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Add jack2 as test
by GitHub pull_request - opened 20 Jan '21

20 Jan '21

alsa-project/alsa-utils pull request #74 was opened from bengan: Jack2 been around for a long time. It should be included in the test. Request URL : https://github.com/alsa-project/alsa-utils/pull/74 Patch URL : https://github.com/alsa-project/alsa-utils/pull/74.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 1/2] ASoC: dt-bindings: rt5682: add LDO2 power control device property
by shumingf＠realtek.com 20 Jan '21

20 Jan '21

From: Shuming Fan <shumingf(a)realtek.com> The "realtek,ldo2-pow-en" device property uses to enable LDO2 power while the headset recording. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5682.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index 9c5fadb6ac82..eba2ea588564 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -46,6 +46,8 @@ Optional properties: - realtek,dmic-clk-driving-high : Set the high drving of the DMIC clock out. +- realtek,ldo2-pow-en : Enable LDO2 power control if the application circuit uses VBAT, not MICVDD. + Pins on the device (for linking into audio routes) for RT5682: * DMIC L1 -- 2.29.0

1 0

[PATCH] Add basic capture support for Pioneer DJ DJM-250MK2
by František Kučera 20 Jan '21

20 Jan '21

From: František Kučera <franta-linux(a)frantovo.cz> This patch extends support for DJM-250MK2 and allows recording. However, DVS is not possible yet (see the comment in code). Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> --- sound/usb/pcm.c | 1 + sound/usb/quirks-table.h | 62 ++++++++++++++++++++++++++++++++++------ sound/usb/quirks.c | 1 + 3 files changed, 56 insertions(+), 8 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 5600751803cf..f5fa11deea4d 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -374,6 +374,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 2; goto add_sync_ep_from_ifnum; case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ ep = 0x82; ifnum = 0; goto add_sync_ep_from_ifnum; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f4fb002e3ef4..c6b2f0b8d3e2 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3549,14 +3549,40 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), { /* * Pioneer DJ DJM-250MK2 - * PCM is 8 channels out @ 48 fixed (endpoints 0x01). - * The output from computer to the mixer is usable. - * - * The input (phono or line to computer) is not working. - * It should be at endpoint 0x82 and probably also 8 channels, - * but it seems that it works only with Pioneer proprietary software. - * Even on officially supported OS, the Audacity was unable to record - * and Mixxx to recognize the control vinyls. + * PCM is 8 channels out @ 48 fixed (endpoint 0x01) + * and 8 channels in @ 48 fixed (endpoint 0x82). + * + * Both playback and recording is working, even simultaneously. + * + * Playback channels could be mapped to: + * - CH1 + * - CH2 + * - AUX + * + * Recording channels could be mapped to: + * - Post CH1 Fader + * - Post CH2 Fader + * - Cross Fader A + * - Cross Fader B + * - MIC + * - AUX + * - REC OUT + * + * There is remaining problem with recording directly from PHONO/LINE. + * If we map a channel to: + * - CH1 Control Tone PHONO + * - CH1 Control Tone LINE + * - CH2 Control Tone PHONO + * - CH2 Control Tone LINE + * it is silent. + * There is no signal even on other operating systems with official drivers. + * The signal appears only when a supported application is started. + * This needs to be investigated yet... + * (there is quite a lot communication on the USB in both directions) + * + * In current version this mixer could be used for playback + * and for recording from vinyls (through Post CH* Fader) + * but not for DVS (Digital Vinyl Systems) like in Mixxx. */ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { @@ -3580,6 +3606,26 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .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_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 } } }, { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index abf99b814a0f..6a0be7e7f4fe 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1493,6 +1493,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, set_format_emu_quirk(subs, fmt); break; case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ pioneer_djm_set_format_quirk(subs); break; case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ -- 2.20.1

5 9

Dual ALC1220 Bug MSI Godlike X570
by Aeonik 20 Jan '21

20 Jan '21

Hello, I believe I have run into an Alsa bug due to new/unsupported hardware. I am running Arch Linux Kernel version 5.10.8 on my new MSI Godlike X570 motherboard, and I believe Alsa is having issues with the Dual ALC1220 Codecs. I saw that this has been an issue before <https://bugzilla.kernel.org/show_bug.cgi?id=195305> in the and looks solved, but for some reason, my Alsa utilities are crashing when trying to access my sound card (Alsamixer and amixer). Here is an example with the error message. amixer -c 0 ALSA lib simple_none.c:1544:(simple_add1) helem (MIXER,'Master Playback Switch',0,1,0) appears twice or more cannot load mixer controls: Invalid argument I have tried solving the issue in a variety of ways, including explicitly setting the kernel module snd-hda_intel model name. I also experience the same bug in a fresh Arch Linux ISO, which rules out most configuration issues, I think. I reviewed your documentation, and it looks like my next step is to file a bug report. I already reached out to the Arch Linux community, and you can review my post at the following link. I have also attached the output of alsa-info.sh to this email. https://bbs.archlinux.org/viewtopic.php?pid=1949255#p1949255 Let me know if you need anything else from me, I appreciate all your work, and look forward to your response! Aeonik

1 0

[PATCH AUTOSEL 4.4 1/4] ASoC: Intel: haswell: Add missing pm_ops
by Sasha Levin 20 Jan '21

20 Jan '21

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit bb224c3e3e41d940612d4cc9573289cdbd5cb8f5 ] haswell machine board is missing pm_ops what prevents it from undergoing suspend-resume procedure successfully. Assign default snd_soc_pm_ops so this is no longer the case. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20201217105401.27865-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/haswell.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index de955c2e8c4e3..a0e67d5f59683 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -197,6 +197,7 @@ static struct platform_driver haswell_audio = { .probe = haswell_audio_probe, .driver = { .name = "haswell-audio", + .pm = &snd_soc_pm_ops, }, }; -- 2.27.0

1 0

[PATCH AUTOSEL 4.9 1/4] ASoC: Intel: haswell: Add missing pm_ops
by Sasha Levin 20 Jan '21

20 Jan '21

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit bb224c3e3e41d940612d4cc9573289cdbd5cb8f5 ] haswell machine board is missing pm_ops what prevents it from undergoing suspend-resume procedure successfully. Assign default snd_soc_pm_ops so this is no longer the case. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20201217105401.27865-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/haswell.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 11d0cc2b0e390..060da95770416 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -197,6 +197,7 @@ static struct platform_driver haswell_audio = { .probe = haswell_audio_probe, .driver = { .name = "haswell-audio", + .pm = &snd_soc_pm_ops, }, }; -- 2.27.0

1 0

[PATCH AUTOSEL 4.14 1/9] ASoC: Intel: haswell: Add missing pm_ops
by Sasha Levin 20 Jan '21

20 Jan '21

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit bb224c3e3e41d940612d4cc9573289cdbd5cb8f5 ] haswell machine board is missing pm_ops what prevents it from undergoing suspend-resume procedure successfully. Assign default snd_soc_pm_ops so this is no longer the case. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20201217105401.27865-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/haswell.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 8158409921e02..c6007aa95fff1 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -197,6 +197,7 @@ static struct platform_driver haswell_audio = { .probe = haswell_audio_probe, .driver = { .name = "haswell-audio", + .pm = &snd_soc_pm_ops, }, }; -- 2.27.0

1 0

[PATCH AUTOSEL 4.19 01/15] ASoC: Intel: haswell: Add missing pm_ops
by Sasha Levin 20 Jan '21

20 Jan '21

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit bb224c3e3e41d940612d4cc9573289cdbd5cb8f5 ] haswell machine board is missing pm_ops what prevents it from undergoing suspend-resume procedure successfully. Assign default snd_soc_pm_ops so this is no longer the case. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20201217105401.27865-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/haswell.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index a4022983a7ce0..67eb4a446c3cb 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -198,6 +198,7 @@ static struct platform_driver haswell_audio = { .probe = haswell_audio_probe, .driver = { .name = "haswell-audio", + .pm = &snd_soc_pm_ops, }, }; -- 2.27.0

1 0

[PATCH AUTOSEL 5.4 01/26] ASoC: Intel: haswell: Add missing pm_ops
by Sasha Levin 20 Jan '21

20 Jan '21

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit bb224c3e3e41d940612d4cc9573289cdbd5cb8f5 ] haswell machine board is missing pm_ops what prevents it from undergoing suspend-resume procedure successfully. Assign default snd_soc_pm_ops so this is no longer the case. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20201217105401.27865-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/haswell.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 3dadf9bff796a..cf47fd9cd506b 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -206,6 +206,7 @@ static struct platform_driver haswell_audio = { .probe = haswell_audio_probe, .driver = { .name = "haswell-audio", + .pm = &snd_soc_pm_ops, }, }; -- 2.27.0

1 0

[PATCH AUTOSEL 5.10 01/45] ASoC: Intel: haswell: Add missing pm_ops
by Sasha Levin 20 Jan '21

20 Jan '21

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit bb224c3e3e41d940612d4cc9573289cdbd5cb8f5 ] haswell machine board is missing pm_ops what prevents it from undergoing suspend-resume procedure successfully. Assign default snd_soc_pm_ops so this is no longer the case. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20201217105401.27865-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/haswell.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index c55d1239e705b..c763bfeb1f38f 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -189,6 +189,7 @@ static struct platform_driver haswell_audio = { .probe = haswell_audio_probe, .driver = { .name = "haswell-audio", + .pm = &snd_soc_pm_ops, }, }; -- 2.27.0

1 3

[PATCH] ASoC: qcom: lpass: Fix i2s ctl register bit map
by Jun Nie 19 Jan '21

19 Jan '21

Fix bitwidth mapping in i2s ctl register per APQ8016 document. Signed-off-by: Jun Nie <jun.nie(a)linaro.org> --- sound/soc/qcom/lpass-apq8016.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 0aedb3a0a798..7c0e774ad062 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -250,7 +250,7 @@ static struct lpass_variant apq8016_data = { .micmode = REG_FIELD_ID(0x1000, 4, 7, 4, 0x1000), .micmono = REG_FIELD_ID(0x1000, 3, 3, 4, 0x1000), .wssrc = REG_FIELD_ID(0x1000, 2, 2, 4, 0x1000), - .bitwidth = REG_FIELD_ID(0x1000, 0, 0, 4, 0x1000), + .bitwidth = REG_FIELD_ID(0x1000, 0, 1, 4, 0x1000), .rdma_dyncclk = REG_FIELD_ID(0x8400, 12, 12, 2, 0x1000), .rdma_bursten = REG_FIELD_ID(0x8400, 11, 11, 2, 0x1000), -- 2.25.1

2 2

[PATCH v1 0/5] Clock and reset improvements for Tegra ALSA drivers
by Dmitry Osipenko 19 Jan '21

19 Jan '21

This series improves the handling of clock and reset controls of NVIDA Tegra ALSA drivers. Tegra HDA and AHUB drivers aren't handling resets properly, which needs to be fixed in order to unblock other patches related to fixes on the reset controller driver since HDA/AHUB are bound to fail once reset controller driver will be corrected. In particular ALSA drivers are relying on implicit de-assertion of resets which is done by the tegra-clk driver. It's not the business of the clk driver to touch resets and we need to fix this because it breaks reset/clk programming sequences of other Tegra drivers. Dmitry Osipenko (5): ALSA: hda/tegra: Use clk_bulk helpers ALSA: hda/tegra: Reset hardware ASoC: tegra: ahub: Use of_reset_control_array_get_exclusive() ASoC: tegra: ahub: Use clk_bulk helpers ASoC: tegra: ahub: Reset hardware properly sound/pci/hda/hda_tegra.c | 86 +++++++++------------------ sound/soc/tegra/tegra30_ahub.c | 103 ++++++--------------------------- sound/soc/tegra/tegra30_ahub.h | 6 +- 3 files changed, 49 insertions(+), 146 deletions(-) -- 2.29.2

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[PATCH v2] ALSA: hda: Balance runtime/system PM if direct-complete is disabled
by Kai-Heng Feng 19 Jan '21

19 Jan '21

After hibernation, HDA controller can't be runtime-suspended after commit 215a22ed31a1 ("ALSA: hda: Refactor codjc PM to use direct-complete optimization"), which enables direct-complete for HDA codec. The HDA codec driver didn't expect direct-complete will be disabled after it returns a positive value from prepare() callback. However, there are some places that PM core can disable direct-complete. For instance, system hibernation or when codec has subordinates like LEDs. So if the codec is prepared for direct-complete but PM core still calls codec's suspend or freeze callback, partially revert the commit and take the original approach, which uses pm_runtime_force_*() helpers to ensure PM refcount are balanced. Meanwhile, still keep prepare() and complete() callbacks to enable direct-complete and request a resume for jack detection, respectively. Reported-by: Kenneth R. Crudup <kenny(a)panix.com> Fixes: 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization") Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v2: - Use pm_runtime_force_*() helpers to avoid suspend/resume ping pong. sound/pci/hda/hda_codec.c | 24 +++++++----------------- 1 file changed, 7 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 687216e74526..eec1775dfffe 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2934,7 +2934,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) snd_hdac_leave_pm(&codec->core); } -static int hda_codec_suspend(struct device *dev) +static int hda_codec_runtime_suspend(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); unsigned int state; @@ -2953,7 +2953,7 @@ static int hda_codec_suspend(struct device *dev) return 0; } -static int hda_codec_resume(struct device *dev) +static int hda_codec_runtime_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); @@ -2968,16 +2968,6 @@ static int hda_codec_resume(struct device *dev) return 0; } -static int hda_codec_runtime_suspend(struct device *dev) -{ - return hda_codec_suspend(dev); -} - -static int hda_codec_runtime_resume(struct device *dev) -{ - return hda_codec_resume(dev); -} - #endif /* CONFIG_PM */ #ifdef CONFIG_PM_SLEEP @@ -2998,31 +2988,31 @@ static void hda_codec_pm_complete(struct device *dev) static int hda_codec_pm_suspend(struct device *dev) { dev->power.power_state = PMSG_SUSPEND; - return hda_codec_suspend(dev); + return pm_runtime_force_suspend(dev); } static int hda_codec_pm_resume(struct device *dev) { dev->power.power_state = PMSG_RESUME; - return hda_codec_resume(dev); + return pm_runtime_force_resume(dev); } static int hda_codec_pm_freeze(struct device *dev) { dev->power.power_state = PMSG_FREEZE; - return hda_codec_suspend(dev); + return pm_runtime_force_suspend(dev); } static int hda_codec_pm_thaw(struct device *dev) { dev->power.power_state = PMSG_THAW; - return hda_codec_resume(dev); + return pm_runtime_force_resume(dev); } static int hda_codec_pm_restore(struct device *dev) { dev->power.power_state = PMSG_RESTORE; - return hda_codec_resume(dev); + return pm_runtime_force_resume(dev); } #endif /* CONFIG_PM_SLEEP */ -- 2.29.2

