" cset "name='codec0_out mo media0_in mi Switch' off" cset "name='Playback Digital Volume' 111" cset "name='Playback Digital Switch' 1" cset "name='Out DACL Mux' DAIL" cset "name='Out DACR Mux' DAIR" cset "name='Mixer Out FilterL DACL Switch' 1" cset "name='Mixer Out FilterR DACR Switch' 1" cset "name='ST Mixer Out FilterL Out FilterL Switch' 1" cset "name='ST Mixer Out FilterR Out FilterR Switch' 1" cset "name='Headphone Jack Switch' off" cset "name='Headset Mic Switch' off" cset "name='Mic Volume' 5" cset "name='Capture Digital Volume' 111" cset "name='Capture Digital Switch' 1" cset "name='Mixin Volume' 10" cset "name='Mixin Switch' 1" cset "name='Mixer In Mic Switch' 1" cset "name='Out DAIL Mux' ADC" cset "name='Out DAIR Mux' ADC" cset "name='media0_out mo codec0_in mi Switch' off" cset "name='media0_out mo dmic01_hifi_in mi Switch' off" cset "name='Pin5-Port0 Mux' 1" cset "name='Pin6-Port0 Mux' 2" cset "name='Pin7-Port0 Mux' 3" cset "name='Gain Ramp Rate' 1" ] DisableSequence [ ] } SectionDevice."Speaker".0 { Value { PlaybackPCM "hw:kblda7219max,0" } EnableSequence [ cdev "hw:kblda7219max" cset "name='codec0_out mo media0_in mi Switch' on" cset "name='Spk Switch' on" ] DisableSequence [ cdev "hw:kblda7219max" cset "name='codec0_out mo media0_in mi Switch' off" cset "name='Spk Switch' off" ] } SectionDevice."Headphone".0 { Value { PlaybackPCM "hw:kblda7219max,0" MixerName "Headphone" JackName "kblda7219max Headset Jack" JackType "gpio" JackSwitch "2" OutputDspName "" } EnableSequence [ cdev "hw:kblda7219max" cset "name='codec1_out mo media0_in mi Switch' on" cset "name='Headphone Jack Switch' on" cset "name='Headphone Switch' 1" ] DisableSequence [ cdev "hw:kblda7219max" cset "name='codec1_out mo media0_in mi Switch' off" cset "name='Headphone Jack Switch' off" cset "name='Headphone Switch' 0" ] } SectionDevice."Line Out".0 { Value { PlaybackPCM "hw:kblda7219max,0" MixerName "Headphone" JackName "kblda7219max Headset Jack" JackType "gpio" JackSwitch "6" OutputDspName "" } EnableSequence [ cdev "hw:kblda7219max" cset "name='codec1_out mo media0_in mi Switch' on" cset "name='Headphone Jack Switch' on" cset "name='Headphone Switch' 1" ] DisableSequence [ cdev "hw:kblda7219max" cset "name='codec1_out mo media0_in mi Switch' off" cset "name='Headphone Jack Switch' off" cset "name='Headphone Switch' 0" ] } SectionDevice."Internal Mic".0 { Value { CapturePCM "hw:kblda7219max,3" CaptureChannelMap "0 1 -1 -1 -1 -1 -1 -1 -1 -1 -1" MaxSoftwareGain "2000" } EnableSequence [ cdev "hw:kblda7219max" cset "name='media0_out mo dmic01_hifi_in mi Switch' on" ] DisableSequence [ cdev "hw:kblda7219max" cset "name='media0_out mo dmic01_hifi_in mi Switch' off" ] } SectionDevice."Mic".0 { Value { CapturePCM "hw:kblda7219max,1" MixerName "Headset Mic" JackName "kblda7219max Headset Jack" JackType "gpio" } EnableSequence [ cdev "hw:kblda7219max" cset "name='Headset Mic Switch' on" cset "name='media0_out mo codec0_in mi Switch' on" cset "name='Mic Switch' on" ] DisableSequence [ cdev "hw:kblda7219max" cset "name='Headset Mic Switch' off" cset "name='media0_out mo codec0_in mi Switch' off" cset "name='Mic Switch' off" ] } SectionDevice."HDMI1".0 { Value { PlaybackPCM "hw:kblda7219max,4" JackName "kblda7219max HDMI/DP, pcm=4 Jack" JackType "gpio" OutputDspName "" } EnableSequence [ ] DisableSequence [ ] } SectionDevice."HDMI2".0 { Value { PlaybackPCM "hw:kblda7219max,5" JackName "kblda7219max HDMI/DP, pcm=5 Jack" JackType "gpio" OutputDspName "" } EnableSequence [ ] DisableSequence [ ] } SectionDevice."HDMI3".0 { Value { PlaybackPCM "hw:kblda7219max,6" JackName "kblda7219max HDMI/DP, pcm=6 Jack" JackType "gpio" OutputDspName "" } EnableSequence [ ] DisableSequence [ ] } Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/82 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2/conf.d: add symlinks for Qualcomm cards
by GitHub pull_request - opened 12 Feb '21

12 Feb '21

alsa-project/alsa-ucm-conf pull request #81 was opened from lumag: Add ucm2/conf.d symlinks for all defined Qualcomm sound cards. Signed-off-by: Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/81 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/81.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH alsa-ucm-conf] codecs/rt5672: Remove Intel SST specific HeadsetMic workaround
by Hans de Goede 11 Feb '21

11 Feb '21

There used to be an issue with the HeadsetMic on BYT devices with a rt5672 codec only being recorded on the left channel, while the right channel recorded silence. The ucm2/codecs/rt5672/HeadsetMic.conf contains a workaround for this which changes the SST record-stream demuxer settings to use "slot 0" for both the left and right channels when recording from the HeadsetMic. There are 2 problems with this workaround: 1. It uses SST specific mixer settings causing things to break when using the SOF driver. 2. Files under uc2m/codecs should be architecture agnostic and should not depend on platform dependent mixer settings such as the SST driver mixer settings. The problem which the workaround used to work around has since been fixed in the kernel by switching the wire format from TDM/DSP A to I2S: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… This fix has landed in 5.8 and has been backported to all recent stable kernel releases. Since this is fixed in the kernel now, we can drop the workaround, fixing the 2 problems mentioned above. This was tested on a ThinkPad 8 tablet, which is the same device as where the HeadsetMic problem was originally noticed. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/codecs/rt5672/HeadsetMic.conf | 7 ------- 1 file changed, 7 deletions(-) diff --git a/ucm2/codecs/rt5672/HeadsetMic.conf b/ucm2/codecs/rt5672/HeadsetMic.conf index a764d82..fc061b0 100644 --- a/ucm2/codecs/rt5672/HeadsetMic.conf +++ b/ucm2/codecs/rt5672/HeadsetMic.conf @@ -12,11 +12,6 @@ SectionDevice."Headset" { cset "name='RECMIXR BST1 Switch' on" cset "name='Sto1 ADC MIXL ADC1 Switch' on" cset "name='Sto1 ADC MIXR ADC1 Switch' on" - # For unknown reasons the headset mic sound is only recorded - # on the left channel, we tweak the data-stream reception to - # use the left samples for both slots, so that we get a stereo - # stream with the mic sound on both channels - cset "name='codec_in rx deinterleaver codec_in0_1' 1" ] DisableSequence [ @@ -25,8 +20,6 @@ SectionDevice."Headset" { cset "name='RECMIXR BST1 Switch' off" cset "name='Sto1 ADC MIXL ADC1 Switch' off" cset "name='Sto1 ADC MIXR ADC1 Switch' off" - # Undo data-stream reception mono-mix workaround - cset "name='codec_in rx deinterleaver codec_in0_1' 2" ] Value { -- 2.30.0

2 1

[PATCH alsa-ucm-conf] bytcr-wm5102: Add new UCM profile for BYT boards with a WM5102 codec
by Hans de Goede 11 Feb '21

11 Feb '21

Add new UCM profile for BYT boards with a WM5102 codec. This has been tested with both the SST and the SOF driver on a Lenovo Yoga Tablet 2 1051L. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/SOF/HiFi.conf | 9 +++++++++ ucm2/bytcr-wm5102/HiFi.conf | 27 +++++++++++++++++++++++++ ucm2/bytcr-wm5102/bytcr-wm5102.conf | 6 ++++++ ucm2/codecs/wm5102/EnableSeq.conf | 16 +++++++++++++++ ucm2/codecs/wm5102/HeadPhones.conf | 23 +++++++++++++++++++++ ucm2/codecs/wm5102/IN1-HeadsetMic.conf | 24 ++++++++++++++++++++++ ucm2/codecs/wm5102/IN3-InternalMic.conf | 23 +++++++++++++++++++++ ucm2/codecs/wm5102/Speaker.conf | 22 ++++++++++++++++++++ 8 files changed, 150 insertions(+) create mode 100644 ucm2/bytcr-wm5102/HiFi.conf create mode 100644 ucm2/bytcr-wm5102/bytcr-wm5102.conf create mode 100644 ucm2/codecs/wm5102/EnableSeq.conf create mode 100644 ucm2/codecs/wm5102/HeadPhones.conf create mode 100644 ucm2/codecs/wm5102/IN1-HeadsetMic.conf create mode 100644 ucm2/codecs/wm5102/IN3-InternalMic.conf create mode 100644 ucm2/codecs/wm5102/Speaker.conf diff --git a/ucm2/SOF/HiFi.conf b/ucm2/SOF/HiFi.conf index af0d8c9..adf8cc5 100644 --- a/ucm2/SOF/HiFi.conf +++ b/ucm2/SOF/HiFi.conf @@ -79,6 +79,15 @@ If.bytcht_max98090 { True.Include.main.File "/chtmax98090/HiFi.conf" } +If.bytcht_wm5102 { + Condition { + Type String + Haystack "${CardName}" + Needle "bytcht wm5102" + } + True.Include.main.File "/bytcr-wm5102/HiFi.conf" +} + If.bdw_rt286 { Condition { Type String diff --git a/ucm2/bytcr-wm5102/HiFi.conf b/ucm2/bytcr-wm5102/HiFi.conf new file mode 100644 index 0000000..1269868 --- /dev/null +++ b/ucm2/bytcr-wm5102/HiFi.conf @@ -0,0 +1,27 @@ +SectionVerb { + + Value { + TQ "HiFi" + } + + Include.e.File "/codecs/wm5102/EnableSeq.conf" + + If.Controls { + Condition { + Type ControlExists + Control "name='media0_in Gain 0 Switch'" + } + True { + Include.pe { + File "/platforms/bytcr/PlatformEnableSeq.conf" + Before.EnableSequence "0" + } + Include.pd.File "/platforms/bytcr/PlatformDisableSeq.conf" + } + } +} + +Include.spk.File "/codecs/wm5102/Speaker.conf" +Include.hp.File "/codecs/wm5102/HeadPhones.conf" +Include.mic.File "/codecs/wm5102/IN3-InternalMic.conf" +Include.headset.File "/codecs/wm5102/IN1-HeadsetMic.conf" diff --git a/ucm2/bytcr-wm5102/bytcr-wm5102.conf b/ucm2/bytcr-wm5102/bytcr-wm5102.conf new file mode 100644 index 0000000..af966ef --- /dev/null +++ b/ucm2/bytcr-wm5102/bytcr-wm5102.conf @@ -0,0 +1,6 @@ +Syntax 3 + +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Play HiFi quality Music" +} diff --git a/ucm2/codecs/wm5102/EnableSeq.conf b/ucm2/codecs/wm5102/EnableSeq.conf new file mode 100644 index 0000000..6a857a2 --- /dev/null +++ b/ucm2/codecs/wm5102/EnableSeq.conf @@ -0,0 +1,16 @@ +EnableSequence [ + # Route AIF1 to the speakers and headphones + cset "name='SPKOUTL Input 1' AIF1RX1" + cset "name='SPKOUTR Input 1' AIF1RX2" + cset "name='HPOUT1L Input 1' AIF1RX1" + cset "name='HPOUT1R Input 1' AIF1RX2" + + # Both mics are quite soft by default, boost then + cset "name='IN1L Volume' 28" + cset "name='IN3L Volume' 28" + + cset "name='Headphone Switch' off" + cset "name='Headset Mic Switch' off" + cset "name='Internal Mic Switch' off" + cset "name='Speaker Switch' off" +] diff --git a/ucm2/codecs/wm5102/HeadPhones.conf b/ucm2/codecs/wm5102/HeadPhones.conf new file mode 100644 index 0000000..cb29347 --- /dev/null +++ b/ucm2/codecs/wm5102/HeadPhones.conf @@ -0,0 +1,23 @@ +SectionDevice."Headphones" { + Comment "Headphones" + + Value { + PlaybackPriority 200 + PlaybackPCM "hw:${CardId}" + JackControl "Headphone Jack" + } + + ConflictingDevice [ + "Speaker" + ] + + EnableSequence [ + cset "name='HPOUT1 Digital Switch' on" + cset "name='Headphone Switch' on" + ] + + DisableSequence [ + cset "name='HPOUT1 Digital Switch' off" + cset "name='Headphone Switch' off" + ] +} diff --git a/ucm2/codecs/wm5102/IN1-HeadsetMic.conf b/ucm2/codecs/wm5102/IN1-HeadsetMic.conf new file mode 100644 index 0000000..a607487 --- /dev/null +++ b/ucm2/codecs/wm5102/IN1-HeadsetMic.conf @@ -0,0 +1,24 @@ +SectionDevice."Headset" { + Comment "Headset Microphone" + + ConflictingDevice [ + "Mic" + ] + + EnableSequence [ + cset "name='AIF1TX1 Input 1' IN1L" + cset "name='AIF1TX2 Input 1' IN1L" + + cset "name='Headset Mic Switch' on" + ] + + DisableSequence [ + cset "name='Headset Mic Switch' off" + ] + + Value { + CapturePriority 200 + CapturePCM "hw:${CardId}" + JackControl "Headset Mic Jack" + } +} diff --git a/ucm2/codecs/wm5102/IN3-InternalMic.conf b/ucm2/codecs/wm5102/IN3-InternalMic.conf new file mode 100644 index 0000000..3a03ac3 --- /dev/null +++ b/ucm2/codecs/wm5102/IN3-InternalMic.conf @@ -0,0 +1,23 @@ +SectionDevice."Mic" { + Comment "Internal Microphone" + + ConflictingDevice [ + "Headset" + ] + + EnableSequence [ + cset "name='AIF1TX1 Input 1' IN3L" + cset "name='AIF1TX2 Input 1' IN3L" + + cset "name='Internal Mic Switch' on" + ] + + DisableSequence [ + cset "name='Internal Mic Switch' off" + ] + + Value { + CapturePriority 100 + CapturePCM "hw:${CardId}" + } +} diff --git a/ucm2/codecs/wm5102/Speaker.conf b/ucm2/codecs/wm5102/Speaker.conf new file mode 100644 index 0000000..9f631f5 --- /dev/null +++ b/ucm2/codecs/wm5102/Speaker.conf @@ -0,0 +1,22 @@ +SectionDevice."Speaker" { + Comment "Speakers" + + Value { + PlaybackPriority 100 + PlaybackPCM "hw:${CardId}" + } + + ConflictingDevice [ + "Headphones" + ] + + EnableSequence [ + cset "name='Speaker Digital Switch' on" + cset "name='Speaker Switch' on" + ] + + DisableSequence [ + cset "name='Speaker Digital Switch' off" + cset "name='Speaker Switch' off" + ] +} -- 2.30.0

2 3

[PATCH] ASoC: wm_adsp: Remove unused control callback structure
by Charles Keepax 11 Feb '21

11 Feb '21

This callback structure has never been used and it is not clear why it was added in the first place. Remove it to clear up the code a little. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 262c2db4f81ce..070ca7d8c6610 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -595,13 +595,6 @@ static const struct { [WM_ADSP_FW_MISC] = { .file = "misc" }, }; -struct wm_coeff_ctl_ops { - int (*xget)(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); - int (*xput)(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); -}; - struct wm_coeff_ctl { const char *name; const char *fw_name; @@ -609,7 +602,6 @@ struct wm_coeff_ctl { const char *subname; unsigned int subname_len; struct wm_adsp_alg_region alg_region; - struct wm_coeff_ctl_ops ops; struct wm_adsp *dsp; unsigned int enabled:1; struct list_head list; @@ -1497,8 +1489,6 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, } ctl->enabled = 1; ctl->set = 0; - ctl->ops.xget = wm_coeff_get; - ctl->ops.xput = wm_coeff_put; ctl->dsp = dsp; ctl->flags = flags; -- 2.11.0

2 1

[PATCH v5 0/7] ASoC: codecs: add support for LPASS Codec TX and RX macros
by Srinivas Kandagatla 11 Feb '21

11 Feb '21

This patchset adds support for two Codec Macro blocks(TX and RX) available in Qualcomm LPASS (Low Power Audio SubSystem). There are WSA, VA, TX and RX Macros on LPASS IP, each of the Macro block has specific connectivity like WSA Macros are intended to connect to WSA Smart speaker codecs via SoundWire. VA Macro is intended for DMICs, and TX/RX for Analog codecs via SoundWire like other WCD938x Codecs to provide headphone/ear/lineout/amic/dmic etc .. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. This patchset has been tested on support to SM8250 MTP Development Board. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively and WCD938x TX and RX connected via Soundwire via TX and RX Macros reseptively. Thanks, srini Changes since v4: - One more ktest fix for unused-but-set-variable warning and a fix for W=2 warning and removed unused macros - added bindings ack from Rob Srinivas Kandagatla (7): ASoC: qcom: dt-bindings: add bindings for lpass rx macro codec ASoC: codecs: lpass-rx-macro: add support for lpass rx macro ASoC: codecs: lpass-rx-macro: add dapm widgets and route ASoC: codecs: lpass-rx-macro: add iir widgets ASoC: qcom: dt-bindings: add bindings for lpass tx macro codec ASoC: codecs: lpass-tx-macro: add support for lpass tx macro ASoC: codecs: lpass-tx-macro: add dapm widgets and route .../bindings/sound/qcom,lpass-rx-macro.yaml | 62 + .../bindings/sound/qcom,lpass-tx-macro.yaml | 67 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/lpass-rx-macro.c | 3599 +++++++++++++++++ sound/soc/codecs/lpass-tx-macro.c | 1862 +++++++++ 6 files changed, 5604 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml create mode 100644 sound/soc/codecs/lpass-rx-macro.c create mode 100644 sound/soc/codecs/lpass-tx-macro.c -- 2.21.0

2 8

[PATCH] ALSA: hda/hdmi: Drop bogus check at closing a stream
by Takashi Iwai 11 Feb '21

11 Feb '21

Some users reported the kernel WARNING with stack traces from hdmi_pcm_close(), and it's the line checking the per_cvt->assigned flag. This used to be a valid check in the past because the flag was turned on/off only at opening and closing a PCM stream. Meanwhile, since the introduction of the silent-stream mode, this flag may be turned on/off at the monitor connection/disconnection time, which isn't always associated with the PCM open/close. Hence this may lead to the inconsistent per_cvt->assigned flag at closing. As the check itself became almost useless and confuses users as if it were a serious problem, just drop the check. Fixes: b1a5039759cb ("ALSA: hda/hdmi: fix silent stream for first playback to DP") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210987 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 97adff0cbcab..e405be7929e3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2130,7 +2130,6 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, goto unlock; } per_cvt = get_cvt(spec, cvt_idx); - snd_BUG_ON(!per_cvt->assigned); per_cvt->assigned = 0; hinfo->nid = 0; -- 2.26.2

2 1

[PATCH v4 resend 00/13] MFD/extcon/ASoC: Rework arizona codec jack-detect support
by Hans de Goede 11 Feb '21

11 Feb '21

Hi all, Here is v4 of my series to rework the arizona codec jack-detect support to use the snd_soc_jack helpers instead of direct extcon reporting. This is a resend with some extra *-by tags collected and with the extcon folks added to the "To:" list, which I somehow missed with the original v4 posting, sorry. This is done by reworking the extcon driver into an arizona-jackdet library and then modifying the codec drivers to use that directly, replacing the old separate extcon child-devices and extcon-driver. This brings the arizona-codec jack-detect handling inline with how all other ASoC codec driver do this. This was developed and tested on a Lenovo Yoga Tablet 1051L with a WM5102 codec. This was also tested by Charles Keepax, one of the Cirrus Codec folks. This depends on the previously posted "[PATCH v4 0/5] MFD/ASoC: Add support for Intel Bay Trail boards with WM5102 codec" series and there are various interdependencies between the patches in this series. Lee Jones, the MFD maintainer has agreed to take this series upstream through the MFD tree and to provide an immutable branch for the ASoC and extcon subsystems to merge. Mark and extcon-maintainers may we have your ack for merging these through the MFD tree ? Regards, Hans Hans de Goede (13): mfd: arizona: Drop arizona-extcon cells extcon: arizona: Fix some issues when HPDET IRQ fires after the jack has been unplugged extcon: arizona: Fix various races on driver unbind extcon: arizona: Fix flags parameter to the gpiod_get("wlf,micd-pol") call extcon: arizona: Always use pm_runtime_get_sync() when we need the device to be awake ASoC/extcon: arizona: Move arizona jack code to sound/soc/codecs/arizona-jack.c ASoC: arizona-jack: Move jack-detect variables to struct arizona_priv ASoC: arizona-jack: Use arizona->dev for runtime-pm ASoC: arizona-jack: convert into a helper library for codec drivers ASoC: arizona-jack: Use snd_soc_jack to report jack events ASoC: arizona-jack: Cleanup logging ASoC: arizona: Make the wm5102, wm5110, wm8997 and wm8998 drivers use the new jack library ASoC: Intel: bytcr_wm5102: Add jack detect support MAINTAINERS | 3 +- drivers/extcon/Kconfig | 8 - drivers/extcon/Makefile | 1 - drivers/mfd/arizona-core.c | 20 - sound/soc/codecs/Makefile | 2 +- .../soc/codecs/arizona-jack.c | 577 +++++++----------- sound/soc/codecs/arizona.h | 44 ++ sound/soc/codecs/wm5102.c | 12 +- sound/soc/codecs/wm5110.c | 12 +- sound/soc/codecs/wm8997.c | 14 +- sound/soc/codecs/wm8998.c | 9 + sound/soc/intel/boards/bytcr_wm5102.c | 28 +- 12 files changed, 325 insertions(+), 405 deletions(-) rename drivers/extcon/extcon-arizona.c => sound/soc/codecs/arizona-jack.c (76%) -- 2.29.2

4 24

[RESEND PATCH v2 0/9] soundwire/regmap: use _no_pm routines
by Bard Liao 11 Feb '21

11 Feb '21

When a Slave device is resumed, it may resume the bus and restart the enumeration. And Slave drivers will wait for initialization_complete complete in their resume function, however initialization_complete will complete after sdw_update_slave_status function is finished and codec driver usually call some IO functions in the update_status callback function. It will become a deadlock if we use regular read/write routines during the resuming process. This series touches both soundwire and regmap trees. commit fb5103f9d6ce ("regmap/SoundWire: sdw: add support for SoundWire 1.2 MBQ") is needed for soundwire tree to complie. On the other hands, commit 6e06a85556f9 ("soundwire: bus: add comments to explain interrupt loop filter") to commit 47b8520997a8 ("soundwire: bus: only clear valid DPN interrupts") are needed for regmap tree. v2: - Separate commits according to maintainer's comments. Bard Liao (4): soundwire: export sdw_write/read_no_pm functions regmap: sdw: use _no_pm functions in regmap_read/write regmap: sdw: use no_pm routines for SoundWire 1.2 MBQ regmap: sdw-mbq: use MODULE_LICENSE("GPL") Pierre-Louis Bossart (5): soundwire: bus: use sdw_update_no_pm when initializing a device soundwire: bus: use sdw_write_no_pm when setting the bus scale registers soundwire: bus: use no_pm IO routines for all interrupt handling soundwire: bus: fix confusion on device used by pm_runtime soundwire: bus: clarify dev_err/dbg device references drivers/base/regmap/regmap-sdw-mbq.c | 10 +- drivers/base/regmap/regmap-sdw.c | 4 +- drivers/soundwire/bus.c | 136 +++++++++++++++------------ include/linux/soundwire/sdw.h | 2 + 4 files changed, 85 insertions(+), 67 deletions(-) -- 2.17.1

3 14

Crash in acpi_ns_validate_handle triggered by soundwire on Linux 5.10
by Marcin Ślusarz 11 Feb '21

11 Feb '21

Hey, Linux 5.10 fails to boot on my IceLake laptop, where 5.9 worked fine. I'm not sure whether it's a bug in ACPI or Soundwire subsystem, so I'm sending this to both lists. The W taint flag is because of an unrelated nouveau failure (It was busted on 5.9, and is still on 5.10). Full kernel log down below. It's from a distro kernel, but I can build my own kernel with patches if needed. snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040380 BUG: kernel NULL pointer dereference, address: 0000000000000050 #PF: supervisor read access in kernel mode #PF: error_code(0x0000) - not-present page PGD 0 P4D 0 Oops: 0000 [#1] SMP NOPTI CPU: 6 PID: 472 Comm: systemd-udevd Tainted: G W 5.10.0-1-amd64 #1 Debian 5.10.4-1 Hardware name: HP HP Pavilion Laptop 15-cs3xxx/86E2, BIOS F.05 01/01/2020 RIP: 0010:acpi_ns_validate_handle+0x1a/0x23 Code: 00 48 83 c4 10 5b 5d 41 5c 41 5d 41 5e 41 5f c3 0f 1f 44 00 00 48 8d 57 ff 48 89 f8 48 83 fa fd 76 08 48 8b 05 0c b8 67 01 c3 <80> 7f 08 0f 74 02 31 c0 c3 0f 1f 44 00 00 48 8b 3d f6 b7 67 01 e8 RSP: 0000:ffffc388807c7b20 EFLAGS: 00010213 RAX: 0000000000000048 RBX: ffffc388807c7b70 RCX: 0000000000000000 RDX: 0000000000000047 RSI: 0000000000000246 RDI: 0000000000000048 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: ffffffffc0f5f4d1 R11: ffffffff8f0cb268 R12: 0000000000001001 R13: ffffffff8e33b160 R14: 0000000000000048 R15: 0000000000000000 FS: 00007f24548288c0(0000) GS:ffff9f781fb80000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000000000050 CR3: 0000000106158004 CR4: 0000000000770ee0 PKRU: 55555554 Call Trace: acpi_get_data_full+0x4d/0x92 acpi_bus_get_device+0x1f/0x40 sdw_intel_acpi_scan+0x59/0x230 [soundwire_intel] ? strstr+0x22/0x60 ? dmi_matches+0x76/0xe0 snd_intel_dsp_driver_probe.cold+0xaf/0x163 [snd_intel_dspcfg] azx_probe+0x7a/0x970 [snd_hda_intel] local_pci_probe+0x42/0x80 ? _cond_resched+0x16/0x40 pci_device_probe+0xfd/0x1b0 really_probe+0x205/0x460 driver_probe_device+0xe1/0x150 device_driver_attach+0xa1/0xb0 __driver_attach+0x8a/0x150 ? device_driver_attach+0xb0/0xb0 ? device_driver_attach+0xb0/0xb0 bus_for_each_dev+0x78/0xc0 bus_add_driver+0x12b/0x1e0 driver_register+0x8b/0xe0 ? 0xffffffffc0f65000 do_one_initcall+0x44/0x1d0 ? do_init_module+0x23/0x250 ? kmem_cache_alloc_trace+0xf5/0x200 do_init_module+0x5c/0x250 __do_sys_finit_module+0xb1/0x110 do_syscall_64+0x33/0x80 entry_SYSCALL_64_after_hwframe+0x44/0xa9 RIP: 0033:0x7f2454ce19b9 Code: 00 c3 66 2e 0f 1f 84 00 00 00 00 00 0f 1f 44 00 00 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 8b 0d a7 54 0c 00 f7 d8 64 89 01 48 RSP: 002b:00007fff0b66be58 EFLAGS: 00000246 ORIG_RAX: 0000000000000139 RAX: ffffffffffffffda RBX: 0000557d20c7a8b0 RCX: 00007f2454ce19b9 RDX: 0000000000000000 RSI: 00007f2454e6ce2d RDI: 000000000000001c RBP: 0000000000020000 R08: 0000000000000000 R09: 0000557d20c80d50 R10: 000000000000001c R11: 0000000000000246 R12: 00007f2454e6ce2d R13: 0000000000000000 R14: 0000557d20ae9c00 R15: 0000557d20c7a8b0 Modules linked in: snd_hda_intel(+) snd_intel_dspcfg soundwire_intel soundwire_generic_allocation kvm snd_soc_core snd_compress pcc_cpufreq(-) irqbypass soundwire_cadence acpi_cpufreq(-) rapl snd_hda_codec jitterentropy_rng uvcvideo intel_cstate cfg80211 snd_hda_core videobuf2_vmalloc videobuf2_memops videobuf2_v4l2 snd_hwdep videobuf2_common soundwire_bus intel_uncore videodev snd_pcm drbg snd_timer iTCO_wdt snd ansi_cprng joydev binfmt_misc hp_wmi(+) intel_pmc_bxt sparse_keymap mc pcspkr ecdh_generic serio_raw wmi_bmof intel_wmi_thunderbolt iTCO_vendor_support ecc soundcore watchdog rfkill libarc4 processor_thermal_device mei_me intel_rapl_common mei intel_soc_dts_iosf int3403_thermal int340x_thermal_zone hp_accel lis3lv02d hp_wireless int3400_thermal evdev acpi_thermal_rel intel_pmc_core acpi_pad ac acpi_tad parport_pc ppdev lp parport fuse configfs ip_tables x_tables autofs4 ext4 crc16 mbcache jbd2 crc32c_generic dm_crypt dm_mod nouveau i915 crc32_pclmul crc32c_intel ghash_clmulni_intel nvme aesni_intel libaes crypto_simd mxm_wmi ttm ahci nvme_core xhci_pci i2c_algo_bit libahci xhci_hcd drm_kms_helper t10_pi crc_t10dif libata psmouse cryptd glue_helper cec usbcore r8169 drm crct10dif_generic scsi_mod realtek mdio_devres i2c_i801 libphy i2c_smbus crct10dif_pclmul intel_lpss_pci crct10dif_common intel_lpss usb_common idma64 battery video wmi button CR2: 0000000000000050 ---[ end trace cf1844e25a37ee58 ]--- RIP: 0010:acpi_ns_validate_handle+0x1a/0x23 Code: 00 48 83 c4 10 5b 5d 41 5c 41 5d 41 5e 41 5f c3 0f 1f 44 00 00 48 8d 57 ff 48 89 f8 48 83 fa fd 76 08 48 8b 05 0c b8 67 01 c3 <80> 7f 08 0f 74 02 31 c0 c3 0f 1f 44 00 00 48 8b 3d f6 b7 67 01 e8 RSP: 0000:ffffc388807c7b20 EFLAGS: 00010213 RAX: 0000000000000048 RBX: ffffc388807c7b70 RCX: 0000000000000000 RDX: 0000000000000047 RSI: 0000000000000246 RDI: 0000000000000048 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: ffffffffc0f5f4d1 R11: ffffffff8f0cb268 R12: 0000000000001001 R13: ffffffff8e33b160 R14: 0000000000000048 R15: 0000000000000000 FS: 00007f24548288c0(0000) GS:ffff9f781fb80000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000000000050 CR3: 0000000106158004 CR4: 0000000000770ee0 PKRU: 55555554 Full kernel log: ----- microcode: microcode updated early to revision 0xa0, date = 2020-07-30 Linux version 5.10.0-1-amd64 (debian-kernel(a)lists.debian.org) (gcc-10 (Debian 10.2.1-3) 10.2.1 20201224, GNU ld (GNU Binutils for Debian) 2.35.1) #1 SMP Debian 5.10.4-1 (2020-12-31) Command line: BOOT_IMAGE=/vmlinuz-5.10.0-1-amd64 root=/dev/mapper/debian--vg-root ro quiet x86/split lock detection: warning about user-space split_locks x86/fpu: Supporting XSAVE feature 0x001: 'x87 floating point registers' x86/fpu: Supporting XSAVE feature 0x002: 'SSE registers' x86/fpu: Supporting XSAVE feature 0x004: 'AVX registers' x86/fpu: Supporting XSAVE feature 0x020: 'AVX-512 opmask' x86/fpu: Supporting XSAVE feature 0x040: 'AVX-512 Hi256' x86/fpu: Supporting XSAVE feature 0x080: 'AVX-512 ZMM_Hi256' x86/fpu: Supporting XSAVE feature 0x200: 'Protection Keys User registers' x86/fpu: xstate_offset[2]: 576, xstate_sizes[2]: 256 x86/fpu: xstate_offset[5]: 832, xstate_sizes[5]: 64 x86/fpu: xstate_offset[6]: 896, xstate_sizes[6]: 512 x86/fpu: xstate_offset[7]: 1408, xstate_sizes[7]: 1024 x86/fpu: xstate_offset[9]: 2432, xstate_sizes[9]: 8 x86/fpu: Enabled xstate features 0x2e7, context size is 2440 bytes, using 'compacted' format. BIOS-provided physical RAM map: BIOS-e820: [mem 0x0000000000000000-0x000000000009f7ff] usable BIOS-e820: [mem 0x000000000009f800-0x000000000009ffff] reserved BIOS-e820: [mem 0x00000000000e0000-0x00000000000fffff] reserved BIOS-e820: [mem 0x0000000000100000-0x0000000023902fff] usable BIOS-e820: [mem 0x0000000023903000-0x0000000023903fff] ACPI NVS BIOS-e820: [mem 0x0000000023904000-0x0000000023959fff] usable BIOS-e820: [mem 0x000000002395a000-0x000000002395afff] reserved BIOS-e820: [mem 0x000000002395b000-0x0000000032a2efff] usable BIOS-e820: [mem 0x0000000032a2f000-0x000000003518efff] reserved BIOS-e820: [mem 0x000000003518f000-0x0000000035f7efff] ACPI NVS BIOS-e820: [mem 0x0000000035f7f000-0x0000000035ffefff] ACPI data BIOS-e820: [mem 0x0000000035fff000-0x0000000035ffffff] usable BIOS-e820: [mem 0x0000000036000000-0x0000000039ffffff] reserved BIOS-e820: [mem 0x000000003a200000-0x000000003a3fffff] reserved BIOS-e820: [mem 0x000000003b000000-0x00000000cfffffff] reserved BIOS-e820: [mem 0x00000000fc800000-0x00000000fe7fffff] reserved BIOS-e820: [mem 0x00000000fec00000-0x00000000fec00fff] reserved BIOS-e820: [mem 0x00000000fed00000-0x00000000fed00fff] reserved BIOS-e820: [mem 0x00000000fed10000-0x00000000fed17fff] reserved BIOS-e820: [mem 0x00000000fed20000-0x00000000fed7ffff] reserved BIOS-e820: [mem 0x00000000feda0000-0x00000000feda1fff] reserved BIOS-e820: [mem 0x00000000fee00000-0x00000000fee00fff] reserved BIOS-e820: [mem 0x00000000ff400000-0x00000000ffffffff] reserved BIOS-e820: [mem 0x0000000100000000-0x00000008bfbfffff] usable NX (Execute Disable) protection: active SMBIOS 3.2 present. DMI: HP HP Pavilion Laptop 15-cs3xxx/86E2, BIOS F.05 01/01/2020 tsc: Detected 1500.000 MHz processor tsc: Detected 1497.600 MHz TSC e820: update [mem 0x00000000-0x00000fff] usable ==> reserved e820: remove [mem 0x000a0000-0x000fffff] usable last_pfn = 0x8bfc00 max_arch_pfn = 0x400000000 MTRR default type: write-back MTRR fixed ranges enabled: 00000-9FFFF write-back A0000-BFFFF uncachable C0000-FFFFF write-protect MTRR variable ranges enabled: 0 base 0080000000 mask 7F80000000 uncachable 1 base 0040000000 mask 7FC0000000 uncachable 2 base 003C000000 mask 7FFC000000 uncachable 3 base 003B000000 mask 7FFF000000 uncachable 4 base 2000000000 mask 6000000000 uncachable 5 base 1000000000 mask 7000000000 uncachable 6 base 4000000000 mask 4000000000 uncachable 7 disabled 8 disabled 9 disabled x86/PAT: Configuration [0-7]: WB WC UC- UC WB WP UC- WT last_pfn = 0x36000 max_arch_pfn = 0x400000000 found SMP MP-table at [mem 0x000fe1b0-0x000fe1bf] Using GB pages for direct mapping RAMDISK: [mem 0x2d18f000-0x31342fff] ACPI: Early table checksum verification disabled ACPI: RSDP 0x00000000000FE020 000024 (v02 HPQOEM) ACPI: XSDT 0x0000000035FCF188 0000FC (v01 HPQOEM SLIC-MPC 00000002 HP 01000013) ACPI: FACP 0x0000000035FD4000 00010C (v05 HPQOEM SLIC-MPC 00000002 HP 00040000) ACPI: DSDT 0x0000000035F9C000 032D1B (v02 HPQOEM 86E2 00000002 ACPI 00040000) ACPI: FACS 0x0000000035F19000 000040 ACPI: UEFI 0x0000000035F7E000 000236 (v01 HPQOEM 86E2 00000001 HP 00040000) ACPI: SSDT 0x0000000035FF9000 003DF8 (v02 HPQOEM 86E2 00001000 ACPI 00040000) ACPI: SSDT 0x0000000035FF7000 001B60 (v02 HPQOEM 86E2 00003000 ACPI 00040000) ACPI: SSDT 0x0000000035FF3000 003389 (v02 HPQOEM 86E2 00003000 ACPI 00040000) ACPI: SSDT 0x0000000035FE7000 00B259 (v02 HPQOEM 86E2 00001000 ACPI 00040000) ACPI: SSDT 0x0000000035FE6000 0001D2 (v01 HPQOEM 86E2 00001000 ACPI 00040000) ACPI: SSDT 0x0000000035FDE000 007166 (v01 HPQOEM 86E2 00001000 ACPI 00040000) ACPI: LPIT 0x0000000035FDC000 000094 (v01 HPQOEM 86E2 00000002 HP 00040000) ACPI: WSMT 0x0000000035FDB000 000028 (v01 HPQOEM 86E2 00000002 HP 00040000) ACPI: SSDT 0x0000000035FDA000 000B70 (v02 HPQOEM 86E2 00001000 ACPI 00040000) ACPI: DBGP 0x0000000035FD9000 000034 (v01 HPQOEM SLIC-MPC 00000002 HP 00040000) ACPI: DBG2 0x0000000035FD8000 000054 (v00 HPQOEM 86E2 00000002 HP 00040000) ACPI: SSDT 0x0000000035FD7000 0007EA (v02 HPQOEM 86E2 00001000 ACPI 00040000) ACPI: NHLT 0x0000000035FD6000 00002D (v00 HPQOEM 86E2 00000002 HP 00040000) ACPI: ECDT 0x0000000035FD5000 000069 (v01 HPQOEM 86E2 00000002 HP 00040000) ACPI: HPET 0x0000000035FD3000 000038 (v01 HPQOEM 86E2 00000002 HP 00040000) ACPI: APIC 0x0000000035FD2000 00012C (v03 HPQOEM 86E2 00000002 HP 00040000) ACPI: MCFG 0x0000000035FD1000 00003C (v01 HPQOEM 86E2 00000002 HP 00040000) ACPI: SSDT 0x0000000035FFD000 0000F5 (v01 HPQOEM 86E2 00000002 ACPI 00040000) ACPI: DMAR 0x0000000035F9B000 0000A8 (v02 HPQOEM 86E2 00000002 HP 00040000) ACPI: SSDT 0x0000000035F97000 00333E (v01 HPQOEM 86E2 00001000 ACPI 00040000) ACPI: SSDT 0x0000000035F96000 000987 (v02 HPQOEM 86E2 00001000 ACPI 00040000) ACPI: SSDT 0x0000000035F95000 000164 (v01 HPQOEM 86E2 00001000 ACPI 00040000) ACPI: SSDT 0x0000000035F94000 00005C (v02 HPQOEM 86E2 00001000 ACPI 00040000) ACPI: SSDT 0x0000000035F93000 00005C (v02 HPQOEM 86E2 00001000 ACPI 00040000) ACPI: FPDT 0x0000000035F92000 000044 (v01 HPQOEM SLIC-MPC 00000002 HP 00040000) ACPI: Local APIC address 0xfee00000 No NUMA configuration found Faking a node at [mem 0x0000000000000000-0x00000008bfbfffff] NODE_DATA(0) allocated [mem 0x8bfbd6000-0x8bfbfffff] Zone ranges: DMA [mem 0x0000000000001000-0x0000000000ffffff] DMA32 [mem 0x0000000001000000-0x00000000ffffffff] Normal [mem 0x0000000100000000-0x00000008bfbfffff] Device empty Movable zone start for each node Early memory node ranges node 0: [mem 0x0000000000001000-0x000000000009efff] node 0: [mem 0x0000000000100000-0x0000000023902fff] node 0: [mem 0x0000000023904000-0x0000000023959fff] node 0: [mem 0x000000002395b000-0x0000000032a2efff] node 0: [mem 0x0000000035fff000-0x0000000035ffffff] node 0: [mem 0x0000000100000000-0x00000008bfbfffff] Zeroed struct page in unavailable ranges: 23092 pages Initmem setup node 0 [mem 0x0000000000001000-0x00000008bfbfffff] On node 0 totalpages: 8332748 DMA zone: 64 pages used for memmap DMA zone: 21 pages reserved DMA zone: 3998 pages, LIFO batch:0 DMA32 zone: 3177 pages used for memmap DMA32 zone: 203310 pages, LIFO batch:63 Normal zone: 126960 pages used for memmap Normal zone: 8125440 pages, LIFO batch:63 x86/hpet: Will disable the HPET for this platform because it's not reliable Reserving Intel graphics memory at [mem 0x3c800000-0x403fffff] ACPI: PM-Timer IO Port: 0x1808 ACPI: Local APIC address 0xfee00000 ACPI: LAPIC_NMI (acpi_id[0x01] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x02] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x03] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x04] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x05] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x06] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x07] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x08] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x09] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x0a] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x0b] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x0c] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x0d] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x0e] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x0f] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x10] high edge lint[0x1]) IOAPIC[0]: apic_id 2, version 32, address 0xfec00000, GSI 0-119 ACPI: INT_SRC_OVR (bus 0 bus_irq 0 global_irq 2 dfl dfl) ACPI: INT_SRC_OVR (bus 0 bus_irq 9 global_irq 9 high level) ACPI: IRQ0 used by override. ACPI: IRQ9 used by override. Using ACPI (MADT) for SMP configuration information ACPI: HPET id: 0x8086a201 base: 0xfed00000 TSC deadline timer available smpboot: Allowing 8 CPUs, 0 hotplug CPUs PM: hibernation: Registered nosave memory: [mem 0x00000000-0x00000fff] PM: hibernation: Registered nosave memory: [mem 0x0009f000-0x0009ffff] PM: hibernation: Registered nosave memory: [mem 0x000a0000-0x000dffff] PM: hibernation: Registered nosave memory: [mem 0x000e0000-0x000fffff] PM: hibernation: Registered nosave memory: [mem 0x23903000-0x23903fff] PM: hibernation: Registered nosave memory: [mem 0x2395a000-0x2395afff] PM: hibernation: Registered nosave memory: [mem 0x32a2f000-0x3518efff] PM: hibernation: Registered nosave memory: [mem 0x3518f000-0x35f7efff] PM: hibernation: Registered nosave memory: [mem 0x35f7f000-0x35ffefff] PM: hibernation: Registered nosave memory: [mem 0x36000000-0x39ffffff] PM: hibernation: Registered nosave memory: [mem 0x3a000000-0x3a1fffff] PM: hibernation: Registered nosave memory: [mem 0x3a200000-0x3a3fffff] PM: hibernation: Registered nosave memory: [mem 0x3a400000-0x3affffff] PM: hibernation: Registered nosave memory: [mem 0x3b000000-0xcfffffff] PM: hibernation: Registered nosave memory: [mem 0xd0000000-0xfc7fffff] PM: hibernation: Registered nosave memory: [mem 0xfc800000-0xfe7fffff] PM: hibernation: Registered nosave memory: [mem 0xfe800000-0xfebfffff] PM: hibernation: Registered nosave memory: [mem 0xfec00000-0xfec00fff] PM: hibernation: Registered nosave memory: [mem 0xfec01000-0xfecfffff] PM: hibernation: Registered nosave memory: [mem 0xfed00000-0xfed00fff] PM: hibernation: Registered nosave memory: [mem 0xfed01000-0xfed0ffff] PM: hibernation: Registered nosave memory: [mem 0xfed10000-0xfed17fff] PM: hibernation: Registered nosave memory: [mem 0xfed18000-0xfed1ffff] PM: hibernation: Registered nosave memory: [mem 0xfed20000-0xfed7ffff] PM: hibernation: Registered nosave memory: [mem 0xfed80000-0xfed9ffff] PM: hibernation: Registered nosave memory: [mem 0xfeda0000-0xfeda1fff] PM: hibernation: Registered nosave memory: [mem 0xfeda2000-0xfedfffff] PM: hibernation: Registered nosave memory: [mem 0xfee00000-0xfee00fff] PM: hibernation: Registered nosave memory: [mem 0xfee01000-0xff3fffff] PM: hibernation: Registered nosave memory: [mem 0xff400000-0xffffffff] [mem 0xd0000000-0xfc7fffff] available for PCI devices Booting paravirtualized kernel on bare hardware clocksource: refined-jiffies: mask: 0xffffffff max_cycles: 0xffffffff, max_idle_ns: 7645519600211568 ns setup_percpu: NR_CPUS:8192 nr_cpumask_bits:8 nr_cpu_ids:8 nr_node_ids:1 percpu: Embedded 54 pages/cpu s183960 r8192 d29032 u262144 pcpu-alloc: s183960 r8192 d29032 u262144 alloc=1*2097152 pcpu-alloc: [0] 0 1 2 3 4 5 6 7 Built 1 zonelists, mobility grouping on. Total pages: 8202526 Policy zone: Normal Kernel command line: BOOT_IMAGE=/vmlinuz-5.10.0-1-amd64 root=/dev/mapper/debian--vg-root ro quiet Dentry cache hash table entries: 4194304 (order: 13, 33554432 bytes, linear) Inode-cache hash table entries: 2097152 (order: 12, 16777216 bytes, linear) mem auto-init: stack:off, heap alloc:on, heap free:off Memory: 827144K/33330992K available (12295K kernel code, 2540K rwdata, 4060K rodata, 2380K init, 1692K bss, 735384K reserved, 0K cma-reserved) random: get_random_u64 called from __kmem_cache_create+0x2e/0x550 with crng_init=0 SLUB: HWalign=64, Order=0-3, MinObjects=0, CPUs=8, Nodes=1 ftrace: allocating 35988 entries in 141 pages ftrace: allocated 141 pages with 4 groups rcu: Hierarchical RCU implementation. rcu: RCU restricting CPUs from NR_CPUS=8192 to nr_cpu_ids=8. Rude variant of Tasks RCU enabled. Tracing variant of Tasks RCU enabled. rcu: RCU calculated value of scheduler-enlistment delay is 25 jiffies. rcu: Adjusting geometry for rcu_fanout_leaf=16, nr_cpu_ids=8 NR_IRQS: 524544, nr_irqs: 2048, preallocated irqs: 16 random: crng done (trusting CPU's manufacturer) Console: colour VGA+ 80x25 printk: console [tty0] enabled ACPI: Core revision 20200925 APIC: Switch to symmetric I/O mode setup DMAR: Host address width 39 DMAR: DRHD base: 0x000000fed90000 flags: 0x0 DMAR: dmar0: reg_base_addr fed90000 ver 4:0 cap 1c0000c40660462 ecap 49e2ff0505e DMAR: DRHD base: 0x000000fed91000 flags: 0x1 DMAR: dmar1: reg_base_addr fed91000 ver 1:0 cap d2008c40660462 ecap f050da DMAR: RMRR base: 0x00000035041000 end: 0x00000035060fff DMAR: RMRR base: 0x0000003c000000 end: 0x000000403fffff DMAR-IR: IOAPIC id 2 under DRHD base 0xfed91000 IOMMU 1 DMAR-IR: HPET id 0 under DRHD base 0xfed91000 DMAR-IR: x2apic is disabled because BIOS sets x2apic opt out bit. DMAR-IR: Use 'intremap=no_x2apic_optout' to override the BIOS setting. DMAR-IR: Enabled IRQ remapping in xapic mode x2apic: IRQ remapping doesn't support X2APIC mode clocksource: tsc-early: mask: 0xffffffffffffffff max_cycles: 0x159647815e3, max_idle_ns: 440795269835 ns Calibrating delay loop (skipped), value calculated using timer frequency.. 2995.20 BogoMIPS (lpj=5990400) pid_max: default: 32768 minimum: 301 LSM: Security Framework initializing Yama: disabled by default; enable with sysctl kernel.yama.