2 1

[PATCH] cfg80211: Fix "suspicious RCU usage in wiphy_apply_custom_regulatory" warning/backtrace
by Hans de Goede 19 Jan '21

19 Jan '21

Commit beee24695157 ("cfg80211: Save the regulatory domain when setting custom regulatory") adds a get_wiphy_regdom call to wiphy_apply_custom_regulatory. But as the comment above wiphy_apply_custom_regulatory says: "/* Used by drivers prior to wiphy registration */" this function is used by driver's probe function before the wiphy is registered and at this point wiphy->regd will typically by NULL and calling rcu_dereference_rtnl on a NULL pointer causes the following warning/backtrace: ============================= WARNING: suspicious RCU usage 5.11.0-rc1+ #19 Tainted: G W ----------------------------- net/wireless/reg.c:144 suspicious rcu_dereference_check() usage! other info that might help us debug this: rcu_scheduler_active = 2, debug_locks = 1 2 locks held by kworker/2:0/22: #0: ffff9a4bc104df38 ((wq_completion)events){+.+.}-{0:0}, at: process_one_work+0x1ee/0x570 #1: ffffb6e94010be78 ((work_completion)(&fw_work->work)){+.+.}-{0:0}, at: process_one_work+0x1ee/0x570 stack backtrace: CPU: 2 PID: 22 Comm: kworker/2:0 Tainted: G W 5.11.0-rc1+ #19 Hardware name: LENOVO 60073/INVALID, BIOS 01WT17WW 08/01/2014 Workqueue: events request_firmware_work_func Call Trace: dump_stack+0x8b/0xb0 get_wiphy_regdom+0x57/0x60 [cfg80211] wiphy_apply_custom_regulatory+0xa0/0xf0 [cfg80211] brcmf_cfg80211_attach+0xb02/0x1360 [brcmfmac] brcmf_attach+0x189/0x460 [brcmfmac] brcmf_sdio_firmware_callback+0x78a/0x8f0 [brcmfmac] brcmf_fw_request_done+0x67/0xf0 [brcmfmac] request_firmware_work_func+0x3d/0x70 process_one_work+0x26e/0x570 worker_thread+0x55/0x3c0 ? process_one_work+0x570/0x570 kthread+0x137/0x150 ? __kthread_bind_mask+0x60/0x60 ret_from_fork+0x22/0x30 Add a check for wiphy->regd being NULL before calling get_wiphy_regdom (as is already done in other places) to fix this. wiphy->regd will likely always be NULL when wiphy_apply_custom_regulatory gets called, so arguably the tmp = get_wiphy_regdom() and rcu_free_regdom(tmp) calls should simply be dropped, this patch keeps the 2 calls, to allow drivers to call wiphy_apply_custom_regulatory more then once if necessary. Cc: Ilan Peer <ilan.peer(a)intel.com> Fixes: beee24695157 ("cfg80211: Save the regulatory domain when setting custom regulator") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- net/wireless/reg.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/net/wireless/reg.c b/net/wireless/reg.c index bb72447ad960..9254b9cbaa21 100644 --- a/net/wireless/reg.c +++ b/net/wireless/reg.c @@ -2547,7 +2547,7 @@ static void handle_band_custom(struct wiphy *wiphy, void wiphy_apply_custom_regulatory(struct wiphy *wiphy, const struct ieee80211_regdomain *regd) { - const struct ieee80211_regdomain *new_regd, *tmp; + const struct ieee80211_regdomain *new_regd, *tmp = NULL; enum nl80211_band band; unsigned int bands_set = 0; @@ -2571,7 +2571,8 @@ void wiphy_apply_custom_regulatory(struct wiphy *wiphy, if (IS_ERR(new_regd)) return; - tmp = get_wiphy_regdom(wiphy); + if (wiphy->regd) + tmp = get_wiphy_regdom(wiphy); rcu_assign_pointer(wiphy->regd, new_regd); rcu_free_regdom(tmp); } -- 2.28.0

2 4

[PATCH 0/5] soundwire: fix ACK/NAK handling and improve log
by Bard Liao 19 Jan '21

19 Jan '21

The existing code reports a NAK only when ACK=0 This is not aligned with the SoundWire 1.x specifications. Table 32 in the SoundWire 1.2 specification shows that a Device shall not set NAK=1 if ACK=1. But Table 33 shows the Combined Response may very well be NAK=1/ACK=1, e.g. if another Device than the one addressed reports a parity error. NAK=1 signals a 'Command_Aborted', regardless of the ACK bit value. Move the tests for NAK so that the NAK=1/ACK=1 combination is properly detected according to the specification. Also, improve the demesg log to get more information for debugging. Bard Liao (1): soundwire: bus: add more details to track failed transfers Pierre-Louis Bossart (4): soundwire: use consistent format for Slave devID logs soundwire: cadence: add status in dev_dbg 'State change' log soundwire: cadence: fix ACK/NAK handling soundwire: cadence: adjust verbosity in response handling drivers/soundwire/bus.c | 11 ++++++----- drivers/soundwire/cadence_master.c | 29 +++++++++++++++-------------- drivers/soundwire/slave.c | 10 ++++------ 3 files changed, 25 insertions(+), 25 deletions(-) -- 2.17.1

2 6

[PATCH] MAINTAINERS: soundwire: Add soundwire tree
by Vinod Koul 19 Jan '21

19 Jan '21

Somehow the soundwire MAINTAINERS did not have the tree details, so add it. Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 546aa66428c9..10b234688f22 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -16654,6 +16654,7 @@ R: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> R: Sanyog Kale <sanyog.r.kale(a)intel.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported +T: git git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git F: Documentation/driver-api/soundwire/ F: drivers/soundwire/ F: include/linux/soundwire/ -- 2.26.2

1 1

[PATCH] soundwire: sysfs: Constiyf static struct attribute_group
by Rikard Falkeborn 19 Jan '21

19 Jan '21

The only place sdw_slave_dev_attr_group is used is when its address is passed to devm_device_add_group() which takes a pointer to const struct attribute_group. Make it const to allow the compiler to put it in read-only memory. This makes all attribute_group structs in the file const. Done with the help of Coccinelle. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- drivers/soundwire/sysfs_slave.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/sysfs_slave.c b/drivers/soundwire/sysfs_slave.c index b48b6617a396..3210359cd944 100644 --- a/drivers/soundwire/sysfs_slave.c +++ b/drivers/soundwire/sysfs_slave.c @@ -130,7 +130,7 @@ static struct attribute *slave_dev_attrs[] = { * we don't use ATTRIBUTES_GROUP here since we want to add a subdirectory * for device-level properties */ -static struct attribute_group sdw_slave_dev_attr_group = { +static const struct attribute_group sdw_slave_dev_attr_group = { .attrs = slave_dev_attrs, .name = "dev-properties", }; -- 2.30.0

2 2

[PATCH 0/2] ASoC/SoundWire: fix timeout values
by Bard Liao 19 Jan '21

19 Jan '21

The timeout for an individual transaction w/ the Cadence IP is the same as the entire resume operation for codecs. This doesn't make sense, we need to have at least one order of magnitude between individual transactions and the entire resume operation. Set the timeout on the Cadence side to 500ms and 5s for the codec resume. Both ASoC and SoundWire trees are fine for this series. Pierre-Louis Bossart (2): ASoC: codecs: soundwire: increase resume timeout soundwire: cadence: reduce timeout on transactions drivers/soundwire/cadence_master.c | 2 +- sound/soc/codecs/max98373-sdw.c | 4 +++- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt5682.h | 2 +- sound/soc/codecs/rt700-sdw.c | 2 +- sound/soc/codecs/rt711-sdw.c | 2 +- sound/soc/codecs/rt715-sdw.c | 2 +- 7 files changed, 9 insertions(+), 7 deletions(-) -- 2.17.1

3 4

[PATCH] soundwire: intel: don't return error when clock stop failed
by Bard Liao 19 Jan '21

19 Jan '21

dev->power.runtime_error will be set to the return value of the runtime suspend callback function, and runtime resume function will return -EINVAL if dev->power.runtime_error is not 0. Somehow the codec rarely doesn't return an ACK to the clock prepare command. If we stop the runtime suspend process and return error, we will not be able to resume again. Likewise, if the codec lost sync and did not rejoin, the resume operation will also fail. As a result, the SoundWire bus can not be used anymore. This patch suggests to finish the runtime suspend process even if we fail to stop sdw bus clock. In the case where we do a hardware reset, the codecs will be reconfigured completely. In the case where we use the regular clock stop, the codecs keep their state and worst case will fall off the bus and reattach. The only drawback is that the power consumption may be higher and device-initiated interrupts may be lost, but at least audio function can still work after resume. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/soundwire/intel.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 10b60b7b17bb..a2d5cdaa9998 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1673,10 +1673,12 @@ static int __maybe_unused intel_suspend_runtime(struct device *dev) } else if (clock_stop_quirks & SDW_INTEL_CLK_STOP_BUS_RESET || !clock_stop_quirks) { + bool wake_enable = true; + ret = sdw_cdns_clock_stop(cdns, true); if (ret < 0) { dev_err(dev, "cannot enable clock stop on suspend\n"); - return ret; + wake_enable = false; } ret = sdw_cdns_enable_interrupt(cdns, false); @@ -1691,7 +1693,7 @@ static int __maybe_unused intel_suspend_runtime(struct device *dev) return ret; } - intel_shim_wake(sdw, true); + intel_shim_wake(sdw, wake_enable); } else { dev_err(dev, "%s clock_stop_quirks %x unsupported\n", __func__, clock_stop_quirks); -- 2.17.1

2 1

[PATCH] ASoC: rt5682: remove LDO2 power supply widget
by shumingf＠realtek.com 19 Jan '21

19 Jan '21

From: Shuming Fan <shumingf(a)realtek.com> The application circuit shall provide MICVDD power. Hence, the codec driver doesn't need to enable LDO2. This patch also reduces power consumption while the recording from the headset. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 42ad5bea86b5..81a4b115b538 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1630,8 +1630,6 @@ static const struct snd_kcontrol_new rt5682_adcdat_pin_ctrl = SOC_DAPM_ENUM("ADCDAT", rt5682_adcdat_pin_enum); static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { - SND_SOC_DAPM_SUPPLY("LDO2", RT5682_PWR_ANLG_3, RT5682_PWR_LDO2_BIT, - 0, NULL, 0), SND_SOC_DAPM_SUPPLY("PLL1", RT5682_PWR_ANLG_3, RT5682_PWR_PLL_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("PLL2B", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2B_BIT, @@ -1854,8 +1852,6 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"CLKDET SYS", NULL, "CLKDET"}, - {"IN1P", NULL, "LDO2"}, - {"BST1 CBJ", NULL, "IN1P"}, {"RECMIX1L", "CBJ Switch", "BST1 CBJ"}, -- 2.29.0

2 1

snd_pcm_extplug_callback_t transfer callback and buffering
by Arsen Arsenović 18 Jan '21

18 Jan '21

Hello! I am writing an external noise cancellation plugin for ALSA, based on rnnoise, but I've hit a bit of a roadblock: rnnoise processes chunks of 480 floats at a time and does no buffering, but alsa expects the transfer callback to take all the data from the input and put it onto the output, with no way to signal that any buffering is or should be happening. I have tried to return a multiple of 480 from the transfer callback and keep a buffer with the leftover data, to be prepended to the next payload, but it looks like ALSA will, when a plugin returns too little data, just send back the same non-processed data in a new call, which breaks this idea. Any way I can work around these limitations? Another solution could be for external plugins to report the amount they consumed and output, rather than a single number for both, but I am not aware of how to do that. -- Arsen Arsenović

1 0

[GIT PULL] ASoC fixes for v5.11-rc4
by Mark Brown 18 Jan '21

18 Jan '21

The following changes since commit 7c53f6b671f4aba70ff15e1b05148b10d58c2837: Linux 5.11-rc3 (2021-01-10 14:34:50 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.11-rc4 for you to fetch changes up to 7505c06dabb5e814bda610c8d83338544f15db45: MAINTAINERS: update qcom ASoC drivers list (2021-01-15 17:25:35 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.11 A few more fixes for v5.11, mostly around HDA jack detection, plus a couple of updates to the MAINTAINERS entries. ---------------------------------------------------------------- Kai Vehmanen (1): ASoC: SOF: Intel: fix page fault at probe if i915 init fails Kai-Heng Feng (3): ASoC: SOF: Intel: hda: Resume codec to do jack detection ASoC: SOF: Intel: hda: Modify existing helper to disable WAKEEN ASoC: SOF: Intel: hda: Avoid checking jack on system suspend Mark Brown (1): Merge v5.11-rc3 Mauro Carvalho Chehab (1): MAINTAINERS: update references to stm32 audio bindings Srinivas Kandagatla (2): MAINTAINERS: update maintainers of qcom audio MAINTAINERS: update qcom ASoC drivers list Stephan Gerhold (1): ASoC: hdmi-codec: Fix return value in hdmi_codec_set_jack() .../bindings/sound/ti,j721e-cpb-audio.yaml | 4 +- .../bindings/sound/ti,j721e-cpb-ivi-audio.yaml | 4 +- MAINTAINERS | 18 +++++--- sound/soc/amd/raven/pci-acp3x.c | 16 ++----- sound/soc/amd/renoir/rn-pci-acp3x.c | 14 ++++++ sound/soc/atmel/Kconfig | 4 +- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/hdmi-codec.c | 2 +- sound/soc/codecs/max98373-i2c.c | 20 +++++++++ sound/soc/codecs/max98373-sdw.c | 20 +++++++++ sound/soc/codecs/max98373.c | 34 +++++++++++++-- sound/soc/codecs/max98373.h | 8 ++++ sound/soc/codecs/rt711.c | 6 +++ sound/soc/fsl/imx-hdmi.c | 3 +- sound/soc/intel/boards/haswell.c | 1 + sound/soc/intel/skylake/cnl-sst.c | 1 + sound/soc/meson/axg-tdm-interface.c | 14 +++++- sound/soc/meson/axg-tdmin.c | 13 +----- sound/soc/qcom/lpass-cpu.c | 20 +-------- sound/soc/qcom/lpass-platform.c | 50 +++++++++++++++------- sound/soc/sh/rcar/adg.c | 18 ++++---- sound/soc/soc-dapm.c | 1 + sound/soc/sof/Kconfig | 2 +- sound/soc/sof/intel/hda-codec.c | 37 ++++++++-------- sound/soc/sof/intel/hda-dsp.c | 9 ++-- sound/soc/sof/intel/hda.h | 2 +- 26 files changed, 217 insertions(+), 106 deletions(-)

2 1

[PATCH] ALSA: hda: Balance runtime/system PM if direct-complete is disabled
by Kai-Heng Feng 18 Jan '21