* AppArmor: AppArmor initialized TOMOYO Linux initialized Mount-cache hash table entries: 65536 (order: 7, 524288 bytes, linear) Mountpoint-cache hash table entries: 65536 (order: 7, 524288 bytes, linear) x86/cpu: User Mode Instruction Prevention (UMIP) activated mce: CPU0: Thermal monitoring enabled (TM1) process: using mwait in idle threads Last level iTLB entries: 4KB 0, 2MB 0, 4MB 0 Last level dTLB entries: 4KB 0, 2MB 0, 4MB 0, 1GB 0 Spectre V1 : Mitigation: usercopy/swapgs barriers and __user pointer sanitization Spectre V2 : Mitigation: Enhanced IBRS Spectre V2 : Spectre v2 / SpectreRSB mitigation: Filling RSB on context switch Spectre V2 : mitigation: Enabling conditional Indirect Branch Prediction Barrier Speculative Store Bypass: Mitigation: Speculative Store Bypass disabled via prctl and seccomp Freeing SMP alternatives memory: 32K smpboot: CPU0: Intel(R) Core(TM) i7-1065G7 CPU @ 1.30GHz (family: 0x6, model: 0x7e, stepping: 0x5) Performance Events: PEBS fmt4+-baseline, AnyThread deprecated, Icelake events, 32-deep LBR, full-width counters, Intel PMU driver. ... version: 5 ... bit width: 48 ... generic registers: 8 ... value mask: 0000ffffffffffff ... max period: 00007fffffffffff ... fixed-purpose events: 4 ... event mask: 0001000f000000ff rcu: Hierarchical SRCU implementation. NMI watchdog: Enabled. Permanently consumes one hw-PMU counter. smp: Bringing up secondary CPUs ... x86: Booting SMP configuration: .... node #0, CPUs: #1 #2 #3 #4 #5 #6 #7 smp: Brought up 1 node, 8 CPUs smpboot: Max logical packages: 1 smpboot: Total of 8 processors activated (23961.60 BogoMIPS) node 0 deferred pages initialised in 32ms devtmpfs: initialized x86/mm: Memory block size: 128MB PM: Registering ACPI NVS region [mem 0x23903000-0x23903fff] (4096 bytes) PM: Registering ACPI NVS region [mem 0x3518f000-0x35f7efff] (14614528 bytes) clocksource: jiffies: mask: 0xffffffff max_cycles: 0xffffffff, max_idle_ns: 7645041785100000 ns futex hash table entries: 2048 (order: 5, 131072 bytes, linear) pinctrl core: initialized pinctrl subsystem NET: Registered protocol family 16 audit: initializing netlink subsys (disabled) audit: type=2000 audit(1611164511.048:1): state=initialized audit_enabled=0 res=1 thermal_sys: Registered thermal governor 'fair_share' thermal_sys: Registered thermal governor 'bang_bang' thermal_sys: Registered thermal governor 'step_wise' thermal_sys: Registered thermal governor 'user_space' cpuidle: using governor ladder cpuidle: using governor menu ACPI FADT declares the system doesn't support PCIe ASPM, so disable it ACPI: bus type PCI registered acpiphp: ACPI Hot Plug PCI Controller Driver version: 0.5 PCI: MMCONFIG for domain 0000 [bus 00-ff] at [mem 0xc0000000-0xcfffffff] (base 0xc0000000) PCI: MMCONFIG at [mem 0xc0000000-0xcfffffff] reserved in E820 PCI: Using configuration type 1 for base access ENERGY_PERF_BIAS: Set to 'normal', was 'performance' HugeTLB registered 1.00 GiB page size, pre-allocated 0 pages HugeTLB registered 2.00 MiB page size, pre-allocated 0 pages ACPI: Added _OSI(Module Device) ACPI: Added _OSI(Processor Device) ACPI: Added _OSI(3.0 _SCP Extensions) ACPI: Added _OSI(Processor Aggregator Device) ACPI: Added _OSI(Linux-Dell-Video) ACPI: Added _OSI(Linux-Lenovo-NV-HDMI-Audio) ACPI: Added _OSI(Linux-HPI-Hybrid-Graphics) ACPI: 14 ACPI AML tables successfully acquired and loaded ACPI: EC: EC started ACPI: EC: interrupt blocked ACPI: EC: EC_CMD/EC_SC=0x66, EC_DATA=0x62 ACPI: EC: Boot ECDT EC used to handle transactions ACPI: Dynamic OEM Table Load: ACPI: SSDT 0xFFFF9F733E5DC000 0000F4 (v02 PmRef Cpu0Psd 00003000 INTL 20160422) ACPI: \_SB_.PR00: _OSC native thermal LVT Acked ACPI: Dynamic OEM Table Load: ACPI: SSDT 0xFFFF9F733E547C00 000386 (v02 PmRef Cpu0Cst 00003001 INTL 20160422) ACPI: Dynamic OEM Table Load: ACPI: SSDT 0xFFFF9F70811CF800 000437 (v02 PmRef Cpu0Ist 00003000 INTL 20160422) ACPI: Dynamic OEM Table Load: ACPI: SSDT 0xFFFF9F733E59F200 00012C (v02 PmRef Cpu0Hwp 00003000 INTL 20160422) ACPI: Dynamic OEM Table Load: ACPI: SSDT 0xFFFF9F70811CE800 000724 (v02 PmRef HwpLvt 00003000 INTL 20160422) ACPI: Dynamic OEM Table Load: ACPI: SSDT 0xFFFF9F70811CE000 0005FC (v02 PmRef ApIst 00003000 INTL 20160422) ACPI: Dynamic OEM Table Load: ACPI: SSDT 0xFFFF9F733E543400 000317 (v02 PmRef ApHwp 00003000 INTL 20160422) ACPI: Dynamic OEM Table Load: ACPI: SSDT 0xFFFF9F7080140000 000AB0 (v02 PmRef ApPsd 00003000 INTL 20160422) ACPI: Dynamic OEM Table Load: ACPI: SSDT 0xFFFF9F733E540800 00030A (v02 PmRef ApCst 00003000 INTL 20160422) ACPI: Interpreter enabled ACPI: (supports S0 S3 S4 S5) ACPI: Using IOAPIC for interrupt routing PCI: Using host bridge windows from ACPI; if necessary, use "pci=nocrs" and report a bug ACPI: Enabled 10 GPEs in block 00 to 7F ACPI: Power Resource [PC01] (on) ACPI: Power Resource [V0PR] (on) ACPI: Power Resource [V1PR] (on) ACPI: Power Resource [V2PR] (on) ACPI: Power Resource [WRST] (on) acpi ABCD0000:00: ACPI dock station (docks/bays count: 1) ACPI: Power Resource [TBT0] (on) ACPI: Power Resource [TBT1] (on) ACPI BIOS Error (bug): Could not resolve symbol [\_SB.IPPF._STA.POS1], AE_NOT_FOUND (20200925/psargs-330) ACPI Error: Aborting method \_SB.IPPF._STA due to previous error (AE_NOT_FOUND) (20200925/psparse-529) ACPI: Power Resource [PIN] (off) ACPI: PCI Root Bridge [PCI0] (domain 0000 [bus 00-fe]) acpi PNP0A08:00: _OSC: OS supports [ExtendedConfig ASPM ClockPM Segments MSI HPX-Type3] acpi PNP0A08:00: _OSC: platform does not support [AER] acpi PNP0A08:00: _OSC: OS now controls [PCIeHotplug SHPCHotplug PME PCIeCapability LTR] acpi PNP0A08:00: FADT indicates ASPM is unsupported, using BIOS configuration PCI host bridge to bus 0000:00 pci_bus 0000:00: root bus resource [io 0x0000-0x0cf7 window] pci_bus 0000:00: root bus resource [io 0x0d00-0xffff window] pci_bus 0000:00: root bus resource [mem 0x000a0000-0x000bffff window] pci_bus 0000:00: root bus resource [mem 0x40400000-0xbfffffff window] pci_bus 0000:00: root bus resource [bus 00-fe] pci 0000:00:00.0: [8086:8a12] type 00 class 0x060000 pci 0000:00:02.0: [8086:8a52] type 00 class 0x030000 pci 0000:00:02.0: reg 0x10: [mem 0x6b000000-0x6bffffff 64bit] pci 0000:00:02.0: reg 0x18: [mem 0x70000000-0x7fffffff 64bit pref] pci 0000:00:02.0: reg 0x20: [io 0x8000-0x803f] pci 0000:00:02.0: DMAR: Skip IOMMU disabling for graphics pci 0000:00:04.0: [8086:8a03] type 00 class 0x118000 pci 0000:00:04.0: reg 0x10: [mem 0x6c100000-0x6c10ffff 64bit] pci 0000:00:14.0: [8086:34ed] type 00 class 0x0c0330 pci 0000:00:14.0: reg 0x10: [mem 0x6c110000-0x6c11ffff 64bit] pci 0000:00:14.0: PME# supported from D3hot D3cold pci 0000:00:14.2: [8086:34ef] type 00 class 0x050000 pci 0000:00:14.2: reg 0x10: [mem 0x6c124000-0x6c125fff 64bit] pci 0000:00:14.2: reg 0x18: [mem 0x6c128000-0x6c128fff 64bit] pci 0000:00:15.0: [8086:34e8] type 00 class 0x0c8000 pci 0000:00:15.0: reg 0x10: [mem 0x00000000-0x00000fff 64bit] pci 0000:00:15.1: [8086:34e9] type 00 class 0x0c8000 pci 0000:00:15.1: reg 0x10: [mem 0x00000000-0x00000fff 64bit] pci 0000:00:16.0: [8086:34e0] type 00 class 0x078000 pci 0000:00:16.0: reg 0x10: [mem 0x6c12b000-0x6c12bfff 64bit] pci 0000:00:16.0: PME# supported from D3hot pci 0000:00:17.0: [8086:282a] type 00 class 0x010400 pci 0000:00:17.0: reg 0x10: [mem 0x6c126000-0x6c127fff] pci 0000:00:17.0: reg 0x14: [mem 0x6c131000-0x6c1310ff] pci 0000:00:17.0: reg 0x18: [io 0x8080-0x8087] pci 0000:00:17.0: reg 0x1c: [io 0x8088-0x808b] pci 0000:00:17.0: reg 0x20: [io 0x8060-0x807f] pci 0000:00:17.0: reg 0x24: [mem 0x6c130000-0x6c1307ff] pci 0000:00:17.0: PME# supported from D3hot pci 0000:00:1c.0: [8086:34b8] type 01 class 0x060400 pci 0000:00:1c.0: PME# supported from D0 D3hot D3cold pci 0000:00:1c.4: [8086:34bc] type 01 class 0x060400 pci 0000:00:1c.4: PME# supported from D0 D3hot D3cold pci 0000:00:1c.4: PTM enabled (root), 4ns granularity pci 0000:00:1d.0: [8086:34b0] type 01 class 0x060400 pci 0000:00:1d.0: PME# supported from D0 D3hot D3cold pci 0000:00:1d.0: PTM enabled (root), 4ns granularity pci 0000:00:1d.1: [8086:34b1] type 01 class 0x060400 pci 0000:00:1d.1: PME# supported from D0 D3hot D3cold pci 0000:00:1d.1: PTM enabled (root), 4ns granularity pci 0000:00:1d.4: [8086:34b4] type 01 class 0x060400 pci 0000:00:1d.4: PME# supported from D0 D3hot D3cold pci 0000:00:1d.4: PTM enabled (root), 4ns granularity pci 0000:00:1e.0: [8086:34a8] type 00 class 0x078000 pci 0000:00:1e.0: reg 0x10: [mem 0x00000000-0x00000fff 64bit] pci 0000:00:1e.2: [8086:34aa] type 00 class 0x0c8000 pci 0000:00:1e.2: reg 0x10: [mem 0x00000000-0x00000fff 64bit] pci 0000:00:1f.0: [8086:3482] type 00 class 0x060100 pci 0000:00:1f.3: [8086:34c8] type 00 class 0x040380 pci 0000:00:1f.3: reg 0x10: [mem 0x6c120000-0x6c123fff 64bit] pci 0000:00:1f.3: reg 0x20: [mem 0x6c000000-0x6c0fffff 64bit] pci 0000:00:1f.3: PME# supported from D3hot D3cold pci 0000:00:1f.4: [8086:34a3] type 00 class 0x0c0500 pci 0000:00:1f.4: reg 0x10: [mem 0x6c12e000-0x6c12e0ff 64bit] pci 0000:00:1f.4: reg 0x20: [io 0x8040-0x805f] pci 0000:00:1f.5: [8086:34a4] type 00 class 0x0c8000 pci 0000:00:1f.5: reg 0x10: [mem 0xfe010000-0xfe010fff] pci 0000:00:1c.0: PCI bridge to [bus 01-05] pci 0000:00:1c.0: bridge window [io 0x7000-0x7fff] pci 0000:00:1c.0: bridge window [mem 0x6a000000-0x6affffff] pci 0000:00:1c.0: bridge window [mem 0x62000000-0x62ffffff 64bit pref] pci 0000:06:00.0: [10de:1d13] type 00 class 0x030200 pci 0000:06:00.0: reg 0x10: [mem 0x69000000-0x69ffffff] pci 0000:06:00.0: reg 0x14: [mem 0x50000000-0x5fffffff 64bit pref] pci 0000:06:00.0: reg 0x1c: [mem 0x60000000-0x61ffffff 64bit pref] pci 0000:06:00.0: reg 0x24: [io 0x6000-0x607f] pci 0000:06:00.0: reg 0x30: [mem 0xfff80000-0xffffffff pref] pci 0000:06:00.0: Enabling HDA controller pci 0000:00:1c.4: PCI bridge to [bus 06-0a] pci 0000:00:1c.4: bridge window [io 0x6000-0x6fff] pci 0000:00:1c.4: bridge window [mem 0x69000000-0x69ffffff] pci 0000:00:1c.4: bridge window [mem 0x50000000-0x61ffffff 64bit pref] pci 0000:0b:00.0: [10ec:c822] type 00 class 0x028000 pci 0000:0b:00.0: reg 0x10: [io 0x5000-0x50ff] pci 0000:0b:00.0: reg 0x18: [mem 0x68000000-0x6800ffff 64bit] pci 0000:0b:00.0: supports D1 D2 pci 0000:0b:00.0: PME# supported from D0 D1 D2 D3hot D3cold pci 0000:00:1d.0: PCI bridge to [bus 0b-0f] pci 0000:00:1d.0: bridge window [io 0x5000-0x5fff] pci 0000:00:1d.0: bridge window [mem 0x68000000-0x68ffffff] pci 0000:00:1d.0: bridge window [mem 0x63000000-0x63ffffff 64bit pref] pci 0000:10:00.0: [10ec:8168] type 00 class 0x020000 pci 0000:10:00.0: reg 0x10: [io 0x4000-0x40ff] pci 0000:10:00.0: reg 0x18: [mem 0x67004000-0x67004fff 64bit] pci 0000:10:00.0: reg 0x20: [mem 0x67000000-0x67003fff 64bit] pci 0000:10:00.0: supports D1 D2 pci 0000:10:00.0: PME# supported from D0 D1 D2 D3hot D3cold pci 0000:00:1d.1: PCI bridge to [bus 10-14] pci 0000:00:1d.1: bridge window [io 0x4000-0x4fff] pci 0000:00:1d.1: bridge window [mem 0x67000000-0x67ffffff] pci 0000:00:1d.1: bridge window [mem 0x64000000-0x64ffffff 64bit pref] pci 0000:15:00.0: [15b7:5003] type 00 class 0x010802 pci 0000:15:00.0: reg 0x10: [mem 0x66000000-0x66003fff 64bit] pci 0000:00:1d.4: PCI bridge to [bus 15] pci 0000:00:1d.4: bridge window [io 0x3000-0x3fff] pci 0000:00:1d.4: bridge window [mem 0x66000000-0x66ffffff] pci 0000:00:1d.4: bridge window [mem 0x65000000-0x65ffffff 64bit pref] ACPI BIOS Error (bug): Could not resolve symbol [\_SB.IPPF._STA.POS1], AE_NOT_FOUND (20200925/psargs-330) ACPI Error: Aborting method \_SB.IPPF._STA due to previous error (AE_NOT_FOUND) (20200925/psparse-529) ACPI: EC: interrupt unblocked ACPI: EC: event unblocked ACPI: EC: EC_CMD/EC_SC=0x66, EC_DATA=0x62 ACPI: EC: GPE=0x6e ACPI: \_SB_.PCI0.LPCB.EC0_: Boot ECDT EC initialization complete ACPI: \_SB_.PCI0.LPCB.EC0_: EC: Used to handle transactions and events iommu: Default domain type: Translated pci 0000:00:02.0: vgaarb: setting as boot VGA device pci 0000:00:02.0: vgaarb: VGA device added: decodes=io+mem,owns=io+mem,locks=none pci 0000:00:02.0: vgaarb: bridge control possible vgaarb: loaded EDAC MC: Ver: 3.0.0 NetLabel: Initializing NetLabel: domain hash size = 128 NetLabel: protocols = UNLABELED CIPSOv4 CALIPSO NetLabel: unlabeled traffic allowed by default PCI: Using ACPI for IRQ routing PCI: pci_cache_line_size set to 64 bytes pci 0000:00:1f.5: can't claim BAR 0 [mem 0xfe010000-0xfe010fff]: no compatible bridge window e820: reserve RAM buffer [mem 0x0009f800-0x0009ffff] e820: reserve RAM buffer [mem 0x23903000-0x23ffffff] e820: reserve RAM buffer [mem 0x2395a000-0x23ffffff] e820: reserve RAM buffer [mem 0x32a2f000-0x33ffffff] e820: reserve RAM buffer [mem 0x36000000-0x37ffffff] e820: reserve RAM buffer [mem 0x8bfc00000-0x8bfffffff] clocksource: Switched to clocksource tsc-early VFS: Disk quotas dquot_6.6.0 VFS: Dquot-cache hash table entries: 512 (order 0, 4096 bytes) AppArmor: AppArmor Filesystem Enabled pnp: PnP ACPI init system 00:00: [io 0x0680-0x069f] has been reserved system 00:00: [io 0x164e-0x164f] has been reserved system 00:00: Plug and Play ACPI device, IDs PNP0c02 (active) pnp 00:01: Plug and Play ACPI device, IDs HPQ8001 PNP0303 (active) pnp 00:02: Plug and Play ACPI device, IDs ETD074c SYN1e00 SYN0002 PNP0f13 (active) system 00:03: [mem 0xfed10000-0xfed17fff] has been reserved system 00:03: [mem 0xfeda0000-0xfeda0fff] has been reserved system 00:03: [mem 0xfeda1000-0xfeda1fff] has been reserved system 00:03: [mem 0xc0000000-0xcfffffff] has been reserved system 00:03: [mem 0xfed20000-0xfed7ffff] has been reserved system 00:03: [mem 0xfed90000-0xfed93fff] could not be reserved system 00:03: [mem 0xfee00000-0xfeefffff] could not be reserved system 00:03: Plug and Play ACPI device, IDs PNP0c02 (active) system 00:04: [io 0x1800-0x18fe] has been reserved system 00:04: [mem 0xfd000000-0xfd68ffff] has been reserved system 00:04: [mem 0xfd6b0000-0xfd6cffff] has been reserved system 00:04: [mem 0xfd6f0000-0xfdffffff] has been reserved system 00:04: [mem 0xfe000000-0xfe01ffff] has been reserved system 00:04: [mem 0xfe200000-0xfe7fffff] has been reserved system 00:04: [mem 0xff000000-0xffffffff] could not be reserved system 00:04: Plug and Play ACPI device, IDs PNP0c02 (active) system 00:05: [io 0x2000-0x20fe] has been reserved system 00:05: Plug and Play ACPI device, IDs PNP0c02 (active) system 00:06: [mem 0xfe038000-0xfe038fff] has been reserved system 00:06: Plug and Play ACPI device, IDs PNP0c02 (active) ACPI BIOS Error (bug): Could not resolve symbol [\_SB.IPPF._STA.POS1], AE_NOT_FOUND (20200925/psargs-330) ACPI Error: Aborting method \_SB.IPPF._STA due to previous error (AE_NOT_FOUND) (20200925/psparse-529) pnp: PnP ACPI: found 7 devices clocksource: acpi_pm: mask: 0xffffff max_cycles: 0xffffff, max_idle_ns: 2085701024 ns NET: Registered protocol family 2 tcp_listen_portaddr_hash hash table entries: 16384 (order: 6, 262144 bytes, linear) TCP established hash table entries: 262144 (order: 9, 2097152 bytes, linear) TCP bind hash table entries: 65536 (order: 8, 1048576 bytes, linear) TCP: Hash tables configured (established 262144 bind 65536) UDP hash table entries: 16384 (order: 7, 524288 bytes, linear) UDP-Lite hash table entries: 16384 (order: 7, 524288 bytes, linear) NET: Registered protocol family 1 NET: Registered protocol family 44 pci 0000:06:00.0: can't claim BAR 6 [mem 0xfff80000-0xffffffff pref]: no compatible bridge window pci 0000:00:15.0: BAR 0: no space for [mem size 0x00001000 64bit] pci 0000:00:15.0: BAR 0: failed to assign [mem size 0x00001000 64bit] pci 0000:00:15.1: BAR 0: no space for [mem size 0x00001000 64bit] pci 0000:00:15.1: BAR 0: failed to assign [mem size 0x00001000 64bit] pci 0000:00:1e.0: BAR 0: no space for [mem size 0x00001000 64bit] pci 0000:00:1e.0: BAR 0: failed to assign [mem size 0x00001000 64bit] pci 0000:00:1e.2: BAR 0: no space for [mem size 0x00001000 64bit] pci 0000:00:1e.2: BAR 0: failed to assign [mem size 0x00001000 64bit] pci 0000:00:1f.5: BAR 0: no space for [mem size 0x00001000] pci 0000:00:1f.5: BAR 0: trying firmware assignment [mem 0xfe010000-0xfe010fff] pci 0000:00:1f.5: BAR 0: [mem 0xfe010000-0xfe010fff] conflicts with Reserved [mem 0xfc800000-0xfe7fffff] pci 0000:00:1f.5: BAR 0: failed to assign [mem size 0x00001000] pci 0000:00:1c.0: PCI bridge to [bus 01-05] pci 0000:00:1c.0: bridge window [io 0x7000-0x7fff] pci 0000:00:1c.0: bridge window [mem 0x6a000000-0x6affffff] pci 0000:00:1c.0: bridge window [mem 0x62000000-0x62ffffff 64bit pref] pci 0000:06:00.0: BAR 6: no space for [mem size 0x00080000 pref] pci 0000:06:00.0: BAR 6: failed to assign [mem size 0x00080000 pref] pci 0000:00:1c.4: PCI bridge to [bus 06-0a] pci 0000:00:1c.4: bridge window [io 0x6000-0x6fff] pci 0000:00:1c.4: bridge window [mem 0x69000000-0x69ffffff] pci 0000:00:1c.4: bridge window [mem 0x50000000-0x61ffffff 64bit pref] pci 0000:00:1d.0: PCI bridge to [bus 0b-0f] pci 0000:00:1d.0: bridge window [io 0x5000-0x5fff] pci 0000:00:1d.0: bridge window [mem 0x68000000-0x68ffffff] pci 0000:00:1d.0: bridge window [mem 0x63000000-0x63ffffff 64bit pref] pci 0000:00:1d.1: PCI bridge to [bus 10-14] pci 0000:00:1d.1: bridge window [io 0x4000-0x4fff] pci 0000:00:1d.1: bridge window [mem 0x67000000-0x67ffffff] pci 0000:00:1d.1: bridge window [mem 0x64000000-0x64ffffff 64bit pref] pci 0000:00:1d.4: PCI bridge to [bus 15] pci 0000:00:1d.4: bridge window [io 0x3000-0x3fff] pci 0000:00:1d.4: bridge window [mem 0x66000000-0x66ffffff] pci 0000:00:1d.4: bridge window [mem 0x65000000-0x65ffffff 64bit pref] pci_bus 0000:00: Some PCI device resources are unassigned, try booting with pci=realloc pci_bus 0000:00: resource 4 [io 0x0000-0x0cf7 window] pci_bus 0000:00: resource 5 [io 0x0d00-0xffff window] pci_bus 0000:00: resource 6 [mem 0x000a0000-0x000bffff window] pci_bus 0000:00: resource 7 [mem 0x40400000-0xbfffffff window] pci_bus 0000:01: resource 0 [io 0x7000-0x7fff] pci_bus 0000:01: resource 1 [mem 0x6a000000-0x6affffff] pci_bus 0000:01: resource 2 [mem 0x62000000-0x62ffffff 64bit pref] pci_bus 0000:06: resource 0 [io 0x6000-0x6fff] pci_bus 0000:06: resource 1 [mem 0x69000000-0x69ffffff] pci_bus 0000:06: resource 2 [mem 0x50000000-0x61ffffff 64bit pref] pci_bus 0000:0b: resource 0 [io 0x5000-0x5fff] pci_bus 0000:0b: resource 1 [mem 0x68000000-0x68ffffff] pci_bus 0000:0b: resource 2 [mem 0x63000000-0x63ffffff 64bit pref] pci_bus 0000:10: resource 0 [io 0x4000-0x4fff] pci_bus 0000:10: resource 1 [mem 0x67000000-0x67ffffff] pci_bus 0000:10: resource 2 [mem 0x64000000-0x64ffffff 64bit pref] pci_bus 0000:15: resource 0 [io 0x3000-0x3fff] pci_bus 0000:15: resource 1 [mem 0x66000000-0x66ffffff] pci_bus 0000:15: resource 2 [mem 0x65000000-0x65ffffff 64bit pref] pci 0000:00:02.0: Video device with shadowed ROM at [mem 0x000c0000-0x000dffff] PCI: CLS 64 bytes, default 64 Trying to unpack rootfs image as initramfs... Freeing initrd memory: 67280K DMAR: No ATSR found DMAR: dmar0: Using Queued invalidation DMAR: dmar1: Using Queued invalidation pci 0000:00:00.0: Adding to iommu group 0 pci 0000:00:02.0: Adding to iommu group 1 pci 0000:00:04.0: Adding to iommu group 2 pci 0000:00:14.0: Adding to iommu group 3 pci 0000:00:14.2: Adding to iommu group 3 pci 0000:00:15.0: Adding to iommu group 4 pci 0000:00:15.1: Adding to iommu group 4 pci 0000:00:16.0: Adding to iommu group 5 pci 0000:00:17.0: Adding to iommu group 6 pci 0000:00:1c.0: Adding to iommu group 7 pci 0000:00:1c.4: Adding to iommu group 8 pci 0000:00:1d.0: Adding to iommu group 9 pci 0000:00:1d.1: Adding to iommu group 10 pci 0000:00:1d.4: Adding to iommu group 11 pci 0000:00:1e.0: Adding to iommu group 12 pci 0000:00:1e.2: Adding to iommu group 12 pci 0000:00:1f.0: Adding to iommu group 13 pci 0000:00:1f.3: Adding to iommu group 13 pci 0000:00:1f.4: Adding to iommu group 13 pci 0000:00:1f.5: Adding to iommu group 13 pci 0000:06:00.0: Adding to iommu group 14 pci 0000:0b:00.0: Adding to iommu group 15 pci 0000:10:00.0: Adding to iommu group 16 pci 0000:15:00.0: Adding to iommu group 17 DMAR: Intel(R) Virtualization Technology for Directed I/O PCI-DMA: Using software bounce buffering for IO (SWIOTLB) software IO TLB: mapped [mem 0x000000002918f000-0x000000002d18f000] (64MB) clocksource: tsc: mask: 0xffffffffffffffff max_cycles: 0x159647815e3, max_idle_ns: 440795269835 ns clocksource: Switched to clocksource tsc platform rtc_cmos: registered platform RTC device (no PNP device found) Initialise system trusted keyrings Key type blacklist registered workingset: timestamp_bits=36 max_order=23 bucket_order=0 zbud: loaded integrity: Platform Keyring initialized Key type asymmetric registered Asymmetric key parser 'x509' registered Block layer SCSI generic (bsg) driver version 0.4 loaded (major 251) io scheduler mq-deadline registered pcieport 0000:00:1c.0: PME: Signaling with IRQ 122 pcieport 0000:00:1c.0: pciehp: Slot #0 AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug+ Surprise+ Interlock- NoCompl+ IbPresDis- LLActRep+ pcieport 0000:00:1c.4: PME: Signaling with IRQ 123 pcieport 0000:00:1d.0: PME: Signaling with IRQ 124 pcieport 0000:00:1d.1: PME: Signaling with IRQ 125 pcieport 0000:00:1d.4: PME: Signaling with IRQ 126 shpchp: Standard Hot Plug PCI Controller Driver version: 0.4 intel_idle: MWAIT substates: 0x11121020 Monitor-Mwait will be used to enter C-1 state Monitor-Mwait will be used to enter C-2 state Monitor-Mwait will be used to enter C-3 state ACPI: \_SB_.PR00: Found 3 idle states intel_idle: v0.5.1 model 0x7E intel_idle: Local APIC timer is reliable in all C-states [Firmware Bug]: Invalid critical threshold (0) thermal LNXTHERM:00: registered as thermal_zone0 ACPI: Thermal Zone [TZ01] (58 C) Serial: 8250/16550 driver, 4 ports, IRQ sharing enabled hpet_acpi_add: no address or irqs in _CRS Linux agpgart interface v0.103 AMD-Vi: AMD IOMMUv2 driver by Joerg Roedel <jroedel(a)suse.de> AMD-Vi: AMD IOMMUv2 functionality not available on this system i8042: PNP: PS/2 Controller [PNP0303:PS2K,PNP0f13:PS2M] at 0x60,0x64 irq 1,12 serio: i8042 KBD port at 0x60,0x64 irq 1 serio: i8042 AUX port at 0x60,0x64 irq 12 mousedev: PS/2 mouse device common for all mice rtc_cmos rtc_cmos: RTC can wake from S4 rtc_cmos rtc_cmos: registered as rtc0 rtc_cmos rtc_cmos: setting system clock to 2021-01-20T17:41:52 UTC (1611164512) rtc_cmos rtc_cmos: alarms up to one month, y3k, 114 bytes nvram intel_pstate: Intel P-state driver initializing intel_pstate: HWP enabled ledtrig-cpu: registered to indicate activity on CPUs NET: Registered protocol family 10 Segment Routing with IPv6 mip6: Mobile IPv6 NET: Registered protocol family 17 mpls_gso: MPLS GSO support microcode: sig=0x706e5, pf=0x80, revision=0xa0 microcode: Microcode Update Driver: v2.2. IPI shorthand broadcast: enabled sched_clock: Marking stable (1117818972, 8996860)->(1171767835, -44952003) registered taskstats version 1 Loading compiled-in X.509 certificates Loaded X.509 cert 'Debian Secure Boot CA: 6ccece7e4c6c0d1f6149f3dd27dfcc5cbb419ea1' Loaded X.509 cert 'Debian Secure Boot Signer 2020: 00b55eb3b9' zswap: loaded using pool lzo/zbud Key type ._fscrypt registered Key type .fscrypt registered Key type fscrypt-provisioning registered AppArmor: AppArmor sha1 policy hashing enabled Freeing unused kernel image (initmem) memory: 2380K input: AT Translated Set 2 keyboard as /devices/platform/i8042/serio0/input/input0 Write protecting the kernel read-only data: 18432k Freeing unused kernel image (text/rodata gap) memory: 2040K Freeing unused kernel image (rodata/data gap) memory: 36K x86/mm: Checked W+X mappings: passed, no W+X pages found. Run /init as init process with arguments: /init with environment: HOME=/ TERM=linux BOOT_IMAGE=/vmlinuz-5.10.0-1-amd64 input: Lid Switch as /devices/LNXSYSTM:00/LNXSYBUS:00/PNP0C0D:00/input/input2 ACPI: Lid Switch [LID0] input: Power Button as /devices/LNXSYSTM:00/LNXSYBUS:00/PNP0C0C:00/input/input3 ACPI: Power Button [PWRB] input: Power Button as /devices/LNXSYSTM:00/LNXPWRBN:00/input/input4 wmi_bus wmi_bus-PNP0C14:01: WQ data block query control method not found wmi_bus wmi_bus-PNP0C14:01: WQ data block query control method not found wmi_bus wmi_bus-PNP0C14:01: WQ data block query control method not found wmi_bus wmi_bus-PNP0C14:01: WQ data block query control method not found acpi PNP0C14:04: duplicate WMI GUID 05901221-D566-11D1-B2F0-00A0C9062910 (first instance was on PNP0C14:03) acpi PNP0C14:05: duplicate WMI GUID 05901221-D566-11D1-B2F0-00A0C9062910 (first instance was on PNP0C14:03) wmi_bus wmi_bus-PNP0C14:05: WQBJ data block query control method not found wmi_bus wmi_bus-PNP0C14:05: WQBK data block query control method not found i801_smbus 0000:00:1f.4: SPD Write Disable is set i801_smbus 0000:00:1f.4: SMBus using PCI interrupt i2c i2c-0: 2/2 memory slots populated (from DMI) i2c i2c-0: Successfully instantiated SPD at 0x50 SCSI subsystem initialized r8169 0000:10:00.0: can't disable ASPM; OS doesn't have ASPM control ACPI: bus type USB registered usbcore: registered new interface driver usbfs usbcore: registered new interface driver hub usbcore: registered new device driver usb libata version 3.00 loaded. ACPI: Power Button [PWRF] xhci_hcd 0000:00:14.0: xHCI Host Controller xhci_hcd 0000:00:14.0: new USB bus registered, assigned bus number 1 ahci 0000:00:17.0: version 3.0 xhci_hcd 0000:00:14.0: hcc params 0x20007fc1 hci version 0x110 quirks 0x0000000000009810 xhci_hcd 0000:00:14.0: cache line size of 64 is not supported ahci 0000:00:17.0: AHCI 0001.0301 32 slots 2 ports 6 Gbps 0x1 impl RAID mode ahci 0000:00:17.0: flags: 64bit ncq sntf pm clo only pio slum part deso sadm sds apst usb usb1: New USB device found, idVendor=1d6b, idProduct=0002, bcdDevice= 5.10 usb usb1: New USB device strings: Mfr=3, Product=2, SerialNumber=1 usb usb1: Product: xHCI Host Controller usb usb1: Manufacturer: Linux 5.10.0-1-amd64 xhci-hcd usb usb1: SerialNumber: 0000:00:14.0 hub 1-0:1.0: USB hub found hub 1-0:1.0: 12 ports detected xhci_hcd 0000:00:14.0: xHCI Host Controller xhci_hcd 0000:00:14.0: new USB bus registered, assigned bus number 2 xhci_hcd 0000:00:14.0: Host supports USB 3.1 Enhanced SuperSpeed usb usb2: New USB device found, idVendor=1d6b, idProduct=0003, bcdDevice= 5.10 usb usb2: New USB device strings: Mfr=3, Product=2, SerialNumber=1 usb usb2: Product: xHCI Host Controller usb usb2: Manufacturer: Linux 5.10.0-1-amd64 xhci-hcd usb usb2: SerialNumber: 0000:00:14.0 hub 2-0:1.0: USB hub found hub 2-0:1.0: 6 ports detected cryptd: max_cpu_qlen set to 1000 scsi host0: ahci AVX2 version of gcm_enc/dec engaged. AES CTR mode by8 optimization enabled nvme nvme0: pci function 0000:15:00.0 scsi host1: ahci ata1: SATA max UDMA/133 abar m2048@0x6c130000 port 0x6c130100 irq 128 ata2: DUMMY libphy: r8169: probed r8169 0000:10:00.0 eth0: RTL8168h/8111h, 38:22:e2:1c:ff:ff, XID 541, IRQ 129 r8169 0000:10:00.0 eth0: jumbo features [frames: 9194 bytes, tx checksumming: ko] r8169 0000:10:00.0 eno1: renamed from eth0 nvme nvme0: 8/0/0 default/read/poll queues nvme0n1: p1 p2 < p5 > battery: ACPI: Battery Slot [BAT0] (battery present) i915 0000:00:02.0: [drm] VT-d active for gfx access i915 0000:00:02.0: vgaarb: deactivate vga console Console: switching to colour dummy device 80x25 ACPI Warning: \_SB.PCI0.GFX0._DSM: Argument #4 type mismatch - Found [Buffer], ACPI requires [Package] (20200925/nsarguments-61) ACPI Warning: \_SB.PCI0.RP05.PXSX._DSM: Argument #4 type mismatch - Found [Buffer], ACPI requires [Package] (20200925/nsarguments-61) pci 0000:06:00.0: optimus capabilities: enabled, status dynamic power, VGA switcheroo: detected Optimus DSM method \_SB_.PCI0.RP05.PXSX handle nouveau: detected PR support, will not use DSM nouveau 0000:06:00.0: enabling device (0006 -> 0007) i915 0000:00:02.0: vgaarb: changed VGA decodes: olddecodes=io+mem,decodes=io+mem:owns=io+mem nouveau 0000:06:00.0: NVIDIA GP108 (138000a1) i915 0000:00:02.0: firmware: direct-loading firmware i915/icl_dmc_ver1_09.bin i915 0000:00:02.0: [drm] Finished loading DMC firmware i915/icl_dmc_ver1_09.bin (v1.9) nouveau 0000:06:00.0: bios: version 86.08.40.00.2e nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/nvdec/scrubber.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/acr/bl.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/acr/ucode_load.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/acr/unload_bl.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/acr/ucode_unload.bin nouveau 0000:06:00.0: pmu: firmware unavailable nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/gr/fecs_bl.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/gr/fecs_inst.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/gr/fecs_data.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/gr/fecs_sig.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/gr/gpccs_bl.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/gr/gpccs_inst.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/gr/gpccs_data.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/gr/gpccs_sig.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/gr/sw_nonctx.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/gr/sw_ctx.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/gr/sw_bundle_init.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/gr/sw_method_init.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/sec2/sig.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/sec2/image.bin nouveau 0000:06:00.0: firmware: direct-loading firmware nvidia/gp108/sec2/desc.bin nouveau 0000:06:00.0: fb: 2048 MiB GDDR5 ata1: SATA link down (SStatus 4 SControl 300) usb 1-3: new high-speed USB device number 2 using xhci_hcd psmouse serio1: elantech: assuming hardware version 4 (with firmware version 0x4d5f02) psmouse serio1: elantech: Synaptics capabilities query result 0x10, 0x1a, 0x0d. usb 1-3: New USB device found, idVendor=04ca, idProduct=707f, bcdDevice= 0.03 usb 1-3: New USB device strings: Mfr=3, Product=1, SerialNumber=2 usb 1-3: Product: HP Wide Vision HD Camera usb 1-3: Manufacturer: DHKQA019ID97Y5 usb 1-3: SerialNumber: 200901010001 psmouse serio1: elantech: Elan sample query result 04, 4d, 86 input: ETPS/2 Elantech Touchpad as /devices/platform/i8042/serio1/input/input5 usb 1-10: new full-speed USB device number 3 using xhci_hcd usb 1-10: New USB device found, idVendor=0bda, idProduct=b00c, bcdDevice= 0.00 usb 1-10: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-10: Product: Bluetooth Radio usb 1-10: Manufacturer: Realtek usb 1-10: SerialNumber: 00e04c000001 ------------[ cut here ]------------ nouveau 0000:06:00.0: timeout WARNING: CPU: 5 PID: 169 at drivers/gpu/drm/nouveau/nvkm/subdev/pmu/base.c:107 nvkm_pmu_reset+0x148/0x160 [nouveau] Modules linked in: nouveau(+) i915(+) crc32_pclmul crc32c_intel ghash_clmulni_intel nvme aesni_intel libaes crypto_simd mxm_wmi ttm ahci nvme_core xhci_pci i2c_algo_bit libahci xhci_hcd drm_kms_helper t10_pi crc_t10dif libata psmouse cryptd glue_helper cec usbcore r8169 drm crct10dif_generic scsi_mod realtek mdio_devres i2c_i801 libphy i2c_smbus crct10dif_pclmul intel_lpss_pci crct10dif_common intel_lpss usb_common idma64 battery video wmi button CPU: 5 PID: 169 Comm: systemd-udevd Not tainted 5.10.0-1-amd64 #1 Debian 5.10.4-1 Hardware name: HP HP Pavilion Laptop 15-cs3xxx/86E2, BIOS F.05 01/01/2020 RIP: 0010:nvkm_pmu_reset+0x148/0x160 [nouveau] Code: 04 24 48 8b 40 10 48 8b 78 10 4c 8b 67 50 4d 85 e4 74 1e e8 9a de 78 cd 4c 89 e2 48 c7 c7 a1 d1 d8 c0 48 89 c6 e8 38 6a 9e cd <0f> 0b e9 50 ff ff ff 4c 8b 27 eb dd 4c 8b 27 eb a5 e8 52 dd a1 cd RSP: 0018:ffffc388804278e0 EFLAGS: 00010282 RAX: 0000000000000000 RBX: ffff9f70b993e800 RCX: ffffffff8f0b3228 RDX: c0000000ffffefff RSI: 00000000ffffefff RDI: 0000000000000247 RBP: ffff9f70a20cee00 R08: 0000000000000000 R09: ffffc38880427700 R10: ffffc388804276f8 R11: ffffffff8f0cb268 R12: ffff9f708129cb50 R13: 0000000058779b71 R14: 0000000053aa5380 R15: ffff9f70b9961e00 FS: 00007f6b768b38c0(0000) GS:ffff9f781fb40000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007fb13a7ba6f4 CR3: 00000003be160004 CR4: 0000000000770ee0 PKRU: 55555554 Call Trace: nvkm_pmu_init+0x16/0x40 [nouveau] nvkm_subdev_init+0x92/0xd0 [nouveau] ? ktime_get+0x38/0xa0 nvkm_device_init+0x10b/0x190 [nouveau] nvkm_udevice_init+0x41/0x60 [nouveau] nvkm_object_init+0x3e/0x100 [nouveau] nvkm_ioctl_new+0x169/0x200 [nouveau] ? nvkm_client_notify+0x30/0x30 [nouveau] ? nvkm_udevice_rd08+0x20/0x20 [nouveau] nvkm_ioctl+0xde/0x180 [nouveau] nvif_object_ctor+0x11e/0x1c0 [nouveau] nvif_device_ctor+0x1f/0x60 [nouveau] nouveau_cli_init+0x197/0x450 [nouveau] ? kmem_cache_alloc_trace+0xf5/0x200 nouveau_drm_device_init+0x74/0x760 [nouveau] ? pci_bus_read_config_word+0x49/0x70 nouveau_drm_probe+0x12f/0x1f0 [nouveau] local_pci_probe+0x42/0x80 ? _cond_resched+0x16/0x40 pci_device_probe+0xfd/0x1b0 really_probe+0x205/0x460 driver_probe_device+0xe1/0x150 device_driver_attach+0xa1/0xb0 __driver_attach+0x8a/0x150 ? device_driver_attach+0xb0/0xb0 ? device_driver_attach+0xb0/0xb0 bus_for_each_dev+0x78/0xc0 bus_add_driver+0x12b/0x1e0 driver_register+0x8b/0xe0 ? 0xffffffffc0769000 do_one_initcall+0x44/0x1d0 ? do_init_module+0x23/0x250 ? kmem_cache_alloc_trace+0xf5/0x200 do_init_module+0x5c/0x250 __do_sys_finit_module+0xb1/0x110 do_syscall_64+0x33/0x80 entry_SYSCALL_64_after_hwframe+0x44/0xa9 RIP: 0033:0x7f6b76d6c9b9 Code: 00 c3 66 2e 0f 1f 84 00 00 00 00 00 0f 1f 44 00 00 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 8b 0d a7 54 0c 00 f7 d8 64 89 01 48 RSP: 002b:00007ffdee6f4268 EFLAGS: 00000246 ORIG_RAX: 0000000000000139 RAX: ffffffffffffffda RBX: 000055dfe3ed3ef0 RCX: 00007f6b76d6c9b9 RDX: 0000000000000000 RSI: 00007f6b76ef7e2d RDI: 000000000000000f RBP: 0000000000020000 R08: 0000000000000000 R09: 000055dfe3ed41a0 R10: 000000000000000f R11: 0000000000000246 R12: 00007f6b76ef7e2d R13: 0000000000000000 R14: 000055dfe3eb9340 R15: 000055dfe3ed3ef0 ---[ end trace cf1844e25a37ee57 ]--- [drm] Initialized i915 1.6.0 20200917 for 0000:00:02.0 on minor 0 [TTM] Zone kernel: Available graphics memory: 16333688 KiB [TTM] Zone dma32: Available graphics memory: 2097152 KiB [TTM] Initializing pool allocator [TTM] Initializing DMA pool allocator nouveau 0000:06:00.0: DRM: VRAM: 2048 MiB nouveau 0000:06:00.0: DRM: GART: 536870912 MiB nouveau 0000:06:00.0: DRM: BIT table 'A' not found nouveau 0000:06:00.0: DRM: BIT table 'L' not found nouveau 0000:06:00.0: DRM: Pointer to TMDS table not found nouveau 0000:06:00.0: DRM: DCB version 4.1 nouveau 0000:06:00.0: DRM: MM: using COPY for buffer copies [drm] Initialized nouveau 1.3.1 20120801 for 0000:06:00.0 on minor 1 ACPI: Video Device [GFX0] (multi-head: yes rom: no post: no) input: Video Bus as /devices/LNXSYSTM:00/LNXSYBUS:00/PNP0A08:00/LNXVIDEO:00/input/input6 [Firmware Bug]: ACPI(PXSX) defines _DOD but not _DOS ACPI: Video Device [PXSX] (multi-head: yes rom: yes post: no) input: Video Bus as /devices/LNXSYSTM:00/LNXSYBUS:00/PNP0A08:00/device:44/LNXVIDEO:01/input/input7 vga_switcheroo: enabled fbcon: i915drmfb (fb0) is primary device Console: switching to colour frame buffer device 240x67 i915 0000:00:02.0: [drm] fb0: i915drmfb frame buffer device device-mapper: uevent: version 1.0.3 device-mapper: ioctl: 4.43.0-ioctl (2020-10-01) initialised: dm-devel(a)redhat.com PM: Image not found (code -22) EXT4-fs (dm-1): mounted filesystem with ordered data mode. Opts: (null) Not activating Mandatory Access Control as /sbin/tomoyo-init does not exist. EXT4-fs (dm-1): re-mounted. Opts: errors=remount-ro fuse: init (API version 7.32) lp: driver loaded but no devices found ppdev: user-space parallel port driver intel_pmc_core INT33A1:00: initialized ACPI: AC Adapter [ADP1] (on-line) input: HP Wireless hotkeys as /devices/virtual/input/input8 hp_accel: laptop model unknown, using default axes configuration lis3lv02d: 8 bits 3DC sensor found mei_me 0000:00:16.0: enabling device (0000 -> 0002) input: ST LIS3LV02DL Accelerometer as /devices/platform/lis3lv02d/input/input9 intel_rapl_common: Found RAPL domain package proc_thermal 0000:00:04.0: Creating sysfs group for PROC_THERMAL_PCI EXT4-fs (nvme0n1p1): mounting ext2 file system using the ext4 subsystem EXT4-fs (nvme0n1p1): mounted filesystem without journal. Opts: (null) ext2 filesystem being mounted at /boot supports timestamps until 2038 (0x7fffffff) iTCO_vendor_support: vendor-support=0 input: PC Speaker as /devices/platform/pcspkr/input/input10 mc: Linux media interface: v0.10 iTCO_wdt: Intel TCO WatchDog Timer Driver v1.11 iTCO_wdt: Found a Intel PCH TCO device (Version=6, TCOBASE=0x0400) iTCO_wdt: initialized. heartbeat=30 sec (nowayout=0) alg: No test for fips(ansi_cprng) (fips_ansi_cprng) videodev: Linux video capture interface: v2.00 audit: type=1400 audit(1611164581.415:2): apparmor="STATUS" operation="profile_load" profile="unconfined" name="/usr/sbin/haveged" pid=563 comm="apparmor_parser" audit: type=1400 audit(1611164581.415:3): apparmor="STATUS" operation="profile_load" profile="unconfined" name="nvidia_modprobe" pid=569 comm="apparmor_parser" audit: type=1400 audit(1611164581.415:4): apparmor="STATUS" operation="profile_load" profile="unconfined" name="nvidia_modprobe//kmod" pid=569 comm="apparmor_parser" audit: type=1400 audit(1611164581.415:5): apparmor="STATUS" operation="profile_load" profile="unconfined" name="mysqld_akonadi" pid=567 comm="apparmor_parser" audit: type=1400 audit(1611164581.415:6): apparmor="STATUS" operation="profile_load" profile="unconfined" name="mariadbd_akonadi" pid=562 comm="apparmor_parser" audit: type=1400 audit(1611164581.415:7): apparmor="STATUS" operation="profile_load" profile="unconfined" name="/usr/sbin/cups-browsed" pid=564 comm="apparmor_parser" audit: type=1400 audit(1611164581.415:8): apparmor="STATUS" operation="profile_load" profile="unconfined" name="/usr/bin/akonadiserver" pid=565 comm="apparmor_parser" audit: type=1400 audit(1611164581.415:9): apparmor="STATUS" operation="profile_load" profile="unconfined" name="postgresql_akonadi" pid=568 comm="apparmor_parser" audit: type=1400 audit(1611164581.419:10): apparmor="STATUS" operation="profile_load" profile="unconfined" name="libreoffice-senddoc" pid=572 comm="apparmor_parser" audit: type=1400 audit(1611164581.419:11): apparmor="STATUS" operation="profile_load" profile="unconfined" name="libreoffice-xpdfimport" pid=578 comm="apparmor_parser" uvcvideo: Found UVC 1.00 device HP Wide Vision HD Camera (04ca:707f) input: HP Wide Vision HD Camera: HP Wi as /devices/pci0000:00/0000:00:14.0/usb1/1-3/1-3:1.0/input/input12 usbcore: registered new interface driver uvcvideo USB Video Class driver (1.1.1) cfg80211: Loading compiled-in X.509 certificates for regulatory database cfg80211: Loaded X.509 cert 'benh(a)debian.org: 577e021cb980e0e820821ba7b54b4961b8b4fadf' cfg80211: Loaded X.509 cert 'romain.perier(a)gmail.com: 3abbc6ec146e09d1b6016ab9d6cf71dd233f0328' cfg80211: Loaded X.509 cert 'sforshee: 00b28ddf47aef9cea7' platform regulatory.0: firmware: direct-loading firmware regulatory.db platform regulatory.0: firmware: direct-loading firmware regulatory.db.p7s RAPL PMU: API unit is 2^-32 Joules, 3 fixed counters, 655360 ms ovfl timer RAPL PMU: hw unit of domain pp0-core 2^-14 Joules RAPL PMU: hw unit of domain package 2^-14 Joules RAPL PMU: hw unit of domain pp1-gpu 2^-14 Joules Adding 1003516k swap on /dev/mapper/debian--vg-swap_1. Priority:-2 extents:1 across:1003516k SSFS snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040380 BUG: kernel NULL pointer dereference, address: 0000000000000050 #PF: supervisor read access in kernel mode #PF: error_code(0x0000) - not-present page PGD 0 P4D 0 Oops: 0000 [#1] SMP NOPTI CPU: 6 PID: 472 Comm: systemd-udevd Tainted: G W 5.10.0-1-amd64 #1 Debian 5.10.4-1 Hardware name: HP HP Pavilion Laptop 15-cs3xxx/86E2, BIOS F.05 01/01/2020 RIP: 0010:acpi_ns_validate_handle+0x1a/0x23 Code: 00 48 83 c4 10 5b 5d 41 5c 41 5d 41 5e 41 5f c3 0f 1f 44 00 00 48 8d 57 ff 48 89 f8 48 83 fa fd 76 08 48 8b 05 0c b8 67 01 c3 <80> 7f 08 0f 74 02 31 c0 c3 0f 1f 44 00 00 48 8b 3d f6 b7 67 01 e8 RSP: 0000:ffffc388807c7b20 EFLAGS: 00010213 RAX: 0000000000000048 RBX: ffffc388807c7b70 RCX: 0000000000000000 RDX: 0000000000000047 RSI: 0000000000000246 RDI: 0000000000000048 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: ffffffffc0f5f4d1 R11: ffffffff8f0cb268 R12: 0000000000001001 R13: ffffffff8e33b160 R14: 0000000000000048 R15: 0000000000000000 FS: 00007f24548288c0(0000) GS:ffff9f781fb80000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000000000050 CR3: 0000000106158004 CR4: 0000000000770ee0 PKRU: 55555554 Call Trace: acpi_get_data_full+0x4d/0x92 acpi_bus_get_device+0x1f/0x40 sdw_intel_acpi_scan+0x59/0x230 [soundwire_intel] ? strstr+0x22/0x60 ? dmi_matches+0x76/0xe0 snd_intel_dsp_driver_probe.cold+0xaf/0x163 [snd_intel_dspcfg] azx_probe+0x7a/0x970 [snd_hda_intel] local_pci_probe+0x42/0x80 ? _cond_resched+0x16/0x40 pci_device_probe+0xfd/0x1b0 really_probe+0x205/0x460 driver_probe_device+0xe1/0x150 device_driver_attach+0xa1/0xb0 __driver_attach+0x8a/0x150 ? device_driver_attach+0xb0/0xb0 ? device_driver_attach+0xb0/0xb0 bus_for_each_dev+0x78/0xc0 bus_add_driver+0x12b/0x1e0 driver_register+0x8b/0xe0 ? 0xffffffffc0f65000 do_one_initcall+0x44/0x1d0 ? do_init_module+0x23/0x250 ? kmem_cache_alloc_trace+0xf5/0x200 do_init_module+0x5c/0x250 __do_sys_finit_module+0xb1/0x110 do_syscall_64+0x33/0x80 entry_SYSCALL_64_after_hwframe+0x44/0xa9 RIP: 0033:0x7f2454ce19b9 Code: 00 c3 66 2e 0f 1f 84 00 00 00 00 00 0f 1f 44 00 00 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 8b 0d a7 54 0c 00 f7 d8 64 89 01 48 RSP: 002b:00007fff0b66be58 EFLAGS: 00000246 ORIG_RAX: 0000000000000139 RAX: ffffffffffffffda RBX: 0000557d20c7a8b0 RCX: 00007f2454ce19b9 RDX: 0000000000000000 RSI: 00007f2454e6ce2d RDI: 000000000000001c RBP: 0000000000020000 R08: 0000000000000000 R09: 0000557d20c80d50 R10: 000000000000001c R11: 0000000000000246 R12: 00007f2454e6ce2d R13: 0000000000000000 R14: 0000557d20ae9c00 R15: 0000557d20c7a8b0 Modules linked in: snd_hda_intel(+) snd_intel_dspcfg soundwire_intel soundwire_generic_allocation kvm snd_soc_core snd_compress pcc_cpufreq(-) irqbypass soundwire_cadence acpi_cpufreq(-) rapl snd_hda_codec jitterentropy_rng uvcvideo intel_cstate cfg80211 snd_hda_core videobuf2_vmalloc videobuf2_memops videobuf2_v4l2 snd_hwdep videobuf2_common soundwire_bus intel_uncore videodev snd_pcm drbg snd_timer iTCO_wdt snd ansi_cprng joydev binfmt_misc hp_wmi(+) intel_pmc_bxt sparse_keymap mc pcspkr ecdh_generic serio_raw wmi_bmof intel_wmi_thunderbolt iTCO_vendor_support ecc soundcore watchdog rfkill libarc4 processor_thermal_device mei_me intel_rapl_common mei intel_soc_dts_iosf int3403_thermal int340x_thermal_zone hp_accel lis3lv02d hp_wireless int3400_thermal evdev acpi_thermal_rel intel_pmc_core acpi_pad ac acpi_tad parport_pc ppdev lp parport fuse configfs ip_tables x_tables autofs4 ext4 crc16 mbcache jbd2 crc32c_generic dm_crypt dm_mod nouveau i915 crc32_pclmul crc32c_intel ghash_clmulni_intel nvme aesni_intel libaes crypto_simd mxm_wmi ttm ahci nvme_core xhci_pci i2c_algo_bit libahci xhci_hcd drm_kms_helper t10_pi crc_t10dif libata psmouse cryptd glue_helper cec usbcore r8169 drm crct10dif_generic scsi_mod realtek mdio_devres i2c_i801 libphy i2c_smbus crct10dif_pclmul intel_lpss_pci crct10dif_common intel_lpss usb_common idma64 battery video wmi button CR2: 0000000000000050 ---[ end trace cf1844e25a37ee58 ]--- RIP: 0010:acpi_ns_validate_handle+0x1a/0x23 Code: 00 48 83 c4 10 5b 5d 41 5c 41 5d 41 5e 41 5f c3 0f 1f 44 00 00 48 8d 57 ff 48 89 f8 48 83 fa fd 76 08 48 8b 05 0c b8 67 01 c3 <80> 7f 08 0f 74 02 31 c0 c3 0f 1f 44 00 00 48 8b 3d f6 b7 67 01 e8 RSP: 0000:ffffc388807c7b20 EFLAGS: 00010213 RAX: 0000000000000048 RBX: ffffc388807c7b70 RCX: 0000000000000000 RDX: 0000000000000047 RSI: 0000000000000246 RDI: 0000000000000048 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: ffffffffc0f5f4d1 R11: ffffffff8f0cb268 R12: 0000000000001001 R13: ffffffff8e33b160 R14: 0000000000000048 R15: 0000000000000000 FS: 00007f24548288c0(0000) GS:ffff9f781fb80000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000000000050 CR3: 0000000106158004 CR4: 0000000000770ee0 PKRU: 55555554 Bluetooth: Core ver 2.22 intel_rapl_common: Found RAPL domain package intel_rapl_common: Found RAPL domain core intel_rapl_common: Found RAPL domain uncore NET: Registered protocol family 31 Bluetooth: HCI device and connection manager initialized Bluetooth: HCI socket layer initialized Bluetooth: L2CAP socket layer initialized Bluetooth: SCO socket layer initialized rtw_8822ce 0000:0b:00.0: firmware: direct-loading firmware rtw88/rtw8822c_wow_fw.bin rtw_8822ce 0000:0b:00.0: Firmware version 9.9.4, H2C version 15 rtw_8822ce 0000:0b:00.0: firmware: direct-loading firmware rtw88/rtw8822c_fw.bin rtw_8822ce 0000:0b:00.0: Firmware version 9.9.4, H2C version 15 r8169 0000:10:00.0: firmware: direct-loading firmware rtl_nic/rtl8168h-2.fw Generic FE-GE Realtek PHY r8169-1000:00: attached PHY driver [Generic FE-GE Realtek PHY] (mii_bus:phy_addr=r8169-1000:00, irq=IGNORE) usbcore: registered new interface driver btusb Bluetooth: hci0: RTL: examining hci_ver=0a hci_rev=000c lmp_ver=0a lmp_subver=8822 Bluetooth: hci0: RTL: rom_version status=0 version=3 Bluetooth: hci0: RTL: loading rtl_bt/rtl8822cu_fw.bin bluetooth hci0: firmware: direct-loading firmware rtl_bt/rtl8822cu_fw.bin Bluetooth: hci0: RTL: loading rtl_bt/rtl8822cu_config.bin bluetooth hci0: firmware: failed to load rtl_bt/rtl8822cu_config.bin (-2) firmware_class: See https://wiki.debian.org/Firmware for information about missing firmware bluetooth hci0: Direct firmware load for rtl_bt/rtl8822cu_config.bin failed with error -2 Bluetooth: hci0: RTL: cfg_sz -2, total sz 34332 Bluetooth: BNEP (Ethernet Emulation) ver 1.3 Bluetooth: BNEP filters: protocol multicast Bluetooth: BNEP socket layer initialized r8169 0000:10:00.0 eno1: Link is Down Bluetooth: hci0: RTL: fw version 0x099a281a sysrq: Keyboard mode set to system default sysrq: Emergency Sync Emergency Sync complete sysrq: Emergency Remount R/O