18 Jan '21

HDA controller can't be runtime-suspended after commit 215a22ed31a1 ("ALSA: hda: Refactor codjc PM to use direct-complete optimization"), which enables direct-complete for HDA codec. The HDA codec driver didn't expect direct-complete will be disabled after it returns a positive value from prepare() callback. However, there are some places that PM core can disable direct-complete. For instance, system hibernation or when codec has subordinates like LEDs. So if a device is prepared for direct-complete but PM core still calls codec's suspend or freeze callback, resume the device to keep PM operations balanced. Reported-by: Kenneth R. Crudup <kenny(a)panix.com> Fixes: 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization") Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_codec.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 687216e74526..0afbced979df 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2997,6 +2997,9 @@ static void hda_codec_pm_complete(struct device *dev) static int hda_codec_pm_suspend(struct device *dev) { + if (pm_runtime_status_suspended(dev)) + pm_runtime_resume(dev); + dev->power.power_state = PMSG_SUSPEND; return hda_codec_suspend(dev); } @@ -3009,6 +3012,9 @@ static int hda_codec_pm_resume(struct device *dev) static int hda_codec_pm_freeze(struct device *dev) { + if (pm_runtime_status_suspended(dev)) + pm_runtime_resume(dev); + dev->power.power_state = PMSG_FREEZE; return hda_codec_suspend(dev); } -- 2.29.2

2 2

[PATCH] ALSA: usb-audio: Add support for Pioneer DJM-750
by Olivia Mackintosh 18 Jan '21

18 Jan '21

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> --- 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 521cc846d9d9..e7216d0b860d 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 c8a4bdf18207..98b31c3160ad 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 -- 2.30.0

2 1

[PATCH] ASoC: wm_adsp: Fix control name parsing for multi-fw
by James Schulman 18 Jan '21

18 Jan '21

When switching between firmware types, the wrong control can be selected when requesting control in kernel API. Use the currently selected DSP firwmare type to select the proper mixer control. Signed-off-by: James Schulman <james.schulman(a)cirrus.com> --- sound/soc/codecs/wm_adsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 1fc7bc1970ea..262c2db4f81c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2033,11 +2033,14 @@ static struct wm_coeff_ctl *wm_adsp_get_ctl(struct wm_adsp *dsp, unsigned int alg) { struct wm_coeff_ctl *pos, *rslt = NULL; + const char *fw_txt = wm_adsp_fw_text[dsp->fw]; list_for_each_entry(pos, &dsp->ctl_list, list) { if (!pos->subname) continue; if (strncmp(pos->subname, name, pos->subname_len) == 0 && + strncmp(pos->fw_name, fw_txt, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 0 && pos->alg_region.alg == alg && pos->alg_region.type == type) { rslt = pos; -- 2.17.1

3 2

[PATCH] ASoC: dt-bindings: mt8192-mt6359: Fix indentation
by Jonathan Neuschäfer 18 Jan '21

18 Jan '21

The items of the 'maintainers' list are indented with three spaces. Use the usual two spaces instead, for consistency and to silence yamllint. Signed-off-by: Jonathan Neuschäfer <j.neuschaefer(a)gmx.net> --- .../bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml index bf8c8ba25009d..54650823b29a4 100644 --- a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml +++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml @@ -7,8 +7,8 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Mediatek MT8192 with MT6359, RT1015 and RT5682 ASoC sound card driver maintainers: - - Jiaxin Yu <jiaxin.yu(a)mediatek.com> - - Shane Chien <shane.chien(a)mediatek.com> + - Jiaxin Yu <jiaxin.yu(a)mediatek.com> + - Shane Chien <shane.chien(a)mediatek.com> description: This binding describes the MT8192 sound card. -- 2.29.2

2 1

[RESEND PATCH] ASoC: audio-graph-card: Drop remote-endpoint as required property
by Sameer Pujar 18 Jan '21

18 Jan '21

The remote-endpoint may not be available if it is part of some pluggable module. One such example would be an audio card, the Codec endpoint will not be available until it is plugged in. Hence drop 'remote-endpoint' as a required property. Cc: Rob Herring <robh(a)kernel.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- Documentation/devicetree/bindings/sound/audio-graph-port.yaml | 3 --- 1 file changed, 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index 2005014..766e910 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -71,9 +71,6 @@ properties: description: CPU to Codec rate channels. $ref: /schemas/types.yaml#/definitions/uint32 - required: - - remote-endpoint - ports: description: multi OF-Graph subnode type: object -- 2.7.4

2 1

[PATCH] dt-bindings: sound: intel, keembay-i2s: Fix dt binding errors
by Michael Sit Wei Hong 18 Jan '21

18 Jan '21

Fix devicetree binding errors caused by newly added parameters Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit(a)intel.com> --- .../devicetree/bindings/sound/intel,keembay-i2s.yaml | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml index e0658f122cbb..dba25c33f0b0 100644 --- a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml @@ -47,13 +47,11 @@ properties: dmas: items: - - description: DMA controller phandle and DMA channel - for TX and RX + - description: DMA TX channel + - description: DMA RX channel dma-names: items: - - description: "tx" for the transmit channel - "rx" for the receive channel - const: tx - const: rx -- 2.17.1

2 2

[PATCH] ASoC: SOF: Intel: initial support to AlderLake-P
by Kai Vehmanen 18 Jan '21

18 Jan '21

Add PCI id for the AlderLake-P. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/sof/sof-pci-dev.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 63b989e3ec40..9c3cb719e268 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -512,6 +512,8 @@ static const struct pci_device_id sof_pci_ids[] = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_ALDERLAKE) { PCI_DEVICE(0x8086, 0x7ad0), .driver_data = (unsigned long)&adls_desc}, + { PCI_DEVICE(0x8086, 0x51c8), + .driver_data = (unsigned long)&tgl_desc}, #endif { 0, } }; -- 2.29.2

4 4

[PATCH v2] ASoC: SOF: Intel: initial support to AlderLake-P
by Kai Vehmanen 18 Jan '21

18 Jan '21

Add PCI id for the AlderLake-P. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/sof/sof-pci-dev.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) V2 changes: - fix compile error if CONFIG_SND_SOC_SOF_TIGERLAKE=n and CONFIG_SND_SOC_SOF_ALDERLAKE is enabled diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 63b989e3ec40..ebd58d49d570 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -213,7 +213,7 @@ static const struct sof_dev_desc icl_desc = { }; #endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) +#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) || IS_ENABLED(CONFIG_SND_SOC_SOF_ALDERLAKE) static const struct sof_dev_desc tgl_desc = { .machines = snd_soc_acpi_intel_tgl_machines, .alt_machines = snd_soc_acpi_intel_tgl_sdw_machines, @@ -230,7 +230,9 @@ static const struct sof_dev_desc tgl_desc = { .nocodec_tplg_filename = "sof-tgl-nocodec.tplg", .ops = &sof_tgl_ops, }; +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) static const struct sof_dev_desc tglh_desc = { .machines = snd_soc_acpi_intel_tgl_machines, .alt_machines = snd_soc_acpi_intel_tgl_sdw_machines, @@ -512,6 +514,8 @@ static const struct pci_device_id sof_pci_ids[] = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_ALDERLAKE) { PCI_DEVICE(0x8086, 0x7ad0), .driver_data = (unsigned long)&adls_desc}, + { PCI_DEVICE(0x8086, 0x51c8), + .driver_data = (unsigned long)&tgl_desc}, #endif { 0, } }; -- 2.29.2

3 4

chtnau8824: Add support for using the SOF driver
by GitHub pull_request - opened 18 Jan '21

18 Jan '21

alsa-project/alsa-ucm-conf pull request #74 was opened from perexg: The old (and currently the default) SST driver uses TDM 4 slots 24 bit as wire format to the codec. Where as the new SOF driver uses standard I2S 2 channel 24 bit. Normally this should not impact the UCM settings, but on the NAU8824 the "DAC Right Channel Source" mixer setting must be set to 1 when using TDM 4 slots and to 0 when using I2S 2 channel mode. Getting this wrong (in either case) results in the right channel not outputting any sound. This commit introduces a RightOutputChannel variable which gets set to 0/1 depending on the driver and then uses that for the "DAC Right Channel Source" mixer setting so that we do the right thing depending on the driver. This has been tested on the following devices: Medion E2215T: Stereo speakers, analog mic Medion E2228T: Stereo speakers, stereo digital mics Cube iWork 8 Air: Mono speaker, analog mic Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/74 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/74.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/3] ALSA: usb-audio: Yet more fixes for Pioneer devices
by Takashi Iwai 18 Jan '21

18 Jan '21

Hi, the Pioneer UAC1 devices still didn't work with the recent changes, and after debugging sessions, a few issues were discovered. Below are the patches to cover those errors. Takashi === Takashi Iwai (3): ALSA: usb-audio: Fix UAC1 rate setup for secondary endpoints ALSA: usb-audio: Set sample rate for all sharing EPs on UAC1 ALSA: usb-audio: Avoid implicit feedback on Pioneer devices sound/usb/clock.c | 21 ++++++--------------- sound/usb/endpoint.c | 9 +++++++++ sound/usb/implicit.c | 17 +++++++++-------- sound/usb/quirks.c | 28 ---------------------------- 4 files changed, 24 insertions(+), 51 deletions(-) -- 2.26.2

2 4

[PATCH v2 0/5] MFD/ASoC: Add support for Intel Bay Trail boards with WM5102 codec
by Hans de Goede 17 Jan '21

17 Jan '21

Hi All, Here is v2 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 biggest difference from v1 is that, based on all the discussions about the jack-detect stuff, I've decided to split this into 2 series. One series adding the basic support for this setup, which is this series. And a second series which reworks the extcon driver into an arizona-jackdet library and then modifies the codec drivers to use that directly, replacing the old separate extcon child-device and extcon-driver. The are 2 reasons for the split: 1. With the new jack-det rework, the series really address 2 separate (but related) enhancements. 2. I expect this series to be ready for merging, while the jack-det stuff likely will need a couple more revisions. Other then the split there are some minor changes addressing various review comments, see the individual patch changelogs. The MFD and ASoC parts can be merged independent from each-other (both must be merged to get working sound on these boards, but that is only a runtime dependency and a part missing won't have any bad side-effects). Or the entire series could be merged through the MFD tree if people prefer that. Regards, Hans

2 8

[PATCH 00/14] MFD/extcon/ASoC: Add support for Intel Bay Trail boards with WM5102 codec
by Hans de Goede 16 Jan '21

16 Jan '21

Hi All, This patch series adds support for Intel Bay Trail based device which use a WM5102 codec for audio output/input. This was developed and tested on a Lenovo Yoga Tablet 1051L. This series consists of 3 parts: 1. Arizona MFD drv patches for ACPI bindings, better jack-detect support and misc. fixes 2. extcon-arizona driver fixes and improved jack reporting (this depends on the MFD changes) 3. ASoC patches in the form of a quirk for BYTCR detecting, a new machine driver for BYT + WM5102 and jack-detect support for the new machine driver (which again depends on the MFD changes). Given that 2. and 3. depend on the MFD changes I believe that it is best if all patches in this series are merged through the MFD tree (once reviewed and acked) and then Lee can provide a immutable branch for the ASoC and extcon maintainers to merge into their trees. I have a patch with matching UCM profile changes available here: https://github.com/jwrdegoede/alsa-ucm-conf/commit/316109e7814926ba984322c1… This series + the UCM profile has been tested with both the SST and SOF ASoC drivers for BYT devices. Regards, Hans BugLink: https://github.com/thesofproject/linux/issues/2485

7 52

[PATCH 1/2] ASoC: qcom: lpass: Fix hardcoded SC7810 DAI IDs
by Stephan Gerhold 15 Jan '21