5 32

[PATCH v4 0/7] ASoC: codecs: add support for LPASS Codec TX and RX macros
by Srinivas Kandagatla 10 Feb '21

10 Feb '21

This patchset adds support for two Codec Macro blocks(TX and RX) available in Qualcomm LPASS (Low Power Audio SubSystem). There are WSA, VA, TX and RX Macros on LPASS IP, each of the Macro block has specific connectivity like WSA Macros are intended to connect to WSA Smart speaker codecs via SoundWire. VA Macro is intended for DMICs, and TX/RX for Analog codecs via SoundWire like other WCD938x Codecs to provide headphone/ear/lineout/amic/dmic etc .. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. This patchset has been tested on support to SM8250 MTP Development Board. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively and WCD938x TX and RX connected via Soundwire via TX and RX Macros reseptively. Thanks, srini Changes since v3: - fixed two minor warnings reported by kernel test robot using clang Srinivas Kandagatla (7): ASoC: qcom: dt-bindings: add bindings for lpass rx macro codec ASoC: codecs: lpass-rx-macro: add support for lpass rx macro ASoC: codecs: lpass-rx-macro: add dapm widgets and route ASoC: codecs: lpass-rx-macro: add iir widgets ASoC: qcom: dt-bindings: add bindings for lpass tx macro codec ASoC: codecs: lpass-tx-macro: add support for lpass tx macro ASoC: codecs: lpass-tx-macro: add dapm widgets and route .../bindings/sound/qcom,lpass-rx-macro.yaml | 62 + .../bindings/sound/qcom,lpass-tx-macro.yaml | 67 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/lpass-rx-macro.c | 3606 +++++++++++++++++ sound/soc/codecs/lpass-tx-macro.c | 1874 +++++++++ 6 files changed, 5623 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml create mode 100644 sound/soc/codecs/lpass-rx-macro.c create mode 100644 sound/soc/codecs/lpass-tx-macro.c -- 2.21.0

3 10

[PATCH] ASoC: soc-pcm: change error message to debug message
by Shengjiu Wang 10 Feb '21

10 Feb '21

This log message should be a debug message, because it doesn't return directly but continue next loop. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 605acec48971..cd9e919d7b99 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1344,8 +1344,8 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, /* is there a valid BE rtd for this widget */ be = dpcm_get_be(card, widget, stream); if (!be) { - dev_err(fe->dev, "ASoC: no BE found for %s\n", - widget->name); + dev_dbg(fe->dev, "ASoC: no BE found for %s\n", + widget->name); continue; } -- 2.27.0

4 5

[PATCH] ASoC: SOF: sof-pci-dev: add missing Up-Extreme quirk
by Pierre-Louis Bossart 10 Feb '21

10 Feb '21

The UpExtreme board supports the community key and was missed in previous contributions. Add it to make sure the open firmware is picked by default without needing a symlink on the target. Fixes: 46207ca24545 ('ASoC: SOF: pci: change the default firmware path when the community key is used') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/sof-pci-dev.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 84990cc8aa7d..fd1f0d8c2853 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -65,6 +65,13 @@ static const struct dmi_system_id community_key_platforms[] = { DMI_MATCH(DMI_BOARD_NAME, "UP-APL01"), } }, + { + .ident = "Up Extreme", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + DMI_MATCH(DMI_BOARD_NAME, "UP-WHL01"), + } + }, { .ident = "Google Chromebooks", .matches = { -- 2.25.1

2 1

[PATCH 0/5] ASoC: SOF: cleanups
by Pierre-Louis Bossart 10 Feb '21

10 Feb '21

Minor cleanups for error formats, missing cases, useless functions and simplifications. Curtis Malainey (2): ASoC: SOF: add missing pm debug ASoC: SOF: fix string format for errors Fred Oh (1): ASoC: SOF: ext_manifest: use explicit number for elem_type Guennadi Liakhovetski (2): ASoC: SOF: remove unused functions ASoC: SOF: HDA: (cosmetic) simplify hda_dsp_d0i3_work() include/sound/sof/ext_manifest.h | 6 ++-- sound/soc/sof/intel/hda-compress.c | 2 +- sound/soc/sof/intel/hda-dsp.c | 18 +++++------ sound/soc/sof/intel/hda-loader.c | 6 ++-- sound/soc/sof/intel/hda-pcm.c | 2 +- sound/soc/sof/intel/hda-trace.c | 4 +-- sound/soc/sof/intel/hda.c | 50 ------------------------------ sound/soc/sof/intel/hda.h | 1 - sound/soc/sof/ipc.c | 2 ++ 9 files changed, 19 insertions(+), 72 deletions(-) -- 2.25.1

2 6

[PATCH v2 00/11] ASoC: SOF/Intel/SoundWire: add missing quirks and DMIC support
by Pierre-Louis Bossart 10 Feb '21

10 Feb '21

HP Spectre x360 convertible devices rely on a mixed SoundWire+DMIC configuration which wasn't well supported. This lead to the discovery that we missed the TGL_HDMI quirk on a number of Dell devices, the addition of DMIC autodetection (based on NHLT tables), the addition of new component strings needed by UCM, and work-arounds due to problematic DSDT tables. Changes since v1: Rebase to remove first three patches already merged. Fix allmodconfig issues (undeclared kernel parameter) Bard Liao (1): ASoC: intel: sof_sdw: add trace for dai links Guennadi Liakhovetski (1): ASoC: SOF: Intel: HDA: don't keep a temporary variable Pierre-Louis Bossart (8): ASoC: Intel: sof_sdw: reorganize quirks by generation ASoC: Intel: sof-sdw: indent and add quirks consistently ASoC: Intel: sof_sdw: add quirk for HP Spectre x360 convertible ASoC: Intel: sof_sdw: add mic:dmic and cfg-mics component strings ASoC: Intel: soc-acpi: add ACPI matching table for HP Spectre x360 ASoC: SOF: Intel: SoundWire: refine ACPI match ASoC: SOF: Intel: detect DMIC number in SoundWire mixed config ASoC: SOF: Intel: hda: add dev_dbg() when DMIC number is overridden Rander Wang (1): ASoC: Intel: sof_sdw: detect DMIC number based on mach params sound/soc/intel/boards/sof_sdw.c | 134 ++++++----- .../intel/common/soc-acpi-intel-tgl-match.c | 20 ++ sound/soc/sof/intel/hda.c | 212 ++++++++++++------ 3 files changed, 244 insertions(+), 122 deletions(-) -- 2.25.1

2 12

[RFC PATCH 0/2] ASoC: dapm/pins: handle component prefix
by Pierre-Louis Bossart 10 Feb '21

10 Feb '21

SoundWire machine drivers make a heavy use of component prefixes to uniquify control names. This however results in errors when looking for widgets or pins. This patchset suggests two solutions but feedback or suggestions on how to take the prefix into account would be welcome. Bard Liao (1): ASoC: Intel: boards: max98373: get dapm from cpu_dai Shuming Fan (1): ASoC: dapm: use component prefix when checking widget names sound/soc/intel/boards/sof_maxim_common.c | 5 +++-- sound/soc/soc-dapm.c | 13 ++++++++++++- 2 files changed, 15 insertions(+), 3 deletions(-) -- 2.25.1

2 3

[PATCH] ASoC: SOF: fix runtime pm usage mismatch after probe errors
by Kai Vehmanen 10 Feb '21

10 Feb '21

With current delayed probe implementation, sof_probe_complete is not called in case of errors. And as this function is responsible for decrementing runtime pm usage counter, this will result in following problem: - probe driver in conditions where probe will fail (to force the condition on Intel SOF systems, set "snd_sof_intel_hda_common.codec_mask=0") - unload driver (runtime-pm usage_count is leaked) - fix the issue by installing missing fw, modifying module parameters, etc actions - try to load driver again -> success, probe ok -> device never enters runtime suspend Fix the issue by storing result of delayed probe to a state variable and providing new snd_sof_device_probe_completed() to be queried from SOF PCI/ACPI/OF drivers. If probe never completed successfully, runtime PM was not set up and thus at remove(), we should not increment usage count anymore. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)intel.com> --- sound/soc/sof/core.c | 10 ++++++++++ sound/soc/sof/sof-pci-dev.c | 3 ++- sound/soc/sof/sof-priv.h | 2 ++ 3 files changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 8d13eb13fe08..6d8f7d9fd192 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -246,6 +246,8 @@ static int sof_probe_continue(struct snd_sof_dev *sdev) if (plat_data->sof_probe_complete) plat_data->sof_probe_complete(sdev->dev); + sdev->probe_completed = true; + return 0; fw_trace_err: @@ -340,6 +342,14 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) } EXPORT_SYMBOL(snd_sof_device_probe); +bool snd_sof_device_probe_completed(struct device *dev) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + + return sdev->probe_completed; +} +EXPORT_SYMBOL(snd_sof_device_probe_completed); + int snd_sof_device_remove(struct device *dev) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 388772f9c4d2..84990cc8aa7d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -447,7 +447,8 @@ static void sof_pci_remove(struct pci_dev *pci) snd_sof_device_remove(&pci->dev); /* follow recommendation in pci-driver.c to increment usage counter */ - if (!(sof_pci_debug & SOF_PCI_DISABLE_PM_RUNTIME)) + if (snd_sof_device_probe_completed(&pci->dev) && + !(sof_pci_debug & SOF_PCI_DISABLE_PM_RUNTIME)) pm_runtime_get_noresume(&pci->dev); /* release pci regions and disable device */ diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 682c4b6d01ef..ad0d7ba2708c 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -389,6 +389,7 @@ struct snd_sof_dev { /* work queue in case the probe is implemented in two steps */ struct work_struct probe_work; + bool probe_completed; /* DSP HW differentiation */ struct snd_sof_pdata *pdata; @@ -464,6 +465,7 @@ struct snd_sof_dev { int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data); int snd_sof_device_remove(struct device *dev); int snd_sof_device_shutdown(struct device *dev); +bool snd_sof_device_probe_completed(struct device *dev); int snd_sof_runtime_suspend(struct device *dev); int snd_sof_runtime_resume(struct device *dev); base-commit: d40dac7ae8c0f213ac1da7896c35ddc2c58419ab -- 2.29.2

2 1

[PATCH v2 0/3] ASoC: mediatek: mt8192: apply some cleanup
by Tzung-Bi Shih 10 Feb '21

10 Feb '21

The series applies after [1]. The 1st and 2nd patch refactor to use asoc_substream_to_rtd(). Just realized there is such a helper for the purpose. The 3rd patch simplies ops of Capture1 DAI link. Changes from previous version: - Separated from the series [2]. [1]: https://patchwork.kernel.org/project/alsa-devel/patch/20210125061453.105653… [2]: https://patchwork.kernel.org/project/alsa-devel/list/?series=419769 Tzung-Bi Shih (3): ASoC: mediatek: mt8192-mt6359: use asoc_substream_to_rtd() ASoC: mediatek: mt8192: use asoc_substream_to_rtd() ASoC: mediatek: mt8192-mt6359: simply ops for Capture1 DAI link sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 4 ++-- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 24 ++++++++----------- 2 files changed, 12 insertions(+), 16 deletions(-) -- 2.30.0.280.ga3ce27912f-goog

2 4

[PATCH 00/14] ASoC: SOF/Intel/SoundWire: add missing quirks and DMIC support
by Pierre-Louis Bossart 10 Feb '21

10 Feb '21

HP Spectre x360 convertible devices rely on a mixed SoundWire+DMIC configuration which wasn't well supported. This lead to the discovery that we missed the TGL_HDMI quirk on a number of Dell devices, the addition of DMIC autodetection (based on NHLT tables), the addition of new component strings needed by UCM, and work-arounds due to problematic DSDT tables. Bard Liao (1): ASoC: Intel: sof_sdw: add trace for dai links Guennadi Liakhovetski (1): ASoC: SOF: Intel: HDA: don't keep a temporary variable Pierre-Louis Bossart (11): ASoC: Intel: sof_sdw: add missing TGL_HDMI quirk for Dell SKU 0A3E ASoC: Intel: sof_sdw: add missing TGL_HDMI quirk for Dell SKU 0A5E ASoC: Intel: sof_sdw: add missing TGL_HDMI quirk for Dell SKU 0A32 ASoC: Intel: sof_sdw: reorganize quirks by generation ASoC: Intel: sof-sdw: indent and add quirks consistently ASoC: Intel: sof_sdw: add quirk for HP Spectre x360 convertible ASoC: Intel: sof_sdw: add mic:dmic and cfg-mics component strings ASoC: Intel: soc-acpi: add ACPI matching table for HP Spectre x360 ASoC: SOF: Intel: SoundWire: refine ACPI match ASoC: SOF: Intel: detect DMIC number in SoundWire mixed config ASoC: SOF: Intel: hda: add dev_dbg() when DMIC number is overridden Rander Wang (1): ASoC: Intel: sof_sdw: detect DMIC number based on mach params sound/soc/intel/boards/sof_sdw.c | 131 ++++++----- .../intel/common/soc-acpi-intel-tgl-match.c | 20 ++ sound/soc/sof/intel/hda.c | 208 ++++++++++++------ 3 files changed, 241 insertions(+), 118 deletions(-) -- 2.25.1

2 21

[PATCH] ALSA: hda: intel-dsp-config: add Alder Lake support
by Kai Vehmanen 10 Feb '21

10 Feb '21

Add rules to select SOF driver for Alder Lake systems if a digital microphone or SoundWire codecs are present in the system. This is following same rules as for older Tiger Lake systems. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Xiuli Pan <xiuli.pan(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) NOTES: - PCI id 0x51c8 is not yet in tiwai/master, but it's in Mark's tree added with patch "ASoC: SOF: Intel: initial support to AlderLake-P" diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 68bb977c6a37..d1eb9d34993a 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -321,6 +321,18 @@ static const struct config_entry config_table[] = { }, #endif +/* Alder Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ALDERLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x7ad0, + }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x51c8, + }, +#endif + }; static const struct config_entry *snd_intel_dsp_find_config base-commit: fbd0a1d723513f82c9521c363e93967618517bb8 -- 2.29.2

2 1

[PATCH] [RFC] ALSA: control - add generic LED API to sound core routines
by Jaroslav Kysela 10 Feb '21