15 Jan '21

LPASS is broken on MSM8916/DragonBoard 410c since Linux 5.10. Attempting to play HDMI audio results in: ADV7533: lpass_platform_pcmops_hw_params: invalid interface: 3 ADV7533: lpass_platform_pcmops_trigger: invalid 3 interface apq8016-lpass-cpu 7708000.audio-controller: ASoC: error at soc_component_trigger on 7708000.audio-controller: -22 Attempting to record analog audio results in: Unable to handle kernel NULL pointer dereference at virtual address 00000000000001e4 Internal error: Oops: 96000004 [#1] PREEMPT SMP CPU: 1 PID: 1568 Comm: arecord Not tainted 5.11.0-rc3 #20 Hardware name: Qualcomm Technologies, Inc. APQ 8016 SBC (DT) pc : regmap_write+0x14/0x80 lr : lpass_platform_pcmops_open+0xd8/0x210 [snd_soc_lpass_platform] Call trace: regmap_write+0x14/0x80 lpass_platform_pcmops_open+0xd8/0x210 [snd_soc_lpass_platform] snd_soc_component_open+0x2c/0x94 ... The problem was introduced in commit 7cb37b7bd0d3 ("ASoC: qcom: Add support for lpass hdmi driver"). The mistake made there is that lpass.h now contains #include <dt-bindings/sound/sc7180-lpass.h> and then the SC7810 DAI IDs are hardcoded all over lpass-cpu.c and lpass-platform.c. But apq8016 and ipq806x have completely different DAI IDs so now MI2S_QUATERNARY (HDMI) is invalid and MI2S_TERTIARY (= LPASS_DP_RX in sc7180-lpass.h) is treated as HDMI port. Therefore, LPASS is broken on all platforms except SC7810. This commit fixes the problem by removing the include from lpass.h. To check if a DAI is an HDMI port, we compare the dai_id with a new "hdmi_dai" variable in lpass_variant. For SC7180 this is set to LPASS_DP_RX (as before), for all other platform to some unrealistic value (UINT_MAX). Cc: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Fixes: 7cb37b7bd0d3 ("ASoC: qcom: Add support for lpass hdmi driver") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- Srinivas mentioned a potential different fix here: https://lore.kernel.org/alsa-devel/4b34bd4f-e7bc-84f9-5e8a-b2348c17b7aa@lin… Instead of this patch, we could change the dt-bindings for LPASS, so that all SoCs use consistent DAI IDs. In general, I think this might be cleaner, especially in case more special DAIs are added in the future. However, when I made this patch (before Srinivas mentioned it) I tried to avoid changing the dt-bindings because: - Changing dt-bindings after they are added is generally discouraged. but more importantly: - lpass-ipq806x.c does not seem to have PRIMARY, SECONDARY, ... but something completely different. I know nothing about that platform so I don't know how to handle it. --- sound/soc/qcom/lpass-apq8016.c | 1 + sound/soc/qcom/lpass-cpu.c | 9 ++-- sound/soc/qcom/lpass-ipq806x.c | 1 + sound/soc/qcom/lpass-lpaif-reg.h | 2 +- sound/soc/qcom/lpass-platform.c | 77 +++++++++++--------------------- sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 4 +- 7 files changed, 39 insertions(+), 56 deletions(-) diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 8507ef8f6679..40e9c66a74a9 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -273,6 +273,7 @@ static struct lpass_variant apq8016_data = { .num_clks = 2, .dai_driver = apq8016_lpass_cpu_dai_driver, .num_dai = ARRAY_SIZE(apq8016_lpass_cpu_dai_driver), + .hdmi_dai = LPASS_HDMI_DAI_UNSUPPORTED, .dai_osr_clk_names = (const char *[]) { "mi2s-osr-clk0", "mi2s-osr-clk1", diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c5e99c2d89c7..b22992050646 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -707,7 +707,8 @@ static unsigned int of_lpass_cpu_parse_sd_lines(struct device *dev, } static void of_lpass_cpu_parse_dai_data(struct device *dev, - struct lpass_data *data) + struct lpass_data *data, + struct lpass_variant *variant) { struct device_node *node; int ret, id; @@ -724,7 +725,7 @@ static void of_lpass_cpu_parse_dai_data(struct device *dev, dev_err(dev, "valid dai id not found: %d\n", ret); continue; } - if (id == LPASS_DP_RX) { + if (id == variant->hdmi_dai) { data->hdmi_port_enable = 1; dev_err(dev, "HDMI Port is enabled: %d\n", id); } else { @@ -766,7 +767,7 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) drvdata->variant = (struct lpass_variant *)match->data; variant = drvdata->variant; - of_lpass_cpu_parse_dai_data(dev, drvdata); + of_lpass_cpu_parse_dai_data(dev, drvdata, variant); res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "lpass-lpaif"); @@ -820,7 +821,7 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) for (i = 0; i < variant->num_dai; i++) { dai_id = variant->dai_driver[i].id; - if (dai_id == LPASS_DP_RX) + if (dai_id == variant->hdmi_dai) continue; drvdata->mi2s_osr_clk[dai_id] = devm_clk_get(dev, diff --git a/sound/soc/qcom/lpass-ipq806x.c b/sound/soc/qcom/lpass-ipq806x.c index 92f98b4df47f..a0c7c5eb03f1 100644 --- a/sound/soc/qcom/lpass-ipq806x.c +++ b/sound/soc/qcom/lpass-ipq806x.c @@ -149,6 +149,7 @@ static struct lpass_variant ipq806x_data = { .dai_driver = &ipq806x_lpass_cpu_dai_driver, .num_dai = 1, + .hdmi_dai = LPASS_HDMI_DAI_UNSUPPORTED, .dai_osr_clk_names = (const char *[]) { "mi2s-osr-clk", }, diff --git a/sound/soc/qcom/lpass-lpaif-reg.h b/sound/soc/qcom/lpass-lpaif-reg.h index 405542832e99..332f1b9ba674 100644 --- a/sound/soc/qcom/lpass-lpaif-reg.h +++ b/sound/soc/qcom/lpass-lpaif-reg.h @@ -133,7 +133,7 @@ #define LPAIF_WRDMAPERCNT_REG(v, chan) LPAIF_WRDMA_REG_ADDR(v, 0x14, (chan)) #define LPAIF_INTFDMA_REG(v, chan, reg, dai_id) \ - ((v->dai_driver[dai_id].id == LPASS_DP_RX) ? \ + ((v->dai_driver[dai_id].id == v->hdmi_dai) ? \ LPAIF_HDMI_RDMA##reg##_REG(v, chan) : \ LPAIF_RDMA##reg##_REG(v, chan)) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index d1c248590f3a..140609fe835d 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -128,7 +128,7 @@ static int lpass_platform_pcmops_open(struct snd_soc_component *component, return dma_ch; } - if (cpu_dai->driver->id == LPASS_DP_RX) { + if (cpu_dai->driver->id == v->hdmi_dai) { map = drvdata->hdmiif_map; drvdata->hdmi_substream[dma_ch] = substream; } else { @@ -173,7 +173,7 @@ static int lpass_platform_pcmops_close(struct snd_soc_component *component, unsigned int dai_id = cpu_dai->driver->id; data = runtime->private_data; - if (dai_id == LPASS_DP_RX) + if (dai_id == v->hdmi_dai) drvdata->hdmi_substream[data->dma_ch] = NULL; else drvdata->substream[data->dma_ch] = NULL; @@ -205,7 +205,7 @@ static int lpass_platform_pcmops_hw_params(struct snd_soc_component *component, if (dir == SNDRV_PCM_STREAM_PLAYBACK) { id = pcm_data->dma_ch; - if (dai_id == LPASS_DP_RX) + if (dai_id == v->hdmi_dai) dmactl = drvdata->hdmi_rd_dmactl; else dmactl = drvdata->rd_dmactl; @@ -234,8 +234,7 @@ static int lpass_platform_pcmops_hw_params(struct snd_soc_component *component, return ret; } - switch (dai_id) { - case LPASS_DP_RX: + if (dai_id == v->hdmi_dai) { ret = regmap_fields_write(dmactl->burst8, id, LPAIF_DMACTL_BURSTEN_INCR4); if (ret) { @@ -254,9 +253,7 @@ static int lpass_platform_pcmops_hw_params(struct snd_soc_component *component, dev_err(soc_runtime->dev, "error updating dynbursten field: %d\n", ret); return ret; } - break; - case MI2S_PRIMARY: - case MI2S_SECONDARY: + } else { ret = regmap_fields_write(dmactl->intf, id, LPAIF_DMACTL_AUDINTF(dma_port)); if (ret) { @@ -264,12 +261,8 @@ static int lpass_platform_pcmops_hw_params(struct snd_soc_component *component, ret); return ret; } - - break; - default: - dev_err(soc_runtime->dev, "%s: invalid interface: %d\n", __func__, dai_id); - break; } + switch (bitwidth) { case 16: switch (channels) { @@ -299,22 +292,22 @@ static int lpass_platform_pcmops_hw_params(struct snd_soc_component *component, regval = LPAIF_DMACTL_WPSCNT_ONE; break; case 2: - regval = (dai_id == LPASS_DP_RX ? + regval = (dai_id == v->hdmi_dai ? LPAIF_DMACTL_WPSCNT_ONE : LPAIF_DMACTL_WPSCNT_TWO); break; case 4: - regval = (dai_id == LPASS_DP_RX ? + regval = (dai_id == v->hdmi_dai ? LPAIF_DMACTL_WPSCNT_TWO : LPAIF_DMACTL_WPSCNT_FOUR); break; case 6: - regval = (dai_id == LPASS_DP_RX ? + regval = (dai_id == v->hdmi_dai ? LPAIF_DMACTL_WPSCNT_THREE : LPAIF_DMACTL_WPSCNT_SIX); break; case 8: - regval = (dai_id == LPASS_DP_RX ? + regval = (dai_id == v->hdmi_dai ? LPAIF_DMACTL_WPSCNT_FOUR : LPAIF_DMACTL_WPSCNT_EIGHT); break; @@ -354,7 +347,7 @@ static int lpass_platform_pcmops_hw_free(struct snd_soc_component *component, struct regmap *map; unsigned int dai_id = cpu_dai->driver->id; - if (dai_id == LPASS_DP_RX) + if (dai_id == v->hdmi_dai) map = drvdata->hdmiif_map; else map = drvdata->lpaif_map; @@ -386,7 +379,7 @@ static int lpass_platform_pcmops_prepare(struct snd_soc_component *component, ch = pcm_data->dma_ch; if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - if (dai_id == LPASS_DP_RX) { + if (dai_id == v->hdmi_dai) { dmactl = drvdata->hdmi_rd_dmactl; map = drvdata->hdmiif_map; } else { @@ -456,7 +449,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, ch = pcm_data->dma_ch; if (dir == SNDRV_PCM_STREAM_PLAYBACK) { id = pcm_data->dma_ch; - if (dai_id == LPASS_DP_RX) { + if (dai_id == v->hdmi_dai) { dmactl = drvdata->hdmi_rd_dmactl; map = drvdata->hdmiif_map; } else { @@ -480,8 +473,8 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, "error writing to rdmactl reg: %d\n", ret); return ret; } - switch (dai_id) { - case LPASS_DP_RX: + + if (dai_id == v->hdmi_dai) { ret = regmap_fields_write(dmactl->dyncclk, id, LPAIF_DMACTL_DYNCLK_ON); if (ret) { @@ -504,9 +497,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, LPAIF_IRQ_HDMI_REQ_ON_PRELOAD(ch) | LPAIF_IRQ_HDMI_METADONE | LPAIF_IRQ_HDMI_SDEEP_AUD_DIS(ch)); - break; - case MI2S_PRIMARY: - case MI2S_SECONDARY: + } else { reg_irqclr = LPAIF_IRQCLEAR_REG(v, LPAIF_IRQ_PORT_HOST); val_irqclr = LPAIF_IRQ_ALL(ch); @@ -514,10 +505,6 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, reg_irqen = LPAIF_IRQEN_REG(v, LPAIF_IRQ_PORT_HOST); val_mask = LPAIF_IRQ_ALL(ch); val_irqen = LPAIF_IRQ_ALL(ch); - break; - default: - dev_err(soc_runtime->dev, "%s: invalid %d interface\n", __func__, dai_id); - return -EINVAL; } ret = regmap_write(map, reg_irqclr, val_irqclr); @@ -541,8 +528,8 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, "error writing to rdmactl reg: %d\n", ret); return ret; } - switch (dai_id) { - case LPASS_DP_RX: + + if (dai_id == v->hdmi_dai) { ret = regmap_fields_write(dmactl->dyncclk, id, LPAIF_DMACTL_DYNCLK_OFF); if (ret) { @@ -556,16 +543,10 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, LPAIF_IRQ_HDMI_METADONE | LPAIF_IRQ_HDMI_SDEEP_AUD_DIS(ch)); val_irqen = 0; - break; - case MI2S_PRIMARY: - case MI2S_SECONDARY: + } else { reg_irqen = LPAIF_IRQEN_REG(v, LPAIF_IRQ_PORT_HOST); val_mask = LPAIF_IRQ_ALL(ch); val_irqen = 0; - break; - default: - dev_err(soc_runtime->dev, "%s: invalid %d interface\n", __func__, dai_id); - return -EINVAL; } ret = regmap_update_bits(map, reg_irqen, val_mask, val_irqen); @@ -595,7 +576,7 @@ static snd_pcm_uframes_t lpass_platform_pcmops_pointer( struct regmap *map; unsigned int dai_id = cpu_dai->driver->id; - if (dai_id == LPASS_DP_RX) + if (dai_id == v->hdmi_dai) map = drvdata->hdmiif_map; else map = drvdata->lpaif_map; @@ -645,24 +626,18 @@ static irqreturn_t lpass_dma_interrupt_handler( struct regmap *map; unsigned int dai_id = cpu_dai->driver->id; - switch (dai_id) { - case LPASS_DP_RX: + if (dai_id == v->hdmi_dai) { map = drvdata->hdmiif_map; reg = LPASS_HDMITX_APP_IRQCLEAR_REG(v); val = (LPAIF_IRQ_HDMI_REQ_ON_PRELOAD(chan) | LPAIF_IRQ_HDMI_METADONE | LPAIF_IRQ_HDMI_SDEEP_AUD_DIS(chan)); - break; - case MI2S_PRIMARY: - case MI2S_SECONDARY: + } else { map = drvdata->lpaif_map; reg = LPAIF_IRQCLEAR_REG(v, LPAIF_IRQ_PORT_HOST); val = 0; - break; - default: - dev_err(soc_runtime->dev, "%s: invalid %d interface\n", __func__, dai_id); - return -EINVAL; } + if (interrupts & LPAIF_IRQ_PER(chan)) { rv = regmap_write(map, reg, LPAIF_IRQ_PER(chan) | val); @@ -826,10 +801,11 @@ static void lpass_platform_pcm_free(struct snd_soc_component *component, static int lpass_platform_pcmops_suspend(struct snd_soc_component *component) { struct lpass_data *drvdata = snd_soc_component_get_drvdata(component); + struct lpass_variant *v = drvdata->variant; struct regmap *map; unsigned int dai_id = component->id; - if (dai_id == LPASS_DP_RX) + if (dai_id == v->hdmi_dai) map = drvdata->hdmiif_map; else map = drvdata->lpaif_map; @@ -843,10 +819,11 @@ static int lpass_platform_pcmops_suspend(struct snd_soc_component *component) static int lpass_platform_pcmops_resume(struct snd_soc_component *component) { struct lpass_data *drvdata = snd_soc_component_get_drvdata(component); + struct lpass_variant *v = drvdata->variant; struct regmap *map; unsigned int dai_id = component->id; - if (dai_id == LPASS_DP_RX) + if (dai_id == v->hdmi_dai) map = drvdata->hdmiif_map; else map = drvdata->lpaif_map; diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 85db650c2169..76bff06543ea 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -271,6 +271,7 @@ static struct lpass_variant sc7180_data = { .num_clks = 3, .dai_driver = sc7180_lpass_cpu_dai_driver, .num_dai = ARRAY_SIZE(sc7180_lpass_cpu_dai_driver), + .hdmi_dai = LPASS_DP_RX, .dai_osr_clk_names = (const char *[]) { "mclk0", "null", diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 0195372905ed..2de0632c1fb7 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -12,7 +12,6 @@ #include <linux/compiler.h> #include <linux/platform_device.h> #include <linux/regmap.h> -#include <dt-bindings/sound/sc7180-lpass.h> #include "lpass-hdmi.h" #define LPASS_AHBIX_CLOCK_FREQUENCY 131072000 @@ -20,6 +19,8 @@ #define LPASS_MAX_DMA_CHANNELS (8) #define LPASS_MAX_HDMI_DMA_CHANNELS (4) +#define LPASS_HDMI_DAI_UNSUPPORTED UINT_MAX + #define QCOM_REGMAP_FIELD_ALLOC(d, m, f, mf) \ do { \ mf = devm_regmap_field_alloc(d, m, f); \ @@ -245,6 +246,7 @@ struct lpass_variant { /* SOC specific dais */ struct snd_soc_dai_driver *dai_driver; int num_dai; + unsigned int hdmi_dai; const char * const *dai_osr_clk_names; const char * const *dai_bit_clk_names; -- 2.30.0

2 5

[PATCH v4 0/6] Add support for Rpi4b + Cirrus Lochnagar2 and CS47L15
by Richard Fitzgerald 15 Jan '21