10 Feb '21

[DO NOT MERGE!] [This is just an early proposal for comments] [The code is not tested / debugged] The recent laptops have usually two LEDs assigned to reflect the speaker and microphone mute state. This implementation adds a tiny layer on top of the control API which calculates the state for those LEDs using the driver callbacks. Two new access flags are introduced to describe the controls which affects the audio path settings (an easy path for drivers). The LED resource can be shared with multiple sound cards with this code. The user space controls may be added to the state chain, too. This code should replace the LED code in the HDA driver and add a possibility to easy extend the other drivers (ASoC codecs etc.). Note: The snd_ctl_notify_one() changes should be moved to a separate patch. I will do this, when this proposal is agreed. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Perry Yuan <Perry.Yuan(a)dell.com> Cc: Hans de Goede <hdegoede(a)redhat.com> --- include/sound/control.h | 12 +++ include/uapi/sound/asound.h | 2 + sound/core/Kconfig | 6 ++ sound/core/Makefile | 1 + sound/core/control.c | 83 ++++++++++++++------- sound/core/control_led.c | 144 ++++++++++++++++++++++++++++++++++++ sound/pci/hda/Kconfig | 3 +- 7 files changed, 225 insertions(+), 26 deletions(-) create mode 100644 sound/core/control_led.c diff --git a/include/sound/control.h b/include/sound/control.h index 77d9fa10812d..ed967b18ffc9 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -261,4 +261,16 @@ snd_kctl_jack_new(const char *name, struct snd_card *card); void snd_kctl_jack_report(struct snd_card *card, struct snd_kcontrol *kctl, bool status); +#if IS_ENABLED(CONFIG_SND_LED) +void snd_ctl_led_notify(struct snd_card *card, unsigned int mask, + struct snd_kcontrol *kctl, unsigned int ioff); +void snd_ctl_led_register(struct snd_card *card); +void snd_ctl_led_disconnect(struct snd_card *card); +#else +static inline void snd_ctl_led_notify(struct snd_card *card, unsigned int mask, + struct snd_kcontrol *kctl, unsigned int ioff) {} +static inline void snd_ctl_led_register(struct snd_card *card) {} +static inline void snd_ctl_led_disconnect(struct snd_card *card) {} +#endif + #endif /* __SOUND_CONTROL_H */ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 535a7229e1d9..3faacdf5eaea 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -982,6 +982,8 @@ typedef int __bitwise snd_ctl_elem_iface_t; #define SNDRV_CTL_ELEM_ACCESS_INACTIVE (1<<8) /* control does actually nothing, but may be updated */ #define SNDRV_CTL_ELEM_ACCESS_LOCK (1<<9) /* write lock */ #define SNDRV_CTL_ELEM_ACCESS_OWNER (1<<10) /* write lock owner */ +#define SNDRV_CTL_ELEM_ACCESS_SPK_LED (1<<11) /* speaker (output) LED flag */ +#define SNDRV_CTL_ELEM_ACCESS_MIC_LED (1<<12) /* microphone (input) LED flag */ #define SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK (1<<28) /* kernel use a TLV callback */ #define SNDRV_CTL_ELEM_ACCESS_USER (1<<29) /* user space element */ /* bits 30 and 31 are obsoleted (for indirect access) */ diff --git a/sound/core/Kconfig b/sound/core/Kconfig index a4050f87f230..1bde494fa1fe 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -203,4 +203,10 @@ config SND_DMA_SGBUF def_bool y depends on X86 +config SND_LED + bool + select NEW_LEDS if SND_LED + select LEDS_TRIGGERS if SND_LED + select LEDS_TRIGGER_AUDIO if SND_LED + source "sound/core/seq/Kconfig" diff --git a/sound/core/Makefile b/sound/core/Makefile index ee4a4a6b99ba..81b33877a03d 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -13,6 +13,7 @@ snd-$(CONFIG_ISA_DMA_API) += isadma.o snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o +snd-$(CONFIG_SND_LED) += control_led.o snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_misc.o \ pcm_memory.o memalloc.o diff --git a/sound/core/control.c b/sound/core/control.c index 5165741a8400..3171e7f2798e 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -181,6 +181,28 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, } EXPORT_SYMBOL(snd_ctl_notify); +/** + * snd_ctl_notify_one - Send notification to user-space for a control change + * @card: the card to send notification + * @mask: the event mask, SNDRV_CTL_EVENT_* + * @kctl: the pointer with the control instance + * @ioff: the additional offset to the control index + * + * This function calls snd_ctl_notify() and does additional jobs + * like LED state changes. + */ +void snd_ctl_notify_one(struct snd_card *card, unsigned int mask, + struct snd_kcontrol *kctl, unsigned int ioff) +{ + struct snd_ctl_elem_id id = kctl->id; + + id.index += ioff; + id.numid += ioff; + snd_ctl_led_notify(card, mask, kctl, ioff); + snd_ctl_notify(card, mask, &id); +} +EXPORT_SYMBOL(snd_ctl_notify_one); + /** * snd_ctl_new - create a new control instance with some elements * @kctl: the pointer to store new control instance @@ -250,6 +272,8 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK | + SNDRV_CTL_ELEM_ACCESS_SPK_LED | + SNDRV_CTL_ELEM_ACCESS_MIC_LED | SNDRV_CTL_ELEM_ACCESS_SKIP_CHECK); err = snd_ctl_new(&kctl, count, access, NULL); @@ -342,7 +366,6 @@ static int __snd_ctl_add_replace(struct snd_card *card, { struct snd_ctl_elem_id id; unsigned int idx; - unsigned int count; struct snd_kcontrol *old; int err; @@ -376,10 +399,8 @@ static int __snd_ctl_add_replace(struct snd_card *card, kcontrol->id.numid = card->last_numid + 1; card->last_numid += kcontrol->count; - id = kcontrol->id; - count = kcontrol->count; - for (idx = 0; idx < count; idx++, id.index++, id.numid++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id); + for (idx = 0; idx < kcontrol->count; idx++) + snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_ADD, kcontrol, idx); return 0; } @@ -462,16 +483,15 @@ EXPORT_SYMBOL(snd_ctl_replace); */ int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol) { - struct snd_ctl_elem_id id; unsigned int idx; if (snd_BUG_ON(!card || !kcontrol)) return -EINVAL; list_del(&kcontrol->list); card->controls_count -= kcontrol->count; - id = kcontrol->id; - for (idx = 0; idx < kcontrol->count; idx++, id.index++, id.numid++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_REMOVE, &id); + + for (idx = 0; idx < kcontrol->count; idx++) + snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); snd_ctl_free_one(kcontrol); return 0; } @@ -584,11 +604,13 @@ int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, vd->access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; } snd_ctl_build_ioff(id, kctl, index_offset); - ret = 1; + downgrade_write(&card->controls_rwsem); + snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_INFO, kctl, index_offset); + up_read(&card->controls_rwsem); + return 1; + unlock: up_write(&card->controls_rwsem); - if (ret > 0) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, id); return ret; } EXPORT_SYMBOL_GPL(snd_ctl_activate_id); @@ -1110,25 +1132,34 @@ static int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, unsigned int index_offset; int result; + down_write(&card->controls_rwsem); kctl = snd_ctl_find_id(card, &control->id); - if (kctl == NULL) + if (kctl == NULL) { + up_write(&card->controls_rwsem); return -ENOENT; + } index_offset = snd_ctl_get_ioff(kctl, &control->id); vd = &kctl->vd[index_offset]; if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_WRITE) || kctl->put == NULL || (file && vd->owner && vd->owner != file)) { + up_write(&card->controls_rwsem); return -EPERM; } snd_ctl_build_ioff(&control->id, kctl, index_offset); result = kctl->put(kctl, control); - if (result < 0) + if (result < 0) { + up_write(&card->controls_rwsem); return result; + } if (result > 0) { - struct snd_ctl_elem_id id = control->id; - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &id); + downgrade_write(&card->controls_rwsem); + snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_VALUE, kctl, index_offset); + up_read(&card->controls_rwsem); + } else { + up_write(&card->controls_rwsem); } return 0; @@ -1150,9 +1181,7 @@ static int snd_ctl_elem_write_user(struct snd_ctl_file *file, if (result < 0) goto error; - down_write(&card->controls_rwsem); result = snd_ctl_elem_write(card, file, control); - up_write(&card->controls_rwsem); if (result < 0) goto error; @@ -1301,7 +1330,6 @@ static int replace_user_tlv(struct snd_kcontrol *kctl, unsigned int __user *buf, { struct user_element *ue = kctl->private_data; unsigned int *container; - struct snd_ctl_elem_id id; unsigned int mask = 0; int i; int change; @@ -1333,10 +1361,8 @@ static int replace_user_tlv(struct snd_kcontrol *kctl, unsigned int __user *buf, ue->tlv_data_size = size; mask |= SNDRV_CTL_EVENT_MASK_TLV; - for (i = 0; i < kctl->count; ++i) { - snd_ctl_build_ioff(&id, kctl, i); - snd_ctl_notify(ue->card, mask, &id); - } + for (i = 0; i < kctl->count; ++i) + snd_ctl_notify_one(ue->card, mask, kctl, i); return change; } @@ -1998,9 +2024,14 @@ static const struct file_operations snd_ctl_f_ops = static int snd_ctl_dev_register(struct snd_device *device) { struct snd_card *card = device->device_data; + int err; - return snd_register_device(SNDRV_DEVICE_TYPE_CONTROL, card, -1, - &snd_ctl_f_ops, card, &card->ctl_dev); + err = snd_register_device(SNDRV_DEVICE_TYPE_CONTROL, card, -1, + &snd_ctl_f_ops, card, &card->ctl_dev); + if (err < 0) + return err; + snd_ctl_led_register(card); + return 0; } /* @@ -2019,6 +2050,8 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); + snd_ctl_led_disconnect(card); + return snd_unregister_device(&card->ctl_dev); } diff --git a/sound/core/control_led.c b/sound/core/control_led.c new file mode 100644 index 000000000000..e70855ea54d1 --- /dev/null +++ b/sound/core/control_led.c @@ -0,0 +1,144 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * LED state routines for driver control interface + * Copyright (c) 2021 by Jaroslav Kysela <perex(a)perex.cz> + */ + +#include <linux/leds.h> +#include <sound/core.h> +#include <sound/control.h> + +static DEFINE_MUTEX(snd_control_led_mutex); +static unsigned int snd_control_led_last_card; +static struct snd_card *snd_control_led_cards[SNDRV_CARDS]; + +#define UPDATE_ROUTE(route, cb) \ + do { \ + int route2 = (cb); \ + if (route2 >= 0) \ + route = route < 0 ? route2 : (route | route2); \ + } while (0) + +static int snd_ctl_led_get(struct snd_card *card, + struct snd_kcontrol *kctl, unsigned int ioff) +{ + struct snd_ctl_elem_info info; + struct snd_ctl_elem_value value; + unsigned int i; + int result; + + memset(&info, 0, sizeof(info)); + info.id = kctl->id; + info.id.index += ioff; + info.id.numid += ioff; + result = kctl->info(kctl, &info); + if (result < 0) + return -1; + memset(&value, 0, sizeof(value)); + value.id = info.id; + result = kctl->get(kctl, &value); + if (result < 0) + return -1; + if (info.type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || + info.type == SNDRV_CTL_ELEM_TYPE_INTEGER) { + for (i = 0; i < info.count; i++) + if (value.value.integer.value[i] != info.value.integer.min) + return 1; + } else if (info.type == SNDRV_CTL_ELEM_TYPE_INTEGER64) { + for (i = 0; i < info.count; i++) + if (value.value.integer64.value[i] != info.value.integer64.min) + return 1; + } + return 0; +} + +static int snd_ctl_led_set_state_for_card(int card_number, unsigned int group) +{ + struct snd_card *card = snd_control_led_cards[card_number]; + struct snd_kcontrol *kctl; + struct snd_kcontrol_volatile *vd; + unsigned int ioff; + int route = -1; + + down_read(&card->controls_rwsem); + list_for_each_entry(kctl, &card->controls, list) { + for (ioff = 0; ioff < kctl->count; ioff++) { + vd = &kctl->vd[ioff]; + if (vd->access & group) + UPDATE_ROUTE(route, snd_ctl_led_get(card, kctl, ioff)); + } + } + up_read(&card->controls_rwsem); + return route; +} + +static void snd_ctl_led_set_state(struct snd_card *card, unsigned int group) +{ + int card_number; + enum led_audio led_trigger_type; + int route; + + if (group == SNDRV_CTL_ELEM_ACCESS_SPK_LED) { + led_trigger_type = LED_AUDIO_MUTE; + } else if (group == SNDRV_CTL_ELEM_ACCESS_MIC_LED) { + led_trigger_type = LED_AUDIO_MICMUTE; + } else { + return; + } + route = -1; + mutex_lock(&snd_control_led_mutex); + /* the card may not be registered (active) at this point */ + if (snd_control_led_cards[card->number] == NULL) { + mutex_unlock(&snd_control_led_mutex); + return; + } + for (card_number = 0; card_number <= snd_control_led_last_card; card_number++) + UPDATE_ROUTE(route, snd_ctl_led_set_state_for_card(card_number, group)); + mutex_unlock(&snd_control_led_mutex); + if (route >= 0) + ledtrig_audio_set(led_trigger_type, route ? LED_OFF : LED_ON); + +} + +void snd_ctl_led_notify(struct snd_card *card, unsigned int mask, + struct snd_kcontrol *kctl, unsigned int ioff) +{ + if (mask == SNDRV_CTL_EVENT_MASK_REMOVE || + (mask & (SNDRV_CTL_EVENT_MASK_INFO | + SNDRV_CTL_EVENT_MASK_ADD | + SNDRV_CTL_EVENT_MASK_VALUE)) != 0) { + struct snd_kcontrol_volatile *vd = &kctl->vd[ioff]; + if (vd->access & SNDRV_CTL_ELEM_ACCESS_SPK_LED) + snd_ctl_led_set_state(card, SNDRV_CTL_ELEM_ACCESS_SPK_LED); + if (vd->access & SNDRV_CTL_ELEM_ACCESS_MIC_LED) + snd_ctl_led_set_state(card, SNDRV_CTL_ELEM_ACCESS_MIC_LED); + } +} + +void snd_ctl_led_register(struct snd_card *card) +{ + if (snd_BUG_ON(card->number < 0 || + card->number >= ARRAY_SIZE(snd_control_led_cards))) + return; + mutex_lock(&snd_control_led_mutex); + snd_control_led_cards[card->number] = card; + if (card->number > snd_control_led_last_card) + snd_control_led_last_card = card->number; + mutex_unlock(&snd_control_led_mutex); + snd_ctl_led_set_state(card, SNDRV_CTL_ELEM_ACCESS_SPK_LED); + snd_ctl_led_set_state(card, SNDRV_CTL_ELEM_ACCESS_MIC_LED); +} + +void snd_ctl_led_disconnect(struct snd_card *card) +{ + mutex_lock(&snd_control_led_mutex); + if (snd_control_led_last_card == card->number) { + while (snd_control_led_last_card > 0) + if (snd_control_led_cards[--snd_control_led_last_card]) + break; + } + snd_control_led_cards[card->number] = NULL; + mutex_unlock(&snd_control_led_mutex); + snd_ctl_led_set_state(card, SNDRV_CTL_ELEM_ACCESS_SPK_LED); + snd_ctl_led_set_state(card, SNDRV_CTL_ELEM_ACCESS_MIC_LED); +} diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 90759391cbac..ccf788ede8ba 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -9,7 +9,8 @@ config SND_HDA select SND_HDA_CORE config SND_HDA_GENERIC_LEDS - bool + bool + select SND_LED # just for test, ignore, please! config SND_HDA_INTEL tristate "HD Audio PCI" -- 2.29.2

5 9

[PATCH] ALSA: core - add missing compress device type to /proc/asound/devices
by Jaroslav Kysela 10 Feb '21

10 Feb '21

Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Vinod Koul <vkoul(a)kernel.org> --- sound/core/sound.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/sound.c b/sound/core/sound.c index b75f78f2c4b8..302b6d12ab2b 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -337,6 +337,8 @@ static const char *snd_device_type_name(int type) return "sequencer"; case SNDRV_DEVICE_TYPE_TIMER: return "timer"; + case SNDRV_DEVICE_TYPE_COMPRESS: + return "compress"; default: return "?"; } -- 2.29.2

3 2

[PATCH v2 0/9] Tegra186 and Tegra194 audio graph card
by Sameer Pujar 10 Feb '21

10 Feb '21

This series adds support for audio graph based solution on Tegra186 and Tegra194. This enables audio paths for I2S, DMIC and DSPK modules. Depending on the platform Jetson TX2 or Jetson AGX Xavier, required I/O module instances are enabled. Since the latter board has on board audio codec, DT support for the same is enabled and external audio playback and capture can be used. Changelog ========= v1 --> v2: ---------- - Dropped patch "ASoC: tegra: Select SND_SOC_RT5659" as per suggestion from Mark. - Add new patch "ASoC: rt5659: Add Kconfig prompt" - Add new patch "arm64: defconfig: Enable RT5659" - No changes in other patches from earlier series. Sameer Pujar (9): ASoC: dt-bindings: rt5659: Update binding doc ASoC: dt-bindings: tegra: Add iommus property to Tegra graph card ASoC: audio-graph-card: Add clocks property to endpoint node ASoC: rt5659: Add Kconfig prompt arm64: defconfig: Enable RT5659 arm64: tegra: Add RT5658 device entry Revert "arm64: tegra: Disable the ACONNECT for Jetson TX2" arm64: tegra: Audio graph sound card for Jetson TX2 arm64: tegra: Audio graph sound card for Jetson AGX Xavier .../bindings/sound/audio-graph-port.yaml | 3 + .../sound/nvidia,tegra-audio-graph-card.yaml | 3 + Documentation/devicetree/bindings/sound/rt5659.txt | 11 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 609 +++++++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra186.dtsi | 22 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 468 ++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra194.dtsi | 20 + arch/arm64/configs/defconfig | 1 + sound/soc/codecs/Kconfig | 2 +- 9 files changed, 1138 insertions(+), 1 deletion(-) -- 2.7.4

5 17

[PATCH v2 0/7] ASoC: codecs: add support for LPASS Codec TX and RX macros
by Srinivas Kandagatla 10 Feb '21

10 Feb '21

Thanks for reviewing v1, here is v2 patchset addressing comments from v1. This patchset adds support for two Codec Macro blocks(TX and RX) available in Qualcomm LPASS (Low Power Audio SubSystem). There are WSA, VA, TX and RX Macros on LPASS IP, each of the Macro block has specific connectivity like WSA Macros are intended to connect to WSA Smart speaker codecs via SoundWire. VA Macro is intended for DMICs, and TX/RX for Analog codecs via SoundWire like other WCD938x Codecs to provide headphone/ear/lineout/amic/dmic etc .. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. This patchset has been tested on support to SM8250 MTP Development Board. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively and WCD938x TX and RX connected via Soundwire via TX and RX Macros reseptively. Thanks, srini Changes since v1 - make copyright header line with C++ one - correct usage of regcache! - moved ON/OFF enums to proper switch controls - Align all the control naming! - inline swrm_clock function. Srinivas Kandagatla (7): ASoC: qcom: dt-bindings: add bindings for lpass rx macro codec ASoC: codec: lpass-rx-macro: add support for lpass rx macro ASoC: codec: lpass-rx-macro: add dapm widgets and route ASoC: codec: lpass-rx-macro: add iir widgets ASoC: qcom: dt-bindings: add bindings for lpass tx macro codec ASoC: codec: lpass-tx-macro: add support for lpass tx macro ASoC: codec: lpass-tx-macro: add dapm widgets and route .../bindings/sound/qcom,lpass-rx-macro.yaml | 62 + .../bindings/sound/qcom,lpass-tx-macro.yaml | 67 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/lpass-rx-macro.c | 3604 +++++++++++++++++ sound/soc/codecs/lpass-tx-macro.c | 1877 +++++++++ 6 files changed, 5624 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml create mode 100644 sound/soc/codecs/lpass-rx-macro.c create mode 100644 sound/soc/codecs/lpass-tx-macro.c -- 2.21.0

3 11

[PATCH v3 0/9] ALSA: add virtio sound driver
by Anton Yakovlev 09 Feb '21

09 Feb '21

This series implements a driver part of the virtio sound device specification v8 [1]. The driver supports PCM playback and capture substreams, jack and channel map controls. A message-based transport is used to write/read PCM frames to/from a device. As a device part was used OpenSynergy proprietary implementation. Any comments are very welcome. v3 changes: - Fixed license headers for all files. - Many coding style and kernel doc fixes. - Replaced devm_kmalloc/devm_kfree with kmalloc/kfree wherever appropriate. - Made the names of the card and PCM devices more informative. - To process the DEVICE_NEEDS_RESET status, simply call device_reprobe(). - For control messages replaced atomic_t by refcount_t for the reference counter and simplified the general logic. - Use vmalloc'ed managed buffer for PCM substreams. - Replaced all atomic fields in the virtio substream structure with non-atomic + spinlock. - Use the non-atomic PCM ops. - Use ops->sync_stop() to release the substream on the device side. - Rebased and tested on top of v5.11-rc6. v2 changes: - For some reason, in the previous patch series, several patches were squashed. Fixed this issue to make the review easier. - Added mst(a)redhat.com to the MAINTAINERS. - When creating virtqueues, now only the event virtqueue is disabled. It's enabled only after successful initialization of the device. - Added additional comments to the reset worker function: [2/9] virtio_card.c:virtsnd_reset_fn() - Added check that VIRTIO_F_VERSION_1 feature bit is set. - Added additional comments to the device removing function: [2/9] virtio_card.c:virtsnd_remove() - Added additional comments to the tx/rx interrupt handler: [5/9] virtio_pcm_msg.c:virtsnd_pcm_msg_complete() - Added additional comments to substream release wait function. [6/9] virtio_pcm_ops.c:virtsnd_pcm_released() [1] https://lists.oasis-open.org/archives/virtio-dev/202003/msg00185.html Anton Yakovlev (9): uapi: virtio_ids: add a sound device type ID from OASIS spec ALSA: virtio: add virtio sound driver ALSA: virtio: handling control messages ALSA: virtio: build PCM devices and substream hardware descriptors ALSA: virtio: handling control and I/O messages for the PCM device ALSA: virtio: PCM substream operators ALSA: virtio: introduce jack support ALSA: virtio: introduce PCM channel map support ALSA: virtio: introduce device suspend/resume support MAINTAINERS | 9 + include/uapi/linux/virtio_ids.h | 1 + include/uapi/linux/virtio_snd.h | 334 +++++++++++++++++++++ sound/Kconfig | 2 + sound/Makefile | 3 +- sound/virtio/Kconfig | 10 + sound/virtio/Makefile | 13 + sound/virtio/virtio_card.c | 462 +++++++++++++++++++++++++++++ sound/virtio/virtio_card.h | 113 ++++++++ sound/virtio/virtio_chmap.c | 219 ++++++++++++++ sound/virtio/virtio_ctl_msg.c | 311 ++++++++++++++++++++ sound/virtio/virtio_ctl_msg.h | 78 +++++ sound/virtio/virtio_jack.c | 233 +++++++++++++++ sound/virtio/virtio_pcm.c | 499 ++++++++++++++++++++++++++++++++ sound/virtio/virtio_pcm.h | 121 ++++++++ sound/virtio/virtio_pcm_msg.c | 393 +++++++++++++++++++++++++ sound/virtio/virtio_pcm_ops.c | 493 +++++++++++++++++++++++++++++++ 17 files changed, 3293 insertions(+), 1 deletion(-) create mode 100644 include/uapi/linux/virtio_snd.h create mode 100644 sound/virtio/Kconfig create mode 100644 sound/virtio/Makefile create mode 100644 sound/virtio/virtio_card.c create mode 100644 sound/virtio/virtio_card.h create mode 100644 sound/virtio/virtio_chmap.c create mode 100644 sound/virtio/virtio_ctl_msg.c create mode 100644 sound/virtio/virtio_ctl_msg.h create mode 100644 sound/virtio/virtio_jack.c create mode 100644 sound/virtio/virtio_pcm.c create mode 100644 sound/virtio/virtio_pcm.h create mode 100644 sound/virtio/virtio_pcm_msg.c create mode 100644 sound/virtio/virtio_pcm_ops.c -- 2.30.0

1 9

[PATCH v3 0/7] ASoC: codecs: add support for LPASS Codec TX and RX macros
by Srinivas Kandagatla 09 Feb '21

09 Feb '21

Thanks for reviewing v2, here is v3 patchset addressing comments from v2. This patchset adds support for two Codec Macro blocks(TX and RX) available in Qualcomm LPASS (Low Power Audio SubSystem). There are WSA, VA, TX and RX Macros on LPASS IP, each of the Macro block has specific connectivity like WSA Macros are intended to connect to WSA Smart speaker codecs via SoundWire. VA Macro is intended for DMICs, and TX/RX for Analog codecs via SoundWire like other WCD938x Codecs to provide headphone/ear/lineout/amic/dmic etc .. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. This patchset has been tested on support to SM8250 MTP Development Board. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively and WCD938x TX and RX connected via Soundwire via TX and RX Macros reseptively. Thanks, srini Changes since v2: - Suffix some of the simple on/off control names with "Switch" Srinivas Kandagatla (7): ASoC: qcom: dt-bindings: add bindings for lpass rx macro codec ASoC: codecs: lpass-rx-macro: add support for lpass rx macro ASoC: codecs: lpass-rx-macro: add dapm widgets and route ASoC: codecs: lpass-rx-macro: add iir widgets ASoC: qcom: dt-bindings: add bindings for lpass tx macro codec ASoC: codecs: lpass-tx-macro: add support for lpass tx macro ASoC: codecs: lpass-tx-macro: add dapm widgets and route .../bindings/sound/qcom,lpass-rx-macro.yaml | 62 + .../bindings/sound/qcom,lpass-tx-macro.yaml | 67 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/lpass-rx-macro.c | 3604 +++++++++++++++++ sound/soc/codecs/lpass-tx-macro.c | 1877 +++++++++ 6 files changed, 5624 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml create mode 100644 sound/soc/codecs/lpass-rx-macro.c create mode 100644 sound/soc/codecs/lpass-tx-macro.c -- 2.21.0

2 8

[PATCH v4 0/5] MFD/ASoC: Add support for Intel Bay Trail boards with WM5102 codec
by Hans de Goede 08 Feb '21