15 Jan '21

This set of patches provides support for using the Raspberry Pi 4b with a Cirrus Logic Lochnagar 2 audio development platform plus Cirrus Logic CS47L15 codec. Patches are needed to audio-graph-card to enable support for setting the component sysclks and plls. These are not specific to CS47L15 - several Cirrus and non-Cirrus codecs also require component clock setup, typically for larger codecs with multiple internal clocking options and clock domains. The codec sysclks and plls cannot be placed under the clock framework because they are typically I2C/SPI/Soundwire-connected peripherals and access to the registers would cause a nested get of the bus clock. The clock framework does not support this and it would result in a deadlock. Richard Fitzgerald (6): of: base: Add of_count_phandle_with_fixed_args() dt-bindings: audio-graph-card: Add plls and sysclks properties ASoC: audio-graph-card: Support setting component plls and sysclks ASoC: madera: Allow codecs to be selected from kernel config ASoC: madera: Export clock config defines to dt-bindings ARM: dts: Add dts for RPi4b + Cirrus Logic Lochnagar2 + CS47L15 .../bindings/sound/audio-graph.yaml | 46 ++++ MAINTAINERS | 1 + arch/arm/boot/dts/Makefile | 1 + .../dts/bcm2711-rpi-4-b-lochnagar-cs47l15.dts | 186 ++++++++++++++ .../boot/dts/bcm2711-rpi-4-b-lochnagar.dtsi | 201 +++++++++++++++ drivers/of/base.c | 73 ++++-- include/dt-bindings/sound/madera.h | 60 +++++ include/linux/of.h | 9 + include/sound/simple_card_utils.h | 25 ++ sound/soc/codecs/Kconfig | 10 +- sound/soc/codecs/madera.h | 56 +---- sound/soc/generic/audio-graph-card.c | 13 + sound/soc/generic/simple-card-utils.c | 238 ++++++++++++++++++ 13 files changed, 836 insertions(+), 83 deletions(-) create mode 100644 arch/arm/boot/dts/bcm2711-rpi-4-b-lochnagar-cs47l15.dts create mode 100644 arch/arm/boot/dts/bcm2711-rpi-4-b-lochnagar.dtsi -- 2.20.1

4 21

[PATCH 0/4] ASoC: qcom: Minor code cleanups for lpass-cpu
by Stephen Boyd 15 Jan '21

15 Jan '21

Here's some minor code cleanups for the lpass-cpu driver. I noticed that it casts away const from the driver data from DT. That's not great but fixing it is a little more involved. I'll get to it later. There's also some questionable clk_get() usage that should probably be clk_get_optional(). For now this should help a little. Cc: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Cc: Srinivasa Rao <srivasam(a)codeaurora.org> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Cheng-Yi Chiang <cychiang(a)chromium.org> Stephen Boyd (4): 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 sound/soc/qcom/lpass-cpu.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) base-commit: 5c8fe583cce542aa0b84adc939ce85293de36e5e -- https://chromeos.dev

3 6

[PATCH 1/2] MAINTAINERS: update maintainers of qcom audio
by Srinivas Kandagatla 15 Jan '21

15 Jan '21

Add myself as maintainer of qcom audio drivers, as Patrick has very little time to look at the patches. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Banajit Goswami <bgoswami(a)codeaurora.org> Acked-by: Patrick Lai <plai(a)codeaurora.org> --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 6eff4f720c72..b1545817c899 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -14512,7 +14512,7 @@ S: Supported F: drivers/crypto/qat/ QCOM AUDIO (ASoC) DRIVERS -M: Patrick Lai <plai(a)codeaurora.org> +M: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> M: Banajit Goswami <bgoswami(a)codeaurora.org> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported -- 2.21.0

2 2

Re: [PATCH 2/2] ASoC: mediatek: mt6359: add MT6359 accdet driver
by Mark Brown 15 Jan '21

15 Jan '21

On Wed, Jan 06, 2021 at 08:19:06PM +0800, Argus Lin wrote: > MT6359 audio codec support accessory detect features, adds MT6359 > accdet driver to support plug detection and key detection. > --- > sound/soc/codecs/Kconfig | 7 + > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/mt6359-accdet.c | 1951 ++++++++++++++++++++++++++++++++++++++ > sound/soc/codecs/mt6359-accdet.h | 136 +++ > sound/soc/codecs/mt6359.h | 1863 +++++++++++++++++++++++++++++++++--- This driver is *huge*. Looking through the code it feels like there's a lot of things that are written with mostly duplicated code that differs only in data so you could shrink things down a lot by refactoring things to have one copy of the code and pass different data into it. > Enable support for the platform which uses MT6359 as > external codec device. > +config SND_SOC_MT6359_ACCDET Missing blank line here. > +++ b/sound/soc/codecs/mt6359-accdet.c > @@ -0,0 +1,1951 @@ > +// SPDX-License-Identifier: GPL-2.0 > +/* > + * Copyright (C) 2021 MediaTek Inc. Please make the entire comment a C++ one so things look more intentional. > +#include "mt6359-accdet.h" > +#include "mt6359.h" > +/* grobal variable definitions */ Spelling mistake and you need more blank lines here. > +#define REGISTER_VAL(x) ((x) - 1) > +#define HAS_CAP(_c, _x) \ > + ({typeof(_c)c = (_c); \ > + typeof(_x)x = (_x); \ > + (((c) & (x)) == (x)); }) These need namepsacing. > +static struct mt63xx_accdet_data *accdet; > + > +static struct head_dts_data accdet_dts; > +struct pwm_deb_settings *cust_pwm_deb; You'd need a *very* good reason to be using global data rather than storing anything in the device's driver data like most drivers. There's extensive use of global data here, and lots of raw pr_ prints rather than dev_ prints as well - this doesn't look like how a Linux driver is supposed to be written. > + > +const struct of_device_id mt6359_accdet_of_match[] = { > + { > + .compatible = "mediatek,mt6359-accdet", > + .data = &mt6359_accdet, Given that this is specific to a particular PMIC why does this need a compatible string? > +/* global function declaration */ > + > +static u64 mt6359_accdet_get_current_time(void) > +{ > + return sched_clock(); > +} It is probably best to remove this wrapper. > +static bool mt6359_accdet_timeout_ns(u64 start_time_ns, u64 timeout_time_ns) > +{ > + u64 cur_time = 0; > + u64 elapse_time = 0; > + > + /* get current tick, ns */ > + cur_time = mt6359_accdet_get_current_time(); > + if (cur_time < start_time_ns) { > + start_time_ns = cur_time; > + /* 400us */ > + timeout_time_ns = 400 * 1000; > + } > + elapse_time = cur_time - start_time_ns; > + > + /* check if timeout */ > + if (timeout_time_ns <= elapse_time) > + return false; > + > + return true; > +} There must be a generic implementation of this already surely? > +static unsigned int check_key(unsigned int v) > +{ This looks a lot like open coding of the functionality of the extcon adc_jack functionality. > +static void send_key_event(unsigned int keycode, unsigned int flag) > +{ > + int report = 0; > + > + switch (keycode) { > + case DW_KEY: > + if (flag != 0) > + report = SND_JACK_BTN_1; What does flag mean? At the very least it needs renaming. > +static void send_status_event(unsigned int cable_type, unsigned int status) > +{ > + int report = 0; This is one of those places that looks like it could be code with different data passed in. > + > + switch (cable_type) { > + case HEADSET_NO_MIC: > + if (status) > + report = SND_JACK_HEADPHONE; > + else > + report = 0; > + snd_soc_jack_report(&accdet->jack, report, SND_JACK_HEADPHONE); > + /* when plug 4-pole out, if both AB=3 AB=0 happen,3-pole plug > + * in will be incorrectly reported, then 3-pole plug-out is > + * reported,if no mantory 4-pole plug-out, icon would be > + * visible. > + */ > + if (status == 0) { > + report = 0; > + snd_soc_jack_report(&accdet->jack, report, SND_JACK_MICROPHONE); > + } > + pr_info("accdet HEADPHONE(3-pole) %s\n", > + status ? "PlugIn" : "PlugOut"); You shouldn't be spamming the logs for normal events like this. > + regmap_read(accdet->regmap, ACCDET_IRQ_ADDR, &val); > + while (val & ACCDET_IRQ_MASK_SFT && > + mt6359_accdet_timeout_ns(cur_time, ACCDET_TIME_OUT)) > + ; This is open coding regmap_read_poll_timeout(), this pattern is repeated in several places. > +static inline void clear_accdet_eint(unsigned int eintid) > +{ > + if ((eintid & PMIC_EINT0) == PMIC_EINT0) { The == part is redundant here, and again this is another place where it feels like there's duplicated code that should be using data. All this interrupt handling code is really extremely difficult to follow, there's *lots* of functions all open coding many individual register bits sometimes redundantly and it's very hard to follow what it's supposed to be doing. I can't help but think that in addition to making things data driven writing more linear code without these abstraction layers would help a lot with comprehensibility. > +static irqreturn_t mtk_accdet_irq_handler_thread(int irq, void *data) > +{ > + accdet_irq_handle(); > + > + return IRQ_HANDLED; > +} Why does this wrapper function exist - AFAICT it's just introducing a bug given that the called function is able to detect spurious interrupts but this unconditionally reports IRQ_HANDLED. > +int mt6359_accdet_init(struct snd_soc_component *component, > + struct snd_soc_card *card) > +{ > + int ret = 0; > + struct mt63xx_accdet_data *priv = > + snd_soc_card_get_drvdata(component->card); > + > + return ret; > +} > +EXPORT_SYMBOL_GPL(mt6359_accdet_init); This is a weird interface, what's going on here? > +int mt6359_accdet_set_drvdata(struct snd_soc_card *card) > +{ > + snd_soc_card_set_drvdata(card, accdet); > + > + return 0; > +} > +EXPORT_SYMBOL_GPL(mt6359_accdet_set_drvdata); This is setting off *massive* alarm bells in that it seems to try to claim the card level driver data for this specific driver, again what's going on here? > +module_init(mt6359_accdet_soc_init); > +module_exit(mt6359_accdet_soc_exit); module_platform_driver()

1 0

[PATCH][next] ALSA: usb-audio: remove initialization of param_period_time_if_needed
by Colin King 15 Jan '21

15 Jan '21

From: Colin Ian King <colin.king(a)canonical.com> Variable param_period_time_if_needed is being initialized with a value that is never read, the assignment is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f71965bf815f..ed654338f6b7 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -934,7 +934,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre { const struct audioformat *fp; unsigned int pt, ptmin; - int param_period_time_if_needed = -1; + int param_period_time_if_needed; int err; runtime->hw.formats = subs->formats; -- 2.29.2

2 1

[PATCH v2 0/5] Clock and reset improvements for Tegra ALSA drivers
by Dmitry Osipenko 15 Jan '21

15 Jan '21

This series improves the handling of clock and reset controls of NVIDA Tegra ALSA drivers. Tegra HDA and AHUB drivers aren't handling resets properly, which needs to be fixed in order to unblock other patches related to fixes of the reset controller driver since HDA/AHUB are bound to fail once reset controller driver will be corrected. In particular ALSA drivers are relying on implicit de-assertion of resets which is done by the tegra-clk driver. It's not the business of the clk driver to touch resets and we need to fix this because it breaks reset/clk programming sequences of other Tegra drivers. Changelog: v2: - Added regcache_sync() to the "ahub: Reset hardware properly" patch, which was missed by accident in v1. - Corrected typo in the format of the error message in "ahub: Use of_reset_control_array_get_exclusive()" patch by s/%p/%pe/. Dmitry Osipenko (5): ALSA: hda/tegra: Use clk_bulk helpers ALSA: hda/tegra: Reset hardware ASoC: tegra: ahub: Use of_reset_control_array_get_exclusive() ASoC: tegra: ahub: Use clk_bulk helpers ASoC: tegra: ahub: Reset hardware properly sound/pci/hda/hda_tegra.c | 86 ++++++++----------------- sound/soc/tegra/tegra30_ahub.c | 113 +++++++++------------------------ sound/soc/tegra/tegra30_ahub.h | 6 +- 3 files changed, 59 insertions(+), 146 deletions(-) -- 2.29.2

1 6

Sound stutters with TC-Helicon GoXLR
by GitHub issues - opened 15 Jan '21

15 Jan '21

alsa-project/alsa-lib issue #113 was opened from Toniob: Hi, I bought a hardware mixer from TC-Helicon. I mainly use it under windows for streaming. But now, I'm trying to make it work under Linux, which I use for work. The hardware is seen as a multi-inputs multi-outputs soundcard under Linux. I use Debian unstable, Gnome (xwayland) and pulseaudio. It's an usb connected hardware (with its own power). The sound was stuttering every 3 or 4 seconds. It was plugged on a usb 3.0 port on the motherboard. But then, I bought a pci-express usb 3.0 card and I plugged the goxlr on it. The stutters are steal present but less often (I'd say on every 30 seconds). I've opened an issue on pulseaudio gitlab but it doesn't seem related to pulse as I've got the issue with alsa only : https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1093 . I know a person, met on the discord of tc-helicon, who has exactly the same symptoms with another distribution. Another fun fact is that the inputs do not stutter. I can record my mic plugged in the goxlr and it never stutters. Issue URL : https://github.com/alsa-project/alsa-lib/issues/113 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: hda: Add Cometlake-R PCI ID
by Kai-Chuan Hsieh 15 Jan '21

15 Jan '21

Add HD Audio Device PCI ID for the Intel Cometlake-R platform Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Kai-Chuan Hsieh <kaichuan.hsieh(a)canonical.com> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 83a4a6290070..313d2c9fb35d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2482,6 +2482,9 @@ static const struct pci_device_id azx_ids[] = { /* CometLake-S */ { PCI_DEVICE(0x8086, 0xa3f0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* CometLake-R */ + { PCI_DEVICE(0x8086, 0xf0c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, -- 2.25.1

2 1

[PATCH] ALSA: dice: add support for Lexicon I-ONIX FW810s
by Takashi Sakamoto 15 Jan '21