08 Feb '21

Hi All, Here is v4 of my series to add support for Intel Bay Trail based devices which use a WM5102 codec for audio output/input. This was developed and tested on a Lenovo Yoga Tablet 1051L. The MFD and ASoC parts do not have any build-time dependencies on each other. But the follow-up jack-detect series does have patches depending on each-other and on this series. So IMHO it would be best if this entire series would be merged through the MFD tree to make merging the follow-up series easier. Mark, if that is ok with you (and you are happy with the ASoC changes) can you please Ack this ? Changes in v4: - Add a comment to the irq-flags override explaining that theoretically DSDTs using IRQF_TRIGGER_FALLING could be correct on boards where the IRQ controller does not support active low level interrupts Changes in v3: - Fix compilation error when CONFIG_ACPI is not set Changes in v2: - Split my WM5102 work into 2 series, one series adding basic support for Bay Trail boards with a WM5102 codec and a second series with the jack-detect work - Various other minor code tweaks Hans de Goede (4): mfd: arizona: Add MODULE_SOFTDEP("pre: arizona_ldo1") mfd: arizona: Replace arizona_of_get_type() with device_get_match_data() mfd: arizona: Add support for ACPI enumeration of WM5102 connected over SPI ASoC: Intel: Add DMI quirk table to soc_intel_is_byt_cr() Pierre-Louis Bossart (1): ASoC: Intel: bytcr_wm5102: Add machine driver for BYT/WM5102 drivers/mfd/arizona-core.c | 11 - drivers/mfd/arizona-i2c.c | 11 +- drivers/mfd/arizona-spi.c | 138 +++++- drivers/mfd/arizona.h | 9 - sound/soc/intel/boards/Kconfig | 12 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcr_wm5102.c | 465 ++++++++++++++++++ .../intel/common/soc-acpi-intel-byt-match.c | 16 + sound/soc/intel/common/soc-intel-quirks.h | 25 + 9 files changed, 661 insertions(+), 28 deletions(-) create mode 100644 sound/soc/intel/boards/bytcr_wm5102.c Regards, Hans

4 34

[PATCH] ASoC: fsl: constify static snd_soc_dai_ops structs
by Rikard Falkeborn 08 Feb '21

08 Feb '21

The only usage of these is to assign their address to the 'ops' field in the snd_soc_dai_driver struct, which is a pointer to const. Make them const to allow the compiler to put them in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_micfil.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 60951a8aabd3..636a702f37a6 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -1530,7 +1530,7 @@ static int fsl_easrc_hw_free(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops fsl_easrc_dai_ops = { +static const struct snd_soc_dai_ops fsl_easrc_dai_ops = { .startup = fsl_easrc_startup, .trigger = fsl_easrc_trigger, .hw_params = fsl_easrc_hw_params, diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 8aedf6590b28..5935af2e5ff4 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -381,7 +381,7 @@ static int fsl_micfil_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, return ret; } -static struct snd_soc_dai_ops fsl_micfil_dai_ops = { +static const struct snd_soc_dai_ops fsl_micfil_dai_ops = { .startup = fsl_micfil_startup, .trigger = fsl_micfil_trigger, .hw_params = fsl_micfil_hw_params, diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index dd228b421e2c..6dd0a5fcd455 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -857,7 +857,7 @@ static struct snd_kcontrol_new fsl_xcvr_tx_ctls[] = { }, }; -static struct snd_soc_dai_ops fsl_xcvr_dai_ops = { +static const struct snd_soc_dai_ops fsl_xcvr_dai_ops = { .prepare = fsl_xcvr_prepare, .startup = fsl_xcvr_startup, .shutdown = fsl_xcvr_shutdown, -- 2.30.0

3 2

[PATCH v3 0/2] Rename audio graph export functions
by Sameer Pujar 08 Feb '21

08 Feb '21

This series renames exported functions from audio graph for a better global visibility. In doing so update the references in audio graph and Tegra audio graph card drivers. Changelog ========= v2 -> v3: --------- [Patch v3 1/2]: - Squashed "patch v2 1/3" and "patch v2 2/3". Updated commit message accordingly. - Add "Acked-by" tag from Morimoto-san. [Patch v3 2/2]: - Add "Acked-by" tag from Morimoto-san. v1 -> v2: --------- [Patch v2 1/3]: - Rename graph_parse_of() to audio_graph_parse_of() and graph_card_probe() to audio_graph_card_probe() as well. - Update above references in audio graph driver. [Patch v2 2/3]: New patch - Update references for audio_graph_parse_of() and audio_graph_card_probe() in Tegra graph driver. - Add "Depends-on" tag [Patch v2 3/3]: - Update commit message to add "Depends-on" tag. Sameer Pujar (2): ASoC: audio-graph: Rename functions needed for export ASoC: tegra: Add driver remove() callback include/sound/graph_card.h | 6 ++++-- sound/soc/generic/audio-graph-card.c | 17 +++++++++-------- sound/soc/tegra/tegra_audio_graph_card.c | 9 +++++---- 3 files changed, 18 insertions(+), 14 deletions(-) -- 2.7.4

2 3

[PATCH v2] ASoC: SOF: debug: Fix a potential issue on string buffer termination
by Hui Wang 08 Feb '21

08 Feb '21

The function simple_write_to_buffer() doesn't add string termination at the end of buf, we need to handle it on our own. This change refers to the function tokenize_input() in debug.c and the function sof_dfsentry_trace_filter_write() in trace.c. Fixes: 091c12e1f50c ("ASoC: SOF: debug: add new debugfs entries for IPC flood test") Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/sof/debug.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index 30213a1beaaa..715a374b33cf 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -352,7 +352,7 @@ static ssize_t sof_dfsentry_write(struct file *file, const char __user *buffer, char *string; int ret; - string = kzalloc(count, GFP_KERNEL); + string = kzalloc(count+1, GFP_KERNEL); if (!string) return -ENOMEM; -- 2.25.1

3 2

[PATCH 1/3] dt-bindings: sound/ingenic: Add compatible strings for JZ4760(B) SoC
by Paul Cercueil 08 Feb '21

08 Feb '21

Add the ingenic,jz4760b-codec and ingenic,jz4760-codec compatible strings. In the process, convert the previous compatible strings to use an enum instead. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> --- .../devicetree/bindings/sound/ingenic,codec.yaml | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/ingenic,codec.yaml b/Documentation/devicetree/bindings/sound/ingenic,codec.yaml index eb4be86464bb..97d5f3819b27 100644 --- a/Documentation/devicetree/bindings/sound/ingenic,codec.yaml +++ b/Documentation/devicetree/bindings/sound/ingenic,codec.yaml @@ -15,9 +15,14 @@ properties: compatible: oneOf: - - const: ingenic,jz4770-codec - - const: ingenic,jz4725b-codec - - const: ingenic,jz4740-codec + - enum: + - ingenic,jz4770-codec + - ingenic,jz4760-codec + - ingenic,jz4725b-codec + - ingenic,jz4740-codec + - items: + - const: ingenic,jz4760b-codec + - const: ingenic,jz4760-codec reg: maxItems: 1 -- 2.29.2

2 3

[PATCH RFC 0/4] Add HDMI support for Intel KeemBay I2S
by Sia Jee Heng 08 Feb '21

08 Feb '21

The below patch series are to support Audio over HDMI. The modification in this patch series shall allow I2S driver to playback standard PCM format and IEC958 encoded format to the ADV7511 HDMI chip. ALSA IEC958 plugin will be used to compose the IEC958 format. Existing hdmi-codec driver only support standard pcm format. Support of IEC958 encoded format passdown from ALSA IEC958 plugin is needed so that the IEC958 encoded data can be streamed to the HDMI chip. Sia Jee Heng (4): ASoC: codec: hdmi-codec: Support IEC958 encoded PCM format drm: bridge: adv7511: Support I2S IEC958 encoded PCM format dt-bindings: sound: Intel, Keembay-i2s: Add hdmi-i2s compatible string ASoC: Intel: KMB: Support IEC958 encoded PCM format .../bindings/sound/intel,keembay-i2s.yaml | 1 + drivers/gpu/drm/bridge/adv7511/adv7511.h | 1 + .../gpu/drm/bridge/adv7511/adv7511_audio.c | 6 ++ include/sound/hdmi-codec.h | 5 ++ sound/soc/codecs/hdmi-codec.c | 4 +- sound/soc/intel/keembay/kmb_platform.c | 73 ++++++++++++++++++- sound/soc/intel/keembay/kmb_platform.h | 1 + 7 files changed, 89 insertions(+), 2 deletions(-) base-commit: 2557c711b87cd42bb22be9ca6ff3fce038624f30 -- 2.18.0

3 7

[PATCH v2] ASoC: rt5682: Fix panic in rt5682_jack_detect_handler happening during system shutdown
by Ranjani Sridharan 08 Feb '21

08 Feb '21

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> During Coldboot stress tests, system encountered the following panic. Panic logs depicts rt5682_i2c_shutdown() happened first and then later jack detect handler workqueue function triggered. This situation causes panic as rt5682_i2c_shutdown() resets codec. Fix this panic by cancelling all jack detection delayed work. Panic log: [ 20.936124] sof_pci_shutdown [ 20.940248] snd_sof_device_shutdown [ 20.945023] snd_sof_shutdown [ 21.126849] rt5682_i2c_shutdown [ 21.286053] rt5682_jack_detect_handler [ 21.291235] BUG: kernel NULL pointer dereference, address: 000000000000037c [ 21.299302] #PF: supervisor read access in kernel mode [ 21.305254] #PF: error_code(0x0000) - not-present page [ 21.311218] PGD 0 P4D 0 [ 21.314155] Oops: 0000 [#1] PREEMPT SMP NOPTI [ 21.319206] CPU: 2 PID: 123 Comm: kworker/2:3 Tainted: G U 5.4.68 #10 [ 21.333687] ACPI: Preparing to enter system sleep state S5 [ 21.337669] Workqueue: events_power_efficient rt5682_jack_detect_handler [snd_soc_rt5682] [ 21.337671] RIP: 0010:rt5682_jack_detect_handler+0x6c/0x279 [snd_soc_rt5682] Fixes: a50067d4f3c1d ('ASoC: rt5682: split i2c driver into separate module') Signed-off-by: Jairaj Arava <jairaj.arava(a)intel.com> Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)intel.com> Reviewed-by: Shuming Fan <shumingf(a)realtek.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/codecs/rt5682-i2c.c | 3 +++ 1 file changed, 3 insertions(+) Changelog ========= v1 -> v2: cancel both jack detection work diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 37d13120f5ba..93c1603b42f1 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -273,6 +273,9 @@ static void rt5682_i2c_shutdown(struct i2c_client *client) { struct rt5682_priv *rt5682 = i2c_get_clientdata(client); + cancel_delayed_work_sync(&rt5682->jack_detect_work); + cancel_delayed_work_sync(&rt5682->jd_check_work); + rt5682_reset(rt5682); } -- 2.25.1

2 1

[PATCH] ASoC: fsl_esai: Remove unused 'imx' field
by Fabio Estevam 08 Feb '21

08 Feb '21

The 'imx' field is not used anywhere, so get rid of it. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/fsl/fsl_esai.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 614e22a95f53..ae679fa4aed9 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -27,7 +27,6 @@ * @reset_at_xrun: flags for enable reset operaton */ struct fsl_esai_soc_data { - bool imx; bool reset_at_xrun; }; @@ -86,17 +85,14 @@ struct fsl_esai { }; static struct fsl_esai_soc_data fsl_esai_vf610 = { - .imx = false, .reset_at_xrun = true, }; static struct fsl_esai_soc_data fsl_esai_imx35 = { - .imx = true, .reset_at_xrun = true, }; static struct fsl_esai_soc_data fsl_esai_imx6ull = { - .imx = true, .reset_at_xrun = false, }; -- 2.17.1

2 1

[PATCH v2] ASoC: fsl_esai: Remove unused 'imx' field
by Fabio Estevam 08 Feb '21

08 Feb '21

The 'imx' field is not used anywhere, so get rid of it. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Changes since v1: - Also remove the comment about the imx member. sound/soc/fsl/fsl_esai.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index bac9f059c9fc..08056fa0a0fa 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -23,11 +23,9 @@ /** * struct fsl_esai_soc_data - soc specific data - * @imx: for imx platform * @reset_at_xrun: flags for enable reset operaton */ struct fsl_esai_soc_data { - bool imx; bool reset_at_xrun; }; @@ -86,17 +84,14 @@ struct fsl_esai { }; static struct fsl_esai_soc_data fsl_esai_vf610 = { - .imx = false, .reset_at_xrun = true, }; static struct fsl_esai_soc_data fsl_esai_imx35 = { - .imx = true, .reset_at_xrun = true, }; static struct fsl_esai_soc_data fsl_esai_imx6ull = { - .imx = true, .reset_at_xrun = false, }; -- 2.25.1

3 2

[PATCH] ASoC: rt1316: Add RT1316 SDCA vendor-specific driver
by shumingf＠realtek.com 08 Feb '21

08 Feb '21

From: Shuming Fan <shumingf(a)realtek.com> This is the initial amplifier driver for rt1316 SDCA version. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt1316-sdw.c | 808 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1316-sdw.h | 57 +++ 4 files changed, 873 insertions(+) create mode 100644 sound/soc/codecs/rt1316-sdw.c create mode 100644 sound/soc/codecs/rt1316-sdw.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index df34b3b3bb14..0b9e9047377e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -181,6 +181,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT711_SDW imply SND_SOC_RT715_SDW imply SND_SOC_RT1308_SDW + imply SND_SOC_RT1316_SDW imply SND_SOC_SGTL5000 imply SND_SOC_SI476X imply SND_SOC_SIMPLE_AMPLIFIER @@ -1129,6 +1130,11 @@ config SND_SOC_RT1308_SDW depends on I2C && SOUNDWIRE select REGMAP_SOUNDWIRE +config SND_SOC_RT1316_SDW + tristate "Realtek RT1316 Codec - SDW" + depends on SOUNDWIRE + select REGMAP_SOUNDWIRE + config SND_SOC_RT5514 tristate depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 628b0c9b3e2a..ddf59532e9ca 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -170,6 +170,7 @@ snd-soc-rt1015p-objs := rt1015p.o snd-soc-rt1305-objs := rt1305.o snd-soc-rt1308-objs := rt1308.o snd-soc-rt1308-sdw-objs := rt1308-sdw.o +snd-soc-rt1316-sdw-objs := rt1316-sdw.o snd-soc-rt274-objs := rt274.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o @@ -483,6 +484,7 @@ obj-$(CONFIG_SND_SOC_RT1015P) += snd-soc-rt1015p.o obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o obj-$(CONFIG_SND_SOC_RT1308) += snd-soc-rt1308.o obj-$(CONFIG_SND_SOC_RT1308_SDW) += snd-soc-rt1308-sdw.o +obj-$(CONFIG_SND_SOC_RT1316_SDW) += snd-soc-rt1316-sdw.o obj-$(CONFIG_SND_SOC_RT274) += snd-soc-rt274.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c new file mode 100644 index 000000000000..a09ebb768667 --- /dev/null +++ b/sound/soc/codecs/rt1316-sdw.c @@ -0,0 +1,808 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt1316-sdw.c -- rt1316 SDCA ALSA SoC amplifier audio driver +// +// Copyright(c) 2020 Realtek Semiconductor Corp. +// +// +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/pm_runtime.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include "rt1316-sdw.h" + +static const struct reg_default rt1316_reg_defaults[] = { + { 0x3004, 0x00 }, + { 0x3005, 0x00 }, + { 0x3206, 0x00 }, + { 0xc001, 0x00 }, + { 0xc002, 0x00 }, + { 0xc003, 0x00 }, + { 0xc004, 0x00 }, + { 0xc005, 0x00 }, + { 0xc006, 0x00 }, + { 0xc007, 0x00 }, + { 0xc008, 0x00 }, + { 0xc009, 0x00 }, + { 0xc00a, 0x00 }, + { 0xc00b, 0x00 }, + { 0xc00c, 0x00 }, + { 0xc00d, 0x00 }, + { 0xc00e, 0x00 }, + { 0xc00f, 0x00 }, + { 0xc010, 0xa5 }, + { 0xc011, 0x00 }, + { 0xc012, 0xff }, + { 0xc013, 0xff }, + { 0xc014, 0x40 }, + { 0xc015, 0x00 }, + { 0xc016, 0x00 }, + { 0xc017, 0x00 }, + { 0xc605, 0x30 }, + { 0xc700, 0x0a }, + { 0xc701, 0xaa }, + { 0xc702, 0x1a }, + { 0xc703, 0x0a }, + { 0xc710, 0x80 }, + { 0xc711, 0x00 }, + { 0xc712, 0x3e }, + { 0xc713, 0x80 }, + { 0xc714, 0x80 }, + { 0xc715, 0x06 }, + { 0xd101, 0x00 }, + { 0xd102, 0x30 }, + { 0xd103, 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_UDMPU21, RT1316_SDCA_CTL_UDMPU_CLUSTER, 0), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_L), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_R), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_XU24, RT1316_SDCA_CTL_BYPASS, 0), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE23, RT1316_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE22, RT1316_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE24, RT1316_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, +}; + +static bool rt1316_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2f0a: + case 0x2f36: + case 0x3203 ... 0x320e: + case 0xc000 ... 0xc7b4: + case 0xcf00 ... 0xcf03: + case 0xd101 ... 0xd103: + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_UDMPU21, RT1316_SDCA_CTL_UDMPU_CLUSTER, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_L): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_R): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE23, RT1316_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE27, RT1316_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE22, RT1316_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE24, RT1316_SDCA_CTL_REQ_POWER_STATE, 0): + return true; + default: + return false; + } +} + +static bool rt1316_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0xc000: + case 0xc093: + case 0xc09d: + case 0xc0a3: + case 0xc201: + case 0xc427 ... 0xc428: + case 0xd102: + return true; + default: + return false; + } +} + +static const struct regmap_config rt1316_sdw_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = rt1316_readable_register, + .volatile_reg = rt1316_volatile_register, + .max_register = 0x4108ffff, + .reg_defaults = rt1316_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rt1316_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt1316_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + prop->is_sdca = true; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x04; /* BITMAP: 00000100 */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + dev_dbg(&slave->dev, "%s\n", __func__); + + return 0; +} + +static int rt1316_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(dev); + + if (rt1316->hw_init) + return 0; + + if (rt1316->first_hw_init) { + regcache_cache_only(rt1316->regmap, false); + regcache_cache_bypass(rt1316->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + /* sw reset */ + regmap_write(rt1316->regmap, 0xc000, 0x02); + + /* initial settings - blind write */ + regmap_write(rt1316->regmap, 0xc710, 0x17); + regmap_write(rt1316->regmap, 0xc711, 0x80); + regmap_write(rt1316->regmap, 0xc712, 0x26); + regmap_write(rt1316->regmap, 0xc713, 0x06); + regmap_write(rt1316->regmap, 0xc714, 0x80); + regmap_write(rt1316->regmap, 0xc715, 0x06); + regmap_write(rt1316->regmap, 0xc702, 0x0a); + regmap_write(rt1316->regmap, 0xc703, 0x0a); + regmap_write(rt1316->regmap, 0xc001, 0x45); + regmap_write(rt1316->regmap, 0xc003, 0x00); + regmap_write(rt1316->regmap, 0xc004, 0x11); + regmap_write(rt1316->regmap, 0xc005, 0x00); + regmap_write(rt1316->regmap, 0xc006, 0x00); + regmap_write(rt1316->regmap, 0xc106, 0x00); + regmap_write(rt1316->regmap, 0xc007, 0x11); + regmap_write(rt1316->regmap, 0xc008, 0x11); + regmap_write(rt1316->regmap, 0xc009, 0x00); + + regmap_write(rt1316->regmap, 0x2f0a, 0x00); + regmap_write(rt1316->regmap, 0xd101, 0xf0); + regmap_write(rt1316->regmap, 0xd103, 0x9b); + regmap_write(rt1316->regmap, 0x2f36, 0x8e); + regmap_write(rt1316->regmap, 0x3206, 0x80); + regmap_write(rt1316->regmap, 0x3211, 0x0b); + regmap_write(rt1316->regmap, 0x3216, 0x06); + regmap_write(rt1316->regmap, 0xc614, 0x20); + regmap_write(rt1316->regmap, 0xc615, 0x0a); + regmap_write(rt1316->regmap, 0xc616, 0x02); + regmap_write(rt1316->regmap, 0xc617, 0x00); + regmap_write(rt1316->regmap, 0xc60b, 0x10); + regmap_write(rt1316->regmap, 0xc60e, 0x05); + regmap_write(rt1316->regmap, 0xc102, 0x00); + regmap_write(rt1316->regmap, 0xc090, 0xb0); + regmap_write(rt1316->regmap, 0xc00f, 0x01); + regmap_write(rt1316->regmap, 0xc09c, 0x7b); + + regmap_write(rt1316->regmap, 0xc602, 0x07); + regmap_write(rt1316->regmap, 0xc603, 0x07); + regmap_write(rt1316->regmap, 0xc0a3, 0x71); + regmap_write(rt1316->regmap, 0xc00b, 0x30); + regmap_write(rt1316->regmap, 0xc093, 0x80); + regmap_write(rt1316->regmap, 0xc09d, 0x80); + regmap_write(rt1316->regmap, 0xc0b0, 0x77); + regmap_write(rt1316->regmap, 0xc010, 0xa5); + regmap_write(rt1316->regmap, 0xc050, 0x83); + regmap_write(rt1316->regmap, 0x2f55, 0x03); + regmap_write(rt1316->regmap, 0x3217, 0xb5); + regmap_write(rt1316->regmap, 0x3202, 0x02); + + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_XU24, RT1316_SDCA_CTL_BYPASS, 0), 0x00); + + /* for IV sense */ + regmap_write(rt1316->regmap, 0x2232, 0x80); + regmap_write(rt1316->regmap, 0xc0b0, 0x77); + regmap_write(rt1316->regmap, 0xc011, 0x00); + regmap_write(rt1316->regmap, 0xc020, 0x00); + regmap_write(rt1316->regmap, 0xc023, 0x00); + regmap_write(rt1316->regmap, 0x3101, 0x00); + regmap_write(rt1316->regmap, 0x3004, 0xa0); + regmap_write(rt1316->regmap, 0x3005, 0xb1); + regmap_write(rt1316->regmap, 0xc007, 0x11); + regmap_write(rt1316->regmap, 0xc008, 0x11); + regmap_write(rt1316->regmap, 0xc009, 0x00); + regmap_write(rt1316->regmap, 0xc022, 0xd6); + regmap_write(rt1316->regmap, 0xc025, 0xd6); + + regmap_write(rt1316->regmap, 0xd001, 0x03); + regmap_write(rt1316->regmap, 0xd002, 0xbf); + regmap_write(rt1316->regmap, 0xd003, 0x03); + regmap_write(rt1316->regmap, 0xd004, 0xbf); + + if (rt1316->first_hw_init) { + regcache_cache_bypass(rt1316->regmap, false); + regcache_mark_dirty(rt1316->regmap); + } else + rt1316->first_hw_init = true; + + /* Mark Slave initialization complete */ + rt1316->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + return 0; +} + +static int rt1316_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(&slave->dev); + + /* Update the status */ + rt1316->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + rt1316->hw_init = false; + + if (status == SDW_SLAVE_ATTACHED) + regcache_mark_dirty(rt1316->regmap); + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt1316->hw_init || rt1316->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt1316_io_init(&slave->dev, slave); +} + +static int rt1316_classd_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 rt1316_sdw_priv *rt1316 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE23, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE27, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE22, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE23, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE27, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE22, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + break; + + default: + break; + } + + return 0; +} + +static int rt1316_pde24_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 rt1316_sdw_priv *rt1316 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE24, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt1316->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_PDE24, + RT1316_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + break; + } + return 0; +} + +static const char * const rt1316_rx_data_ch_select[] = { + "L,R", + "L,L", + "L,R", + "L,L+R", + "R,L", + "R,R", + "R,L+R", + "L+R,L", + "L+R,R", + "L+R,L+R", +}; + +static SOC_ENUM_SINGLE_DECL(rt1316_rx_data_ch_enum, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_UDMPU21, RT1316_SDCA_CTL_UDMPU_CLUSTER, 0), 0, + rt1316_rx_data_ch_select); + +static const char * const rt1316_xu24_bypass_ctl[] = { + "Not Bypass", + "Bypass", +}; + +static SOC_ENUM_SINGLE_DECL(rt1316_xu24_bypass_enum, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_XU24, RT1316_SDCA_CTL_BYPASS, 0), 0, + rt1316_xu24_bypass_ctl); + +static const char * const rt1316_lr_iv_sel[] = { + "0", + "1", + "2", + "3", + "4", + "5", + "6", + "7", +}; + +static SOC_ENUM_SINGLE_DECL(rt1316_l_v_tag_enum, + 0x3004, 0, rt1316_lr_iv_sel); +static SOC_ENUM_SINGLE_DECL(rt1316_l_i_tag_enum, + 0x3004, 4, rt1316_lr_iv_sel); +static SOC_ENUM_SINGLE_DECL(rt1316_r_v_tag_enum, + 0x3005, 0, rt1316_lr_iv_sel); +static SOC_ENUM_SINGLE_DECL(rt1316_r_i_tag_enum, + 0x3005, 4, rt1316_lr_iv_sel); + +static const struct snd_kcontrol_new rt1316_snd_controls[] = { + + /* I2S Data Channel Selection */ + SOC_ENUM("RX Channel Select", rt1316_rx_data_ch_enum), + + /* XU24 Bypass Control */ + SOC_ENUM("XU24 Bypass Switch", rt1316_xu24_bypass_enum), + + /* Left/Right IV tag */ + SOC_ENUM("Left V Tag Select", rt1316_l_v_tag_enum), + SOC_ENUM("Left I Tag Select", rt1316_l_i_tag_enum), + SOC_ENUM("Right V Tag Select", rt1316_r_v_tag_enum), + SOC_ENUM("Right I Tag Select", rt1316_r_i_tag_enum), + + /* IV mixer Control */ + SOC_DOUBLE("Isense Mixer Switch", 0xc605, 2, 0, 1, 1), + SOC_DOUBLE("Vsense Mixer Switch", 0xc605, 3, 1, 1, 1), +}; + +static const struct snd_kcontrol_new rt1316_sto_dac_l = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_L), + 0, 1, 1); + +static const struct snd_kcontrol_new rt1316_sto_dac_r = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_FU21, RT1316_SDCA_CTL_FU_MUTE, CH_R), + 0, 1, 1); + +static const struct snd_soc_dapm_widget rt1316_dapm_widgets[] = { + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("DP1RX", "DP1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP2TX", "DP2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SWITCH("DAC L", SND_SOC_NOPM, 0, 0, &rt1316_sto_dac_l), + SND_SOC_DAPM_SWITCH("DAC R", SND_SOC_NOPM, 0, 0, &rt1316_sto_dac_r), + + /* Output Lines */ + SND_SOC_DAPM_PGA_E("CLASS D", SND_SOC_NOPM, 0, 0, NULL, 0, + rt1316_classd_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + + SND_SOC_DAPM_SUPPLY("PDE 24", SND_SOC_NOPM, 0, 0, + rt1316_pde24_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA("I Sense", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("V Sense", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SIGGEN("I Gen"), + SND_SOC_DAPM_SIGGEN("V Gen"), +}; + +static const struct snd_soc_dapm_route rt1316_dapm_routes[] = { + { "DAC", NULL, "DP1RX" }, + { "DAC L", "Switch", "DAC" }, + { "DAC R", "Switch", "DAC" }, + { "CLASS D", NULL, "DAC L" }, + { "CLASS D", NULL, "DAC R" }, + { "SPOL", NULL, "CLASS D" }, + { "SPOR", NULL, "CLASS D" }, + + { "I Sense", NULL, "I Gen" }, + { "V Sense", NULL, "V Gen" }, + { "I Sense", NULL, "PDE 24" }, + { "V Sense", NULL, "PDE 24" }, + { "DP2TX", NULL, "I Sense" }, + { "DP2TX", NULL, "V Sense" }, +}; + +static int rt1316_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + struct sdw_stream_data *stream; + + if (!sdw_stream) + return 0; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = sdw_stream; + + /* Use tx_mask or rx_mask to configure stream tag and set dma_data */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = stream; + else + dai->capture_dma_data = stream; + + return 0; +} + +static void rt1316_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int rt1316_sdw_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 rt1316_sdw_priv *rt1316 = + snd_soc_component_get_drvdata(component); + + if (tx_mask) + return -EINVAL; + + if (slots > 2) + return -EINVAL; + + rt1316->rx_mask = rx_mask; + rt1316->slots = slots; + /* slot_width is not used since it's irrelevant for SoundWire */ + + return 0; +} + +static int rt1316_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1316_sdw_priv *rt1316 = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_data *stream; + int retval, port, num_channels, ch_mask; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!rt1316->sdw_slave) + return -EINVAL; + + /* SoundWire specific configuration */ + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + port = 1; + } else { + direction = SDW_DATA_DIR_TX; + port = 2; + } + + if (rt1316->slots) { + num_channels = rt1316->slots; + ch_mask = rt1316->rx_mask; + } else { + num_channels = params_channels(params); + ch_mask = (1 << num_channels) - 1; + } + + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = num_channels; + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + port_config.ch_mask = ch_mask; + port_config.num = port; + + retval = sdw_stream_add_slave(rt1316->sdw_slave, &stream_config, + &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + return 0; +} + +static int rt1316_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1316_sdw_priv *rt1316 = + snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt1316->sdw_slave) + return -EINVAL; + + sdw_stream_remove_slave(rt1316->sdw_slave, stream->sdw_stream); + return 0; +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static struct sdw_slave_ops rt1316_slave_ops = { + .read_prop = rt1316_read_prop, + .update_status = rt1316_update_status, +}; + +static const struct snd_soc_component_driver soc_component_sdw_rt1316 = { + .controls = rt1316_snd_controls, + .num_controls = ARRAY_SIZE(rt1316_snd_controls), + .dapm_widgets = rt1316_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt1316_dapm_widgets), + .dapm_routes = rt1316_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt1316_dapm_routes), +}; + +static const struct snd_soc_dai_ops rt1316_aif_dai_ops = { + .hw_params = rt1316_sdw_hw_params, + .hw_free = rt1316_sdw_pcm_hw_free, + .set_sdw_stream = rt1316_set_sdw_stream, + .shutdown = rt1316_sdw_shutdown, + .set_tdm_slot = rt1316_sdw_set_tdm_slot, +}; + +#define RT1316_STEREO_RATES SNDRV_PCM_RATE_48000 +#define RT1316_FORMATS (SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver rt1316_sdw_dai[] = { + { + .name = "rt1316-aif", + .playback = { + .stream_name = "DP1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT1316_STEREO_RATES, + .formats = RT1316_FORMATS, + }, + .capture = { + .stream_name = "DP2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT1316_STEREO_RATES, + .formats = RT1316_FORMATS, + }, + .ops = &rt1316_aif_dai_ops, + }, +}; + +static int rt1316_sdw_init(struct device *dev, struct regmap *regmap, + struct sdw_slave *slave) +{ + struct rt1316_sdw_priv *rt1316; + int ret; + + rt1316 = devm_kzalloc(dev, sizeof(*rt1316), GFP_KERNEL); + if (!rt1316) + return -ENOMEM; + + dev_set_drvdata(dev, rt1316); + rt1316->sdw_slave = slave; + rt1316->regmap = regmap; + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt1316->hw_init = false; + rt1316->first_hw_init = false; + + ret = devm_snd_soc_register_component(dev, + &soc_component_sdw_rt1316, + rt1316_sdw_dai, + ARRAY_SIZE(rt1316_sdw_dai)); + + dev_dbg(&slave->dev, "%s\n", __func__); + + return ret; +} + +static int rt1316_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *regmap; + + /* Regmap Initialization */ + regmap = devm_regmap_init_sdw(slave, &rt1316_sdw_regmap); + if (!regmap) + return -EINVAL; + + return rt1316_sdw_init(&slave->dev, regmap, slave); +} + +static const struct sdw_device_id rt1316_id[] = { + SDW_SLAVE_ENTRY_EXT(0x025d, 0x1316, 0x3, 0x1, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt1316_id); + +static int __maybe_unused rt1316_dev_suspend(struct device *dev) +{ + struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(dev); + + if (!rt1316->hw_init) + return 0; + + regcache_cache_only(rt1316->regmap, true); + + return 0; +} + +#define RT1316_PROBE_TIMEOUT 5000 + +static int __maybe_unused rt1316_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt1316->hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT1316_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt1316->regmap, false); + regcache_sync(rt1316->regmap); + + return 0; +} + +static const struct dev_pm_ops rt1316_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt1316_dev_suspend, rt1316_dev_resume) + SET_RUNTIME_PM_OPS(rt1316_dev_suspend, rt1316_dev_resume, NULL) +}; + +static struct sdw_driver rt1316_sdw_driver = { + .driver = { + .name = "rt1316-sdca", + .owner = THIS_MODULE, + .pm = &rt1316_pm, + }, + .probe = rt1316_sdw_probe, + .ops = &rt1316_slave_ops, + .id_table = rt1316_id, +}; +module_sdw_driver(rt1316_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT1316 driver SDCA SDW"); +MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt1316-sdw.h b/sound/soc/codecs/rt1316-sdw.h new file mode 100644 index 000000000000..9883a5f90695 --- /dev/null +++ b/sound/soc/codecs/rt1316-sdw.h @@ -0,0 +1,57 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt1316-sdw.h -- RT1316 SDCA ALSA SoC audio driver header + * + * Copyright(c) 2020 Realtek Semiconductor Corp. + */ + +#ifndef __RT1316_SDW_H__ +#define __RT1316_SDW_H__ + +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/soundwire/sdw_registers.h> +#include <sound/soc.h> + +/* RT1316 SDCA Control - function number */ +#define FUNC_NUM_SMART_AMP 0x04 + +/* RT1316 SDCA entity */ +#define RT1316_SDCA_ENT_PDE23 0x31 +#define RT1316_SDCA_ENT_PDE27 0x32 +#define RT1316_SDCA_ENT_PDE22 0x33 +#define RT1316_SDCA_ENT_PDE24 0x34 +#define RT1316_SDCA_ENT_XU24 0x24 +#define RT1316_SDCA_ENT_FU21 0x03 +#define RT1316_SDCA_ENT_UDMPU21 0x02 + +/* RT1316 SDCA control */ +#define RT1316_SDCA_CTL_SAMPLE_FREQ_INDEX 0x10 +#define RT1316_SDCA_CTL_REQ_POWER_STATE 0x01 +#define RT1316_SDCA_CTL_BYPASS 0x01 +#define RT1316_SDCA_CTL_FU_MUTE 0x01 +#define RT1316_SDCA_CTL_FU_VOLUME 0x02 +#define RT1316_SDCA_CTL_UDMPU_CLUSTER 0x10 + +/* RT1316 SDCA channel */ +#define CH_L 0x01 +#define CH_R 0x02 + +struct rt1316_sdw_priv { + struct snd_soc_component *component; + struct regmap *regmap; + struct sdw_slave *sdw_slave; + enum sdw_slave_status status; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; + int rx_mask; + int slots; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +#endif /* __RT1316_SDW_H__ */ -- 2.29.0