15 Jan '21

I-ONIX FW810s was shipped in Lexicon brand of HARMAN International industries, Inc 2009. The model uses TCD2220 ASIC as its communication engine. TCAT general protocol is supported, its extension isn't. This patch adds support for the model with hard-coded stream formats. $ python3 ~/git/linux-firewire-utils/src/crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 04042b91 bus_info_length 4, crc_length 4, crc 11153 404 31333934 bus_name "1394" 408 e0008102 irmc 1, cmc 1, isc 1, bmc 0, cyc_clk_acc 0, max_rec 8 (512) 40c 000fd720 company_id 000fd7 | 410 007d7ecf device_id 20007d7ecf | EUI-64 000fd720007d7ecf root directory ----------------------------------------------------------------- 414 00064c2d directory_length 6, crc 19501 418 03000fd7 vendor 41c 8100000a --> descriptor leaf at 444 420 17000001 model 424 8100000d --> descriptor leaf at 458 428 0c0087c0 node capabilities per IEEE 1394 42c d1000001 --> unit directory at 430 unit directory at 430 ----------------------------------------------------------------- 430 000438f2 directory_length 4, crc 14578 434 12000fd7 specifier id 438 13000001 version 43c 17000001 model 440 8100000d --> descriptor leaf at 474 descriptor leaf at 444 ----------------------------------------------------------------- 444 000489d5 leaf_length 4, crc 35285 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 4c657869 "Lexi" 454 636f6e00 "con" descriptor leaf at 458 ----------------------------------------------------------------- 458 0006594b leaf_length 6, crc 22859 45c 00000000 textual descriptor 460 00000000 minimal ASCII 464 492d4f4e "I-ON" 468 49585f46 "IX_F" 46c 57383130 "W810" 470 53000000 "S" descriptor leaf at 474 ----------------------------------------------------------------- 474 0006594b leaf_length 6, crc 22859 478 00000000 textual descriptor 47c 00000000 minimal ASCII 480 492d4f4e "I-ON" 484 49585f46 "IX_F" 488 57383130 "W810" 48c 53000000 "S" Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/Makefile | 3 ++- sound/firewire/dice/dice-harman.c | 26 ++++++++++++++++++++++++++ sound/firewire/dice/dice.c | 12 ++++++++++++ sound/firewire/dice/dice.h | 1 + 4 files changed, 41 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/dice/dice-harman.c diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index 7a62dafd0f78..9bf7b960a720 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -1,5 +1,6 @@ # SPDX-License-Identifier: GPL-2.0-only snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \ - dice-alesis.o dice-extension.o dice-mytek.o dice-presonus.o + dice-alesis.o dice-extension.o dice-mytek.o dice-presonus.o \ + dice-harman.o obj-$(CONFIG_SND_DICE) += snd-dice.o diff --git a/sound/firewire/dice/dice-harman.c b/sound/firewire/dice/dice-harman.c new file mode 100644 index 000000000000..a8ca00c397e8 --- /dev/null +++ b/sound/firewire/dice/dice-harman.c @@ -0,0 +1,26 @@ +// SPDX-License-Identifier: GPL-2.0 +// dice-harman.c - a part of driver for DICE based devices +// +// Copyright (c) 2021 Takashi Sakamoto +// +// Licensed under the terms of the GNU General Public License, version 2. + +#include "dice.h" + +int snd_dice_detect_harman_formats(struct snd_dice *dice) +{ + int i; + + // Lexicon I-ONYX FW810s supports sampling transfer frequency up to + // 96.0 kHz, 12 PCM channels and 1 MIDI channel in its first tx stream + // , 10 PCM channels and 1 MIDI channel in its first rx stream for all + // of the frequencies. + for (i = 0; i < 2; ++i) { + dice->tx_pcm_chs[0][i] = 12; + dice->tx_midi_ports[0] = 1; + dice->rx_pcm_chs[0][i] = 10; + dice->rx_midi_ports[0] = 1; + } + + return 0; +} diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 06c94f009dfb..107a81691f0e 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -20,10 +20,12 @@ MODULE_LICENSE("GPL v2"); #define OUI_MYTEK 0x001ee8 #define OUI_SSL 0x0050c2 // Actually ID reserved by IEEE. #define OUI_PRESONUS 0x000a92 +#define OUI_HARMAN 0x000fd7 #define DICE_CATEGORY_ID 0x04 #define WEISS_CATEGORY_ID 0x00 #define LOUD_CATEGORY_ID 0x10 +#define HARMAN_CATEGORY_ID 0x20 #define MODEL_ALESIS_IO_BOTH 0x000001 @@ -56,6 +58,8 @@ static int check_dice_category(struct fw_unit *unit) category = WEISS_CATEGORY_ID; else if (vendor == OUI_LOUD) category = LOUD_CATEGORY_ID; + else if (vendor == OUI_HARMAN) + category = HARMAN_CATEGORY_ID; else category = DICE_CATEGORY_ID; if (device->config_rom[3] != ((vendor << 8) | category) || @@ -388,6 +392,14 @@ static const struct ieee1394_device_id dice_id_table[] = { .model_id = 0x000008, .driver_data = (kernel_ulong_t)snd_dice_detect_presonus_formats, }, + // Lexicon I-ONYX FW810S. + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_HARMAN, + .model_id = 0x000001, + .driver_data = (kernel_ulong_t)snd_dice_detect_harman_formats, + }, { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index 7fbffcab94c2..adc6f7c84460 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -233,5 +233,6 @@ int snd_dice_detect_alesis_mastercontrol_formats(struct snd_dice *dice); int snd_dice_detect_extension_formats(struct snd_dice *dice); int snd_dice_detect_mytek_formats(struct snd_dice *dice); int snd_dice_detect_presonus_formats(struct snd_dice *dice); +int snd_dice_detect_harman_formats(struct snd_dice *dice); #endif -- 2.27.0

2 1

[PATCH] ALSA: usb-audio: Convert the last strlcpy() usage
by Takashi Iwai 15 Jan '21

15 Jan '21

The last remaining usage of strlcpy() in USB-audio driver is the setup of the card longname string. Basically we need to know whether any non-empty string is set or not, and no real length is needed. Refactor the code and use strscpy() instead. After this change, strlcpy() is gone from all sound/* code. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- I forgot to submit this as well in the previous patch, but they are applied independently. sound/usb/card.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 85e79b9ecb08..45407cb81b66 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -514,18 +514,17 @@ static void usb_audio_make_longname(struct usb_device *dev, s = preset->vendor_name; else if (quirk && quirk->vendor_name) s = quirk->vendor_name; + *card->longname = 0; if (s && *s) { - len = strlcpy(card->longname, s, sizeof(card->longname)); + strscpy(card->longname, s, sizeof(card->longname)); } else { /* retrieve the vendor and device strings as longname */ if (dev->descriptor.iManufacturer) - len = usb_string(dev, dev->descriptor.iManufacturer, - card->longname, sizeof(card->longname)); - else - len = 0; + usb_string(dev, dev->descriptor.iManufacturer, + card->longname, sizeof(card->longname)); /* we don't really care if there isn't any vendor string */ } - if (len > 0) { + if (*card->longname) { strim(card->longname); if (*card->longname) strlcat(card->longname, " ", sizeof(card->longname)); -- 2.26.2

1 0

[PATCH] ALSA: usb-audio: Convert remaining strlcpy() to strscpy()
by Takashi Iwai 15 Jan '21

15 Jan '21

USB-audio driver still contains two calls of strlcpy() because the return size is evaluated. Basically it just checks whether the string is copied or not, but since strcpy() may return a negative error code, we should check the negative value and treat as filled. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 85fed017710e..c7da38348035 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -115,11 +115,14 @@ find_map(const struct usbmix_name_map *p, int unitid, int control) static int check_mapped_name(const struct usbmix_name_map *p, char *buf, int buflen) { + int len; + if (!p || !p->name) return 0; buflen--; - return strlcpy(buf, p->name, buflen); + len = strscpy(buf, p->name, buflen); + return len < 0 ? buflen : len; } /* ignore the error value if ignore_ctl_error flag is set */ @@ -151,12 +154,15 @@ static int check_mapped_selector_name(struct mixer_build *state, int unitid, int index, char *buf, int buflen) { const struct usbmix_selector_map *p; + int len; if (!state->selector_map) return 0; for (p = state->selector_map; p->id; p++) { - if (p->id == unitid && index < p->count) - return strlcpy(buf, p->names[index], buflen); + if (p->id == unitid && index < p->count) { + len = strscpy(buf, p->names[index], buflen); + return len < 0 ? buflen : len; + } } return 0; } -- 2.26.2

1 0

[PATCH] ALSA: seq: oss: Fix missing error check in snd_seq_oss_synth_make_info()
by Takashi Iwai 15 Jan '21

15 Jan '21

snd_seq_oss_synth_make_info() didn't check the error code from snd_seq_oss_midi_make_info(), and this leads to the call of strlcpy() with the uninitialized string as the source, which may lead to the access over the limit. Add the proper error check for avoiding the failure. Reported-by: syzbot+e42504ff21cff05a595f(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/oss/seq_oss_synth.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 11554d0412f0..1b8409ec2c97 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -611,7 +611,8 @@ snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_in if (info->is_midi) { struct midi_info minf; - snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf); + if (snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf)) + return -ENXIO; inf->synth_type = SYNTH_TYPE_MIDI; inf->synth_subtype = 0; inf->nr_voices = 16; -- 2.26.2

1 0

KMSAN: uninit-value in snd_seq_oss_synth_make_info (2)
by syzbot 15 Jan '21

15 Jan '21

Hello, syzbot found the following issue on: HEAD commit: 73d62e81 kmsan: random: prevent boot-time reports in _mix_.. git tree: https://github.com/google/kmsan.git master console output: https://syzkaller.appspot.com/x/log.txt?x=10284200d00000 kernel config: https://syzkaller.appspot.com/x/.config?x=2cdf4151c9653e32 dashboard link: https://syzkaller.appspot.com/bug?extid=e42504ff21cff05a595f compiler: clang version 11.0.0 (https://github.com/llvm/llvm-project.git ca2dcbd030eadbf0aa9b660efe864ff08af6e18b) userspace arch: i386 Unfortunately, I don't have any reproducer for this issue yet. IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+e42504ff21cff05a595f(a)syzkaller.appspotmail.com ===================================================== BUG: KMSAN: uninit-value in strlen lib/string.c:576 [inline] BUG: KMSAN: uninit-value in strlcpy+0x8c/0x1c0 lib/string.c:143 CPU: 1 PID: 10203 Comm: syz-executor.3 Not tainted 5.10.0-rc4-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x21c/0x280 lib/dump_stack.c:118 kmsan_report+0xf7/0x1e0 mm/kmsan/kmsan_report.c:118 __msan_warning+0x5f/0xa0 mm/kmsan/kmsan_instr.c:197 strlen lib/string.c:576 [inline] strlcpy+0x8c/0x1c0 lib/string.c:143 snd_seq_oss_synth_make_info+0x381/0x820 sound/core/seq/oss/seq_oss_synth.c:619 snd_seq_oss_synth_info_user sound/core/seq/oss/seq_oss_ioctl.c:24 [inline] snd_seq_oss_ioctl+0x12b4/0x2530 sound/core/seq/oss/seq_oss_ioctl.c:139 odev_ioctl sound/core/seq/oss/seq_oss.c:180 [inline] odev_ioctl_compat+0x247/0x290 sound/core/seq/oss/seq_oss.c:190 __do_compat_sys_ioctl fs/ioctl.c:842 [inline] __se_compat_sys_ioctl+0x53d/0x1100 fs/ioctl.c:793 __ia32_compat_sys_ioctl+0x4a/0x70 fs/ioctl.c:793 do_syscall_32_irqs_on arch/x86/entry/common.c:80 [inline] __do_fast_syscall_32+0x102/0x160 arch/x86/entry/common.c:139 do_fast_syscall_32+0x6a/0xc0 arch/x86/entry/common.c:162 do_SYSENTER_32+0x73/0x90 arch/x86/entry/common.c:205 entry_SYSENTER_compat_after_hwframe+0x4d/0x5c RIP: 0023:0xf7f8c549 Code: b8 01 10 06 03 74 b4 01 10 07 03 74 b0 01 10 08 03 74 d8 01 00 00 00 00 00 00 00 00 00 00 00 00 00 51 52 55 89 e5 0f 34 cd 80 <5d> 5a 59 c3 90 90 90 90 eb 0d 90 90 90 90 90 90 90 90 90 90 90 90 RSP: 002b:00000000f55860cc EFLAGS: 00000296 ORIG_RAX: 0000000000000036 RAX: ffffffffffffffda RBX: 0000000000000004 RCX: 00000000c08c5114 RDX: 0000000020000080 RSI: 0000000000000000 RDI: 0000000000000000 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Local variable ----minf@snd_seq_oss_synth_make_info created at: snd_seq_oss_synth_make_info+0x22a/0x820 sound/core/seq/oss/seq_oss_synth.c:613 snd_seq_oss_synth_make_info+0x22a/0x820 sound/core/seq/oss/seq_oss_synth.c:613 ===================================================== --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

1 0

Question about daifmt of legacy DT on simple-card
by Kuninori Morimoto 15 Jan '21

15 Jan '21

Hi ALSA SoC Cc Jyri For historical reason, simple-card supports legacy DT daifmt settings. Then, I noticed one strange code. -- simple-card-utils.c --- asoc_simple_parse_daifmt(xxx) { (A) daifmt = snd_soc_of_parse_daifmt(...); daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; if (!bitclkmaster && !framemaster) { /* * No dai-link level and master setting was not found from * sound node level, revert back to legacy DT parsing and * take the settings from codec node. */ (B) daifmt = snd_soc_of_parse_daifmt(codec, NULL, NULL, NULL) | (daifmt & ~SND_SOC_DAIFMT_CLOCK_MASK); } ... ... } It rollbacks to legacy DT parsing at (B) if (A) didn't have master settings. Here, (B) re-try to get daifmt, and use "or" with (daifmt & ~CLOCK mask). Why CLOCK mask ? and shouldn't it use mask when "or" ? Otherwise FORMAT and INV part will be duplicated, I think. for example daifmt = (snd_soc_of_parse_daifmt() & SND_SOC_DAIFMT_CLOCK_MASK) | (daifmt & ~SND_SOC_DAIFMT_CLOCK_MASK) I think using snd_soc_of_parse_daifmt() only is very enough at (B), but am I misunderstanding ?? The original code was from b3ca11ff59bc5842b01f13421a17e6d9a8936784 ("ASoC: simple-card: Move dai-link level properties away from dai subnodes") Thank you for your help !! Best regards --- Kuninori Morimoto

2 2

[PATCH 1/2] [v2] ALSA: hda: fix SND_INTEL_DSP_CONFIG dependency
by Arnd Bergmann 14 Jan '21

14 Jan '21

From: Arnd Bergmann <arnd(a)arndb.de> The sof-pci-dev driver fails to link when built into the kernel and CONFIG_SND_INTEL_DSP_CONFIG is set to =m: arm-linux-gnueabi-ld: sound/soc/sof/sof-pci-dev.o: in function `sof_pci_probe': sof-pci-dev.c:(.text+0x1c): undefined reference to `snd_intel_dsp_driver_probe' All other drivers using this interface already use a 'select SND_INTEL_DSP_CONFIG' statement to force the it to be present, so it seems reasonable to do the same here. The stub implementation in the header makes the problem harder to find, as it avoids the link error when SND_INTEL_DSP_CONFIG is completely disabled, without any obvious upsides. Remove these stubs to make it clearer that the driver is in fact needed here. Fixes: 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- v2: fix various build issues in the first version, now passes all randconfig builds I tried --- include/sound/intel-dsp-config.h | 17 ----------------- sound/soc/sof/Kconfig | 2 ++ sound/soc/sof/intel/Kconfig | 4 ++-- sound/soc/sof/intel/byt.c | 2 +- 4 files changed, 5 insertions(+), 20 deletions(-) diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h index d4609077c258..94667e870029 100644 --- a/include/sound/intel-dsp-config.h +++ b/include/sound/intel-dsp-config.h @@ -18,24 +18,7 @@ enum { SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF }; -#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) - int snd_intel_dsp_driver_probe(struct pci_dev *pci); int snd_intel_acpi_dsp_driver_probe(struct device *dev, const u8 acpi_hid[ACPI_ID_LEN]); -#else - -static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) -{ - return SND_INTEL_DSP_DRIVER_ANY; -} - -static inline -int snd_intel_acpi_dsp_driver_probe(struct device *dev, const u8 acpi_hid[ACPI_ID_LEN]) -{ - return SND_INTEL_DSP_DRIVER_ANY; -} - -#endif - #endif diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 031dad5fc4c7..051fd3d27047 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -12,6 +12,7 @@ if SND_SOC_SOF_TOPLEVEL config SND_SOC_SOF_PCI tristate "SOF PCI enumeration support" depends on PCI + select SND_INTEL_DSP_CONFIG select SND_SOC_SOF select SND_SOC_ACPI if ACPI help @@ -23,6 +24,7 @@ config SND_SOC_SOF_PCI config SND_SOC_SOF_ACPI tristate "SOF ACPI enumeration support" depends on ACPI || COMPILE_TEST + select SND_INTEL_DSP_CONFIG select SND_SOC_SOF select SND_SOC_ACPI if ACPI select IOSF_MBI if X86 && PCI diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index ff9266413a06..67365ce0d86d 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -53,7 +53,7 @@ config SND_SOC_SOF_INTEL_COMMON if SND_SOC_SOF_ACPI config SND_SOC_SOF_BAYTRAIL - bool "SOF support for Baytrail, Braswell and Cherrytrail" + tristate "SOF support for Baytrail, Braswell and Cherrytrail" select SND_SOC_SOF_INTEL_ATOM_HIFI_EP select SND_INTEL_DSP_CONFIG help @@ -70,7 +70,7 @@ config SND_SOC_SOF_BAYTRAIL If unsure select "N". config SND_SOC_SOF_BROADWELL - bool "SOF support for Broadwell" + tristate "SOF support for Broadwell" select SND_INTEL_DSP_CONFIG select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_INTEL_HIFI_EP_IPC diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index 65abcca94521..42dba3da1ca3 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -660,7 +660,7 @@ EXPORT_SYMBOL_NS(tng_chip_info, SND_SOC_SOF_MERRIFIELD); #endif /* CONFIG_SND_SOC_SOF_MERRIFIELD */ -#ifdef CONFIG_SND_SOC_SOF_BAYTRAIL +#if IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) static void byt_reset_dsp_disable_int(struct snd_sof_dev *sdev) { /* Disable Interrupt from both sides */ -- 2.29.2

3 8

[PATCH v2 0/2] Enable DMA mode on Intel Keem Bay platform
by Michael Sit Wei Hong 14 Jan '21

14 Jan '21

v2: Update patch to align with latest kernel release. v1: Initial patch version, to enable DMA mode on Intel Keembay platform. Michael Sit Wei Hong (2): dt-bindings: sound: intel, keembay-i2s: Add info for device to use DMA ASoC: Intel: KMB: Enable DMA transfer mode .../bindings/sound/intel,keembay-i2s.yaml | 14 ++ sound/soc/intel/Kconfig | 2 + sound/soc/intel/keembay/kmb_platform.c | 157 ++++++++++++++++-- sound/soc/intel/keembay/kmb_platform.h | 9 + 4 files changed, 167 insertions(+), 15 deletions(-) -- 2.17.1

3 4

USB-Audio/ALC1220: Bump analog Speaker priority over S/PDIF
by GitHub pull_request - opened 14 Jan '21

14 Jan '21

alsa-project/alsa-ucm-conf pull request #73 was opened from MarijnS95: Suggestion from https://github.com/alsa-project/alsa-ucm-conf/pull/25#issuecomment-760141633 The `S/PDIF` port does not have any jack sensing, and with a priority higher than the `Speakers` it will always be selected by default instead of the `Speakers` even if unplugged. Swapping the priorities around allows analog `Speakers` to be selected first, _if_ they are plugged in. Otherwise `S/PDIF` is used. The sections are reordered to adhere to this priority, including some indentation fixes. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/73 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/2] ALSA: Simplify with dma_set_mask_and_coherent()
by Takashi Iwai 14 Jan '21

14 Jan '21

Simple patches that convert the two calls of dma_set_mask*() into the single dma_set_mask_and_coherent() helper. Just a minor cleanup, there should be no functional changes. Takashi === Takashi Iwai (2): ALSA: pci: Simplify with dma_set_mask_and_coherent() ALSA: x86: Simplify with dma_set_mask_and_coherent() sound/pci/ad1889.c | 3 +-- sound/pci/ali5451/ali5451.c | 3 +-- sound/pci/als300.c | 3 +-- sound/pci/als4000.c | 3 +-- sound/pci/au88x0/au88x0.c | 3 +-- sound/pci/aw2/aw2-alsa.c | 3 +-- sound/pci/azt3328.c | 3 +-- sound/pci/ca0106/ca0106_main.c | 3 +-- sound/pci/cs5535audio/cs5535audio.c | 3 +-- sound/pci/ctxfi/cthw20k1.c | 8 ++------ sound/pci/ctxfi/cthw20k2.c | 8 ++------ sound/pci/es1938.c | 3 +-- sound/pci/es1968.c | 3 +-- sound/pci/hda/hda_intel.c | 8 ++------ sound/pci/ice1712/ice1712.c | 3 +-- sound/pci/maestro3.c | 3 +-- sound/pci/sonicvibes.c | 3 +-- sound/pci/trident/trident_main.c | 3 +-- sound/x86/intel_hdmi_audio.c | 3 +-- 19 files changed, 22 insertions(+), 50 deletions(-) -- 2.26.2

1 2

[PATCH] ALSA: hda/realtek - Limit int mic boost on Acer Aspire E5-575T
by Chris Chiu 14 Jan '21

14 Jan '21

The Acer Apire E5-575T laptop with codec ALC255 has a terrible background noise comes from internal mic capture. And the jack sensing dose not work for headset like some other Acer laptops. This patch limits the internal mic boost on top of the existing ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk for Acer Aspire E5-575T. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3c1d2a3fb1a4..60eb8383a704 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6371,6 +6371,7 @@ enum { ALC256_FIXUP_HP_HEADSET_MIC, ALC236_FIXUP_DELL_AIO_HEADSET_MIC, ALC282_FIXUP_ACER_DISABLE_LINEOUT, + ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -7808,6 +7809,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC255_FIXUP_ACER_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7826,6 +7833,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x1025, 0x1094, "Acer Aspire E5-575T", ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1166, "Acer Veriton N4640G", ALC269_FIXUP_LIFEBOOK), -- 2.20.1

2 1

[PATCH v6 1/2] cros_ec_commands: Add EC_CODEC_I2S_RX_RESET
by Yu-Hsuan Hsu 14 Jan '21

14 Jan '21

Add the new command EC_CODEC_I2S_RX_RESET in ec_codec_i2s_rx_subcmd, which is used for resetting the EC codec. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- include/linux/platform_data/cros_ec_commands.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/linux/platform_data/cros_ec_commands.h b/include/linux/platform_data/cros_ec_commands.h index 86376779ab31..95889ada83a3 100644 --- a/include/linux/platform_data/cros_ec_commands.h +++ b/include/linux/platform_data/cros_ec_commands.h @@ -4600,6 +4600,7 @@ enum ec_codec_i2s_rx_subcmd { EC_CODEC_I2S_RX_SET_SAMPLE_DEPTH = 0x2, EC_CODEC_I2S_RX_SET_DAIFMT = 0x3, EC_CODEC_I2S_RX_SET_BCLK = 0x4, + EC_CODEC_I2S_RX_RESET = 0x5, EC_CODEC_I2S_RX_SUBCMD_COUNT, }; -- 2.30.0.284.gd98b1dd5eaa7-goog

2 3

ucm2: add initial configuration for TRX40 Gigabyte Aorus Master Audio
by GitHub pull_request - opened 14 Jan '21

14 Jan '21

alsa-project/alsa-ucm-conf pull request #72 was opened from perexg: BugLink: https://github.com/alsa-project/alsa-ucm-conf/pull/25 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/72 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/72.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Audio over HDMI via USB-C Dock broke recently - module-alsa-card.c: Failed to find a working profile
by GitHub issues - opened 14 Jan '21

14 Jan '21

alsa-project/alsa-lib issue #112 was opened from asgrim: Hi folks, not sure if this is related to #104 since I get the same error, but slightly different characteristics. Running Ubuntu 20.04.1 here. I've got an XPS 17 9700, which is plugged into a Dell WD19TB dock via USB-C. Off that, I have a Dell 34" screen via HDMI, into which my speakers are plugged in. My headphones are plugged directly into the WD19TB dock. I do have a Yeti mic as well, which is plugged via USB into the dock also. I often switch between my speakers/headphones using a Gnome extension (I select either `Headphones (USB Audio)` or HDMI something (can't remember what it was called exactly). However, I noticed recently (maybe last week or two?) I wasn't able to switch. Looked into it today, it's because the HDMI audio output seems completely missing. Does not show in regular Ubuntu sound settings, nor in `pavucontrol`. Checked /var/log/syslog, and found this `Failed to find a working profile` message. Some Googling bought me here, hopefully you lovely folks can help! * `pacmd ls` output is [here](https://gist.github.com/asgrim/1788fcae11cd1d0ced106c5d94199f29#file-… * `pulseaudio -vvvvv` output is [here](https://gist.github.com/asgrim/1788fcae11cd1d0ced106c5d94199f29#file-… Hopefully this is just user error, but this did used to work, and I don't know where to ask for help as I know nothing about Linux audio stuff :D Side note: I'm aware there's an issue with my XPS 17 "onboard" speakers not working, they've never worked; I believe there's a fix out there, but it looks complicated so I have so far avoided it. I'm ok with this as I've got the external speakers anyway. Issue URL : https://github.com/alsa-project/alsa-lib/issues/112 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v11 0/7] Qualcomm's lpass-hdmi ASoC driver to support audio over dp port
by Srinivasa Rao Mandadapu 14 Jan '21

14 Jan '21

These patches are to support audio over DP port on Qualcomm's SC7180 LPASS Asoc. It includes machine driver, cpu driver, platform driver updates for HDMI path support, device tree documention, lpass variant structure optimization and configuration changes. These patches depends on the DP patch series https://patchwork.kernel.org/project/dri-devel/list/?series=332029 https://lore.kernel.org/patchwork/project/lkml/list/?series=464856 changes since V10: -- Moved hdmi regmap functions from lpass-hdmi.c to lpass-cpu.c -- Moved QCOM_REGMAP_FIELD_ALLOC macro from lpass-hdmi.c to lpass.h changes since V9: -- Removed unused structures lpass_hdmi.h changes since V8: -- Removed redundant structure wrapper for reg map field memebrs -- Updated lpass_hdmi_regmap_volatile API with appropriate registers as true and others as false. changes since V7: -- Fixed typo errors -- Created Separate patch for buffer size change changes since V6: -- Removed compile time define flag, which used for enabling HDMI code, based on corresponding config param is included. -- Updated reg map alloc API with reg map bulk API. -- Removed unnecessary line splits changes since V5: -- Removed unused struct regmap *map in lpass_platform_alloc_hdmidmactl_fields. -- DMA alloc and free API signature change in lpass-apq8016.c, lpass-ipq806x.c -- Keeping API "irqreturn_t lpass_platform_hdmiif_irq" under ifdef macro Changes Since v4: -- Updated with single compatible node for both I2S and HDMI. Changes Since v3: -- Removed id in lpass variant structure and used snd_soc_dai_driver id. Changes Since v2: -- Audio buffer size(i.e. LPASS_PLATFORM_BUFFER_SIZE) in lpass-platform.c increased. Changes Since v1: -- Commit messages are updated -- Addressed Rob Herring review comments V Sujith Kumar Reddy (7): ASoC: Add sc7180-lpass binding header hdmi define ASoC: dt-bindings: Add dt binding for lpass hdmi Asoc:qcom:lpass-cpu:Update dts property read API Asoc: qcom: lpass:Update lpaif_dmactl members order ASoC: qcom: Add support for lpass hdmi driver Asoc: qcom: lpass-platform : Increase buffer size ASoC: qcom: sc7180: Add support for audio over DP .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 74 ++-- include/dt-bindings/sound/sc7180-lpass.h | 1 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 4 +- sound/soc/qcom/lpass-cpu.c | 249 ++++++++++++- sound/soc/qcom/lpass-hdmi.c | 258 ++++++++++++++ sound/soc/qcom/lpass-hdmi.h | 102 ++++++ sound/soc/qcom/lpass-ipq806x.c | 4 +- sound/soc/qcom/lpass-lpaif-reg.h | 49 ++- sound/soc/qcom/lpass-platform.c | 395 +++++++++++++++++---- sound/soc/qcom/lpass-sc7180.c | 116 +++++- sound/soc/qcom/lpass.h | 124 ++++++- 13 files changed, 1240 insertions(+), 143 deletions(-) create mode 100644 sound/soc/qcom/lpass-hdmi.c create mode 100644 sound/soc/qcom/lpass-hdmi.h -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 13

[PATCH] ALSA: hda/via: Add minimum mute flag
by Takashi Iwai 14 Jan '21

14 Jan '21

It turned out that VIA codecs also mute the sound in the lowest mixer level. Turn on the dac_min_mute flag to indicate the mute-as-minimum in TLV like already done in Conexant and IDT codecs. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210559 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_via.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 0ab40a8a68fb..834367dd54e1 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -113,6 +113,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->codec_type = VT1708S; spec->gen.indep_hp = 1; spec->gen.keep_eapd_on = 1; + spec->gen.dac_min_mute = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; codec->power_save_node = 1; -- 2.26.2

1 0

[PATCH v2 0/3] ASoC: soc-pcm: cleanup code
by Kuninori Morimoto 14 Jan '21

14 Jan '21

Hi Mark These are v2 of soc-pcm cleanup code. v1 had 6 patches, but I separated into 2 parts. This is 1st part of it. Link: https://lore.kernel.org/r/87360c99kb.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (3): 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() sound/soc/soc-pcm.c | 110 +++++++++++++++++++++++--------------------- 1 file changed, 58 insertions(+), 52 deletions(-) -- 2.25.1

2 5

[PATCH] ASoC: SOF: Intel: fix page fault at probe if i915 init fails
by Kai Vehmanen 13 Jan '21