3 3

[RFC PATCH] soundwire: bus: use correct driver name in error messages
by Pierre-Louis Bossart 08 Feb '21

08 Feb '21

None of the existing codec drivers set the sdw_driver.name, but instead set sdw_driver.driver.name. This leads to error messages such as [ 23.935355] rt700 sdw:2:25d:700:0: Probe of (null) failed: -19 We could remove this sdw_driver.name if it doesn't have any purpose. This patch only suggests using the proper indirection. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- drivers/soundwire/bus_type.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/bus_type.c b/drivers/soundwire/bus_type.c index 575b9bad99d5..893296f3fe39 100644 --- a/drivers/soundwire/bus_type.c +++ b/drivers/soundwire/bus_type.c @@ -82,6 +82,7 @@ static int sdw_drv_probe(struct device *dev) struct sdw_slave *slave = dev_to_sdw_dev(dev); struct sdw_driver *drv = drv_to_sdw_driver(dev->driver); const struct sdw_device_id *id; + const char *name; int ret; /* @@ -108,7 +109,10 @@ static int sdw_drv_probe(struct device *dev) ret = drv->probe(slave, id); if (ret) { - dev_err(dev, "Probe of %s failed: %d\n", drv->name, ret); + name = drv->name; + if (!name) + name = drv->driver.name; + dev_err(dev, "Probe of %s failed: %d\n", name, ret); dev_pm_domain_detach(dev, false); return ret; } @@ -174,11 +178,16 @@ static void sdw_drv_shutdown(struct device *dev) */ int __sdw_register_driver(struct sdw_driver *drv, struct module *owner) { + const char *name; + drv->driver.bus = &sdw_bus_type; if (!drv->probe) { - pr_err("driver %s didn't provide SDW probe routine\n", - drv->name); + name = drv->name; + if (!name) + name = drv->driver.name; + + pr_err("driver %s didn't provide SDW probe routine\n", name); return -EINVAL; } -- 2.25.1

1 0

[PATCH v2] ALSA: hda: intel-dsp-config: Add SND_INTEL_BYT_PREFER_SOF Kconfig option
by Hans de Goede 08 Feb '21

08 Feb '21

The kernel has 2 drivers for the Low Power Engine audio-block on Bay- and Cherry-Trail SoCs. The old SST driver and the new SOF driver. If both drivers are enabled then the kernel will default to using the old SST driver, unless told otherwise through the snd_intel_dspcfg.dsp_driver module-parameter. Add a boolean SND_INTEL_BYT_PREFER_SOF Kconfig option, which when set to Y will make the kernel default to the new SOF driver instead. The option defaults to n, preserving the current behavior. Making this configurable will help distributions such as Fedora: https://fedoraproject.org/w/index.php?title=Changes/SofDefaultForIntelLpe to test using SOF on BYT/CHT during the transition phase where we have both drivers (eventually the old driver and this option will be removed). Note that this drops the acpi_config_table[] containing 2 entries per ACPI hardware-id if both drivers are enabled. snd_intel_acpi_dsp_find_config() will always return the first hit, so we only need a single entry with the flags value set depending on the Kconfig settings. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Changes in v2: - Rework following Takashi's suggestions --- sound/hda/Kconfig | 14 ++++++++++++++ sound/hda/intel-dsp-config.c | 29 ++++++++++++----------------- 2 files changed, 26 insertions(+), 17 deletions(-) diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3bc9224d5e4f..9ed5cfa3c18c 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -46,3 +46,17 @@ config SND_INTEL_DSP_CONFIG select SND_INTEL_NHLT if ACPI # this config should be selected only for Intel DSP platforms. # A fallback is provided so that the code compiles in all cases. + +config SND_INTEL_BYT_PREFER_SOF + bool "Prefer SOF driver over SST on BY/CHT platforms" + depends on SND_SST_ATOM_HIFI2_PLATFORM_ACPI && SND_SOC_SOF_BAYTRAIL + default n + help + The kernel has 2 drivers for the Low Power Engine audio-block on + Bay- and Cherry-Trail SoCs. The old SST driver and the new SOF + driver. If both drivers are enabled then the kernel will default + to using the old SST driver, unless told otherwise through the + snd_intel_dspcfg.dsp_driver module-parameter. + + Set this option to Y to make the kernel default to the new SOF + driver instead. diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index c45686172517..68bb977c6a37 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -452,35 +452,30 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci) } EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); +/* Should we default to SOF or SST for BYT/CHT ? */ +#if IS_ENABLED(CONFIG_SND_INTEL_BYT_PREFER_SOF) || \ + !IS_ENABLED(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) +#define FLAG_SST_OR_SOF_BYT FLAG_SOF +#else +#define FLAG_SST_OR_SOF_BYT FLAG_SST +#endif + /* * configuration table * - the order of similar ACPI ID entries is important! * - the first successful match will win */ static const struct config_entry acpi_config_table[] = { +#if IS_ENABLED(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) || \ + IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) /* BayTrail */ -#if IS_ENABLED(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) - { - .flags = FLAG_SST, - .acpi_hid = "80860F28", - }, -#endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) { - .flags = FLAG_SOF, + .flags = FLAG_SST_OR_SOF_BYT, .acpi_hid = "80860F28", }, -#endif /* CherryTrail */ -#if IS_ENABLED(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) - { - .flags = FLAG_SST, - .acpi_hid = "808622A8", - }, -#endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) { - .flags = FLAG_SOF, + .flags = FLAG_SST_OR_SOF_BYT, .acpi_hid = "808622A8", }, #endif -- 2.30.0

3 2

[PATCH] ALSA: fireface: fix to parse sync status register of latter protocol
by Takashi Sakamoto 08 Feb '21

08 Feb '21

Fireface UCX, UFX, and FF802 are categorized for latter protocol of the series. Current support for FF802 (and UFX) includes failure to parse sync status register and results in EIO. Further investigation figures out that the content of register differs depending on models. This commit adds tables specific to FF802 and UFX to fix it. Fixes: 062bb452b078b ("ALSA: fireface: add support for RME FireFace 802") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireface/ff-protocol-latter.c | 118 ++++++++++++++++--- 1 file changed, 100 insertions(+), 18 deletions(-) diff --git a/sound/firewire/fireface/ff-protocol-latter.c b/sound/firewire/fireface/ff-protocol-latter.c index 8d3b23778eb2..7ddb7b97f02d 100644 --- a/sound/firewire/fireface/ff-protocol-latter.c +++ b/sound/firewire/fireface/ff-protocol-latter.c @@ -15,6 +15,61 @@ #define LATTER_FETCH_MODE 0xffff00000010ULL #define LATTER_SYNC_STATUS 0x0000801c0000ULL +// The content of sync status register differs between models. +// +// Fireface UCX: +// 0xf0000000: (unidentified) +// 0x0f000000: effective rate of sampling clock +// 0x00f00000: detected rate of word clock on BNC interface +// 0x000f0000: detected rate of ADAT or S/PDIF on optical interface +// 0x0000f000: detected rate of S/PDIF on coaxial interface +// 0x00000e00: effective source of sampling clock +// 0x00000e00: Internal +// 0x00000800: (unidentified) +// 0x00000600: Word clock on BNC interface +// 0x00000400: ADAT on optical interface +// 0x00000200: S/PDIF on coaxial or optical interface +// 0x00000100: Optical interface is used for ADAT signal +// 0x00000080: (unidentified) +// 0x00000040: Synchronized to word clock on BNC interface +// 0x00000020: Synchronized to ADAT or S/PDIF on optical interface +// 0x00000010: Synchronized to S/PDIF on coaxial interface +// 0x00000008: (unidentified) +// 0x00000004: Lock word clock on BNC interface +// 0x00000002: Lock ADAT or S/PDIF on optical interface +// 0x00000001: Lock S/PDIF on coaxial interface +// +// Fireface 802 (and perhaps UFX): +// 0xf0000000: effective rate of sampling clock +// 0x0f000000: detected rate of ADAT-B on 2nd optical interface +// 0x00f00000: detected rate of ADAT-A on 1st optical interface +// 0x000f0000: detected rate of AES/EBU on XLR or coaxial interface +// 0x0000f000: detected rate of word clock on BNC interface +// 0x00000e00: effective source of sampling clock +// 0x00000e00: internal +// 0x00000800: ADAT-B +// 0x00000600: ADAT-A +// 0x00000400: AES/EBU +// 0x00000200: Word clock +// 0x00000080: Synchronized to ADAT-B on 2nd optical interface +// 0x00000040: Synchronized to ADAT-A on 1st optical interface +// 0x00000020: Synchronized to AES/EBU on XLR or 2nd optical interface +// 0x00000010: Synchronized to word clock on BNC interface +// 0x00000008: Lock ADAT-B on 2nd optical interface +// 0x00000004: Lock ADAT-A on 1st optical interface +// 0x00000002: Lock AES/EBU on XLR or 2nd optical interface +// 0x00000001: Lock word clock on BNC interface +// +// The pattern for rate bits: +// 0x00: 32.0 kHz +// 0x01: 44.1 kHz +// 0x02: 48.0 kHz +// 0x04: 64.0 kHz +// 0x05: 88.2 kHz +// 0x06: 96.0 kHz +// 0x08: 128.0 kHz +// 0x09: 176.4 kHz +// 0x0a: 192.0 kHz static int parse_clock_bits(u32 data, unsigned int *rate, enum snd_ff_clock_src *src, enum snd_ff_unit_version unit_version) @@ -23,35 +78,48 @@ static int parse_clock_bits(u32 data, unsigned int *rate, unsigned int rate; u32 flag; } *rate_entry, rate_entries[] = { - { 32000, 0x00000000, }, - { 44100, 0x01000000, }, - { 48000, 0x02000000, }, - { 64000, 0x04000000, }, - { 88200, 0x05000000, }, - { 96000, 0x06000000, }, - { 128000, 0x08000000, }, - { 176400, 0x09000000, }, - { 192000, 0x0a000000, }, + { 32000, 0x00, }, + { 44100, 0x01, }, + { 48000, 0x02, }, + { 64000, 0x04, }, + { 88200, 0x05, }, + { 96000, 0x06, }, + { 128000, 0x08, }, + { 176400, 0x09, }, + { 192000, 0x0a, }, }; static const struct { enum snd_ff_clock_src src; u32 flag; - } *clk_entry, clk_entries[] = { + } *clk_entry, *clk_entries, ucx_clk_entries[] = { { SND_FF_CLOCK_SRC_SPDIF, 0x00000200, }, { SND_FF_CLOCK_SRC_ADAT1, 0x00000400, }, { SND_FF_CLOCK_SRC_WORD, 0x00000600, }, { SND_FF_CLOCK_SRC_INTERNAL, 0x00000e00, }, + }, ufx_ff802_clk_entries[] = { + { SND_FF_CLOCK_SRC_WORD, 0x00000200, }, + { SND_FF_CLOCK_SRC_SPDIF, 0x00000400, }, + { SND_FF_CLOCK_SRC_ADAT1, 0x00000600, }, + { SND_FF_CLOCK_SRC_ADAT2, 0x00000800, }, + { SND_FF_CLOCK_SRC_INTERNAL, 0x00000e00, }, }; + u32 rate_bits; + unsigned int clk_entry_count; int i; - if (unit_version != SND_FF_UNIT_VERSION_UCX) { - // e.g. 0x00fe0f20 but expected 0x00eff002. - data = ((data & 0xf0f0f0f0) >> 4) | ((data & 0x0f0f0f0f) << 4); + if (unit_version == SND_FF_UNIT_VERSION_UCX) { + rate_bits = (data & 0x0f000000) >> 24; + clk_entries = ucx_clk_entries; + clk_entry_count = ARRAY_SIZE(ucx_clk_entries); + } else { + rate_bits = (data & 0xf0000000) >> 28; + clk_entries = ufx_ff802_clk_entries; + clk_entry_count = ARRAY_SIZE(ufx_ff802_clk_entries); } for (i = 0; i < ARRAY_SIZE(rate_entries); ++i) { rate_entry = rate_entries + i; - if ((data & 0x0f000000) == rate_entry->flag) { + if (rate_bits == rate_entry->flag) { *rate = rate_entry->rate; break; } @@ -59,14 +127,14 @@ static int parse_clock_bits(u32 data, unsigned int *rate, if (i == ARRAY_SIZE(rate_entries)) return -EIO; - for (i = 0; i < ARRAY_SIZE(clk_entries); ++i) { + for (i = 0; i < clk_entry_count; ++i) { clk_entry = clk_entries + i; if ((data & 0x000e00) == clk_entry->flag) { *src = clk_entry->src; break; } } - if (i == ARRAY_SIZE(clk_entries)) + if (i == clk_entry_count) return -EIO; return 0; @@ -249,16 +317,22 @@ static void latter_dump_status(struct snd_ff *ff, struct snd_info_buffer *buffer char *const label; u32 locked_mask; u32 synced_mask; - } *clk_entry, clk_entries[] = { + } *clk_entry, *clk_entries, ucx_clk_entries[] = { { "S/PDIF", 0x00000001, 0x00000010, }, { "ADAT", 0x00000002, 0x00000020, }, { "WDClk", 0x00000004, 0x00000040, }, + }, ufx_ff802_clk_entries[] = { + { "WDClk", 0x00000001, 0x00000010, }, + { "AES/EBU", 0x00000002, 0x00000020, }, + { "ADAT-A", 0x00000004, 0x00000040, }, + { "ADAT-B", 0x00000008, 0x00000080, }, }; __le32 reg; u32 data; unsigned int rate; enum snd_ff_clock_src src; const char *label; + unsigned int clk_entry_count; int i; int err; @@ -270,7 +344,15 @@ static void latter_dump_status(struct snd_ff *ff, struct snd_info_buffer *buffer snd_iprintf(buffer, "External source detection:\n"); - for (i = 0; i < ARRAY_SIZE(clk_entries); ++i) { + if (ff->unit_version == SND_FF_UNIT_VERSION_UCX) { + clk_entries = ucx_clk_entries; + clk_entry_count = ARRAY_SIZE(ucx_clk_entries); + } else { + clk_entries = ufx_ff802_clk_entries; + clk_entry_count = ARRAY_SIZE(ufx_ff802_clk_entries); + } + + for (i = 0; i < clk_entry_count; ++i) { clk_entry = clk_entries + i; snd_iprintf(buffer, "%s: ", clk_entry->label); if (data & clk_entry->locked_mask) { -- 2.27.0

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[PATCH v2 0/3] ALSA: hda: add link_power op to hdac_bus_ops
by Kai Vehmanen 08 Feb '21

08 Feb '21

Here's the 2nd iteration of the patch to improve link management granularity between controllers using HDA extended bus and HDA codec drivers. Based on feedback to V1, this now adds a new link_power op. This allows to decouple HDA codec drivers from the hdac_ext core. I also included two patches to SOF to show how this interface is used in SOF. Kai Vehmanen (3): ALSA: hda: add link_power op to hdac_bus_ops ASoC: SOF: Intel: hda: use hdac_ext fine-grained link management ASoC: SOF: Intel: hda: release display power at link_power include/sound/hdaudio.h | 14 ++++------- include/sound/hdaudio_ext.h | 2 ++ sound/hda/ext/hdac_ext_controller.c | 37 +++++++++++++++++++++++++++++ sound/hda/hdac_bus.c | 23 ++++++++++++++++++ sound/hda/hdac_controller.c | 14 +++++++++++ sound/soc/sof/intel/hda-bus.c | 33 ++++++++++++++++++++++++- sound/soc/sof/intel/hda.h | 3 ++- 7 files changed, 115 insertions(+), 11 deletions(-) base-commit: 387740a5e9532db13dfb577bd4b8d1aa496487ab -- 2.29.2

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