13 Jan '21

The earlier commit to fix runtime PM in case i915 init fails, introduces a possibility to hit a page fault. snd_hdac_ext_bus_device_exit() is designed to be called from dev.release(). Calling it outside device reference counting, is not safe and may lead to calling the device_exit() function twice. Additionally, as part of ext_bus_device_init(), the device is also registered with snd_hdac_device_register(). Thus before calling device_exit(), the device must be removed from device hierarchy first. Fix the issue by rolling back init actions by calling hdac_device_unregister() and then releasing device with put_device(). This matches with existing code in hdac-ext module. To complete the fix, add handling for the case where hda_codec_load_module() returns -ENODEV, and clean up the hdac_ext resources also in this case. In future work, hdac-ext interface should be extended to allow clients more flexibility to handle the life-cycle of individual devices, beyond just the current snd_hdac_ext_bus_device_remove(), which removes all devices. BugLink: https://github.com/thesofproject/linux/issues/2646 Reported-by: Jaroslav Kysela <perex(a)perex.cz> Fixes: 6c63c954e1c5 ("ASoC: SOF: fix a runtime pm issue in SOF when HDMI codec doesn't work") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Libin Yang <libin.yang(a)intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> --- sound/soc/sof/intel/hda-codec.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 6875fa570c2c..8b0ddc4b8227 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -156,7 +156,8 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, if (!hdev->bus->audio_component) { dev_dbg(sdev->dev, "iDisp hw present but no driver\n"); - goto error; + ret = -ENOENT; + goto out; } hda_priv->need_display_power = true; } @@ -173,24 +174,23 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, * other return codes without modification */ if (ret == 0) - goto error; + ret = -ENOENT; } - return ret; - -error: - snd_hdac_ext_bus_device_exit(hdev); - return -ENOENT; - +out: + if (ret < 0) { + snd_hdac_device_unregister(hdev); + put_device(&hdev->dev); + } #else hdev = devm_kzalloc(sdev->dev, sizeof(*hdev), GFP_KERNEL); if (!hdev) return -ENOMEM; ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev, HDA_DEV_ASOC); +#endif return ret; -#endif } /* Codec initialization */ -- 2.29.2

2 1

[PATCH 1/4] ASoC: SOF: add .shutdown() callback to snd_sof_dsp_ops
by Kai Vehmanen 13 Jan '21

13 Jan '21

From: Keyon Jie <yang.jie(a)linux.intel.com> Add .shutdown() callback to the struct snd_sof_dsp_ops, for doing platform specific actions at shutdown. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/ops.h | 8 ++++++++ sound/soc/sof/sof-priv.h | 4 +++- 2 files changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 4c1f9daaa6e8..f0c9ca8820d2 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -37,6 +37,14 @@ static inline int snd_sof_remove(struct snd_sof_dev *sdev) return 0; } +static inline int snd_sof_shutdown(struct snd_sof_dev *sdev) +{ + if (sof_ops(sdev)->shutdown) + return sof_ops(sdev)->shutdown(sdev); + + return 0; +} + /* control */ /* diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 28d19fa30614..682c4b6d01ef 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -98,9 +98,10 @@ struct snd_sof_pdata; */ struct snd_sof_dsp_ops { - /* probe and remove */ + /* probe/remove/shutdown */ int (*probe)(struct snd_sof_dev *sof_dev); /* mandatory */ int (*remove)(struct snd_sof_dev *sof_dev); /* optional */ + int (*shutdown)(struct snd_sof_dev *sof_dev); /* optional */ /* DSP core boot / reset */ int (*run)(struct snd_sof_dev *sof_dev); /* mandatory */ @@ -462,6 +463,7 @@ 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); int snd_sof_runtime_suspend(struct device *dev); int snd_sof_runtime_resume(struct device *dev); -- 2.29.2

2 4

[PATCH] ALSA: hda: Add AlderLake-P PCI ID and HDMI codec vid
by Kai Vehmanen 13 Jan '21

13 Jan '21

Add HD Audio PCI ID and HDMI codec vendor ID for Intel AlderLake-P. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 3 +++ sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f36ddf1149a6..279739fa3a7d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2507,6 +2507,9 @@ static const struct pci_device_id azx_ids[] = { /* Alderlake-S */ { PCI_DEVICE(0x8086, 0x7ad0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Alderlake-P */ + { PCI_DEVICE(0x8086, 0x51c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 74d246a0dc6d..97adff0cbcab 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4346,6 +4346,7 @@ HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), -- 2.29.2

2 1

USB Wireless device support
by Bastien Nocera 13 Jan '21

13 Jan '21

Hey, I've recently bought a SteelSeries Arctis 1 wireless headset: https://steelseries.com/gaming-headsets/arctis-1-wireless-xbox and wanted to write a kernel driver to export a few features of the device, and needed some help getting started. The receiver shows up as a audio class USB device, with extra features on a separate USB interface.[1] The extra features that can be accessed are: - Presence detection. I wanted to use that so PulseAudio or Pipewire could make routing decisions based on whether the headset is available, or probably better, implementing a "jack detection" emulation for the main audio interface? - A slider to change the sidetone volume - Export the battery level The latter is the only one I know how to do, so I'd be happy with any pointers or examples about the first two items. Cheers [1]: User-space code that accesses those features: https://github.com/Sapd/HeadsetControl/blob/master/src/devices/steelseries_…

2 8

[PATCH 1/3] ASoC: atmel-pdc: Use managed DMA buffer allocation
by Lars-Peter Clausen 13 Jan '21

13 Jan '21

Instead of manually managing its DMA buffers using dma_{alloc,free}_coherent() lets the sound core take care of this using managed buffers. On one hand this reduces the amount of boiler plate code, but the main motivation for the change is to use the shared code where possible. This makes it easier to argue about correctness and that the code does not contain subtle bugs like data leakage or similar. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/atmel/atmel-pcm-pdc.c | 78 ++------------------------------- 1 file changed, 4 insertions(+), 74 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index 704f700013d3..3e7ea2021b46 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -34,86 +34,21 @@ #include "atmel-pcm.h" -static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, - int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = ATMEL_SSC_DMABUF_SIZE; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_coherent(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", - (void *)buf->area, (void *)(long)buf->addr, size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -static int atmel_pcm_mmap(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, - vma->vm_end - vma->vm_start, vma->vm_page_prot); -} - static int atmel_pcm_new(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; int ret; ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); if (ret) return ret; - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - pr_debug("atmel-pcm: allocating PCM playback DMA buffer\n"); - ret = atmel_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } + snd_pcm_set_managed_buffer_all(rtd->pcm, SNDRV_DMA_TYPE_DEV, + card->dev, ATMEL_SSC_DMABUF_SIZE, + ATMEL_SSC_DMABUF_SIZE); - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - pr_debug("atmel-pcm: allocating PCM capture DMA buffer\n"); - ret = atmel_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -static void atmel_pcm_free(struct snd_soc_component *component, - struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } + return 0; } /*--------------------------------------------------------------------------*\ @@ -210,9 +145,6 @@ static int atmel_pcm_hw_params(struct snd_soc_component *component, /* this may get called several times by oss emulation * with different params */ - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - prtd->params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); prtd->params->dma_intr_handler = atmel_pcm_dma_irq; @@ -384,9 +316,7 @@ static const struct snd_soc_component_driver atmel_soc_platform = { .prepare = atmel_pcm_prepare, .trigger = atmel_pcm_trigger, .pointer = atmel_pcm_pointer, - .mmap = atmel_pcm_mmap, .pcm_construct = atmel_pcm_new, - .pcm_destruct = atmel_pcm_free, }; int atmel_pcm_pdc_platform_register(struct device *dev) -- 2.20.1

3 4

[PATCH v4 1/3] ASoC: SOF: Intel: hda: Resume codec to do jack detection
by Kai-Heng Feng 13 Jan '21

13 Jan '21

Instead of queueing jackpoll_work, runtime resume the codec to let it use different jack detection methods based on jackpoll_interval. This partially matches SOF driver's behavior with commit a6e7d0a4bdb0 ("ALSA: hda: fix jack detection with Realtek codecs when in D3"), the difference is SOF unconditionally resumes the codec. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v4: No change. v3: Remove wrong assumption that only Realtek codec is used by SOF. v2: No change. sound/soc/sof/intel/hda-codec.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 6875fa570c2c..df59c79cfdfc 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -93,8 +93,7 @@ void hda_codec_jack_check(struct snd_sof_dev *sdev) * has been recorded in STATESTS */ if (codec->jacktbl.used) - schedule_delayed_work(&codec->jackpoll_work, - codec->jackpoll_interval); + pm_request_resume(&codec->core.dev); } #else void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) {} -- 2.29.2

3 4

[PATCH 0/6] ASoC: soc-pcm: cleanup code
by Kuninori Morimoto 13 Jan '21

13 Jan '21

Hi Mark These are not so important, but for soc-pcm cleanup patches. Kuninori Morimoto (6): 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: cleanup soc_pcm_params_symmetry() ASoC: soc-pcm: setup pcm at one place in soc_new_pcm() sound/soc/soc-pcm.c | 231 +++++++++++++++++--------------------------- 1 file changed, 90 insertions(+), 141 deletions(-) -- 2.25.1 Thank you for your help !! Best regards --- Kuninori Morimoto

2 8

[PATCH] ASoC: cs42l56: fix up error handling in probe
by Dan Carpenter 13 Jan '21

13 Jan '21

There are two issues with this code. The first error path forgot to set the error code and instead returns success. The second error path doesn't clean up. Fixes: 272b5edd3b8f ("ASoC: Add support for CS42L56 CODEC") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/codecs/cs42l56.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index bb9599cc832b..c44a5cdb796e 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1250,6 +1250,7 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "CS42L56 Device ID (%X). Expected %X\n", devid, CS42L56_DEVID); + ret = -EINVAL; goto err_enable; } alpha_rev = reg & CS42L56_AREV_MASK; @@ -1307,7 +1308,7 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client, ret = devm_snd_soc_register_component(&i2c_client->dev, &soc_component_dev_cs42l56, &cs42l56_dai, 1); if (ret < 0) - return ret; + goto err_enable; return 0; -- 2.29.2

2 1

[PATCH] ASoC: qcom: qdsp6: Move frontend AIFs to q6asm-dai
by Stephan Gerhold 13 Jan '21

13 Jan '21

At the moment it is necessary to set up the DAPM routes between front-end AIF<->DAI explicitly in the device tree, e.g. using audio-routing = "MM_DL1", "MultiMedia1 Playback", "MM_DL3", "MultiMedia3 Playback", "MM_DL4", "MultiMedia4 Playback", "MultiMedia2 Capture", "MM_UL2"; This is prone to mistakes and (sadly) there is no clear error if one of these routes is missing. :( Actually, this should not be necessary because the ASoC core normally automatically links AIF<->DAI within snd_soc_dapm_link_dai_widgets(). This is done using the "stname" parameter of SND_SOC_DAPM_AIF_IN/OUT. For SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0), it should create the route from above: MM_DL1 <-> MultiMedia1 Playback. This does not work at the moment because the AIF widget (MM_DL1) and the DAI widget (MultiMedia1 Playback) belong to different DAPM contexts (q6routing / q6asm-dai). Fix this by declaring the AIF widgets in the same driver as the DAIs (q6asm-dai). Now the routes above are created automatically and no longer need to be specified in the device tree. This is also more consistent with the back-end AIFs which are already declared in q6afe-dais instead of q6routing. q6routing should only link the components together using mixers. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 21 +++++++++++++++++++++ sound/soc/qcom/qdsp6/q6routing.c | 18 ------------------ 2 files changed, 21 insertions(+), 18 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c9ac9c1d26c4..9766725c2916 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -1233,6 +1233,25 @@ static void q6asm_dai_pcm_free(struct snd_soc_component *component, } } +static const struct snd_soc_dapm_widget q6asm_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL3", "MultiMedia3 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL7", "MultiMedia7 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, SND_SOC_NOPM, 0, 0), +}; + static const struct snd_soc_component_driver q6asm_fe_dai_component = { .name = DRV_NAME, .open = q6asm_dai_open, @@ -1245,6 +1264,8 @@ static const struct snd_soc_component_driver q6asm_fe_dai_component = { .pcm_construct = q6asm_dai_pcm_new, .pcm_destruct = q6asm_dai_pcm_free, .compress_ops = &q6asm_dai_compress_ops, + .dapm_widgets = q6asm_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(q6asm_dapm_widgets), }; static struct snd_soc_dai_driver q6asm_fe_dais_template[] = { diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 53185e26fea1..0a6b9433f6ac 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -713,24 +713,6 @@ static const struct snd_kcontrol_new mmul8_mixer_controls[] = { Q6ROUTING_TX_MIXERS(MSM_FRONTEND_DAI_MULTIMEDIA8) }; static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { - /* Frontend AIF */ - SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL3", "MultiMedia3 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL7", "MultiMedia7 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, 0, 0, 0), - /* Mixer definitions */ SND_SOC_DAPM_MIXER("HDMI Mixer", SND_SOC_NOPM, 0, 0, hdmi_mixer_controls, -- 2.29.2

2 1

[PATCH 0/2] Enable DMA mode on Intel Keem Bay platform
by Michael Sit Wei Hong 13 Jan '21

13 Jan '21

v1: Initial patch version, to enable DMA mode on Intel Keembay platform. Michael Sit Wei Hong (2): dt-bindings: sound: intel, keembay-i2s: Add info for device to use DMA ASoC: Intel: KMB: Enable DMA transfer mode .../bindings/sound/intel,keembay-i2s.yaml | 14 ++ sound/soc/intel/Kconfig | 2 + sound/soc/intel/keembay/kmb_platform.c | 157 ++++++++++++++++-- sound/soc/intel/keembay/kmb_platform.h | 9 + 4 files changed, 167 insertions(+), 15 deletions(-) -- 2.17.1

4 6

[RFC PATCH v2 0/2] Enable using multiple kcontrol types in a widget
by Jaska Uimonen 13 Jan '21

13 Jan '21

Hi, I'm requesting comments to this small series to enable multiple different kcontrol types for a single widget. Currently asoc allows to define and parse multiple same type kcontrols for single widget. So you can have for example two volume controls in a widget but not for example one byte and one enum control. I personally had this kind of use case where I wanted to set filter coefficients with bytes and something else with an enum in one processing widget. I don't have that good perspective on the asoc history on this part so I don't know is there some reason for the existing logic or is it just ok to change it like in this series. I've been told some drivers use virtual widget for this kind of situation, so they define two widget's (one virtual) with separate controls, but in reality bind the controls into the same component in lower level. Or can we just actually use separate types in one widget? First patch is changing the kcontrol_type uint in dapm widget to a pointer to hold an array of kcontrol types and then creating these different types in a loop in dapm widget load. Second patch is me utilizing this in sof driver. Changes since v1: - don't add kcontrol type to core i.e. to kcontrol and kcontrol_news - extend existing dapm widget kcontrol_type into holding an array of types Jaska Uimonen (2): ASoC: soc-dapm: enable use of multiple control types ASoC: SOF: topology: use individual kcontrol types include/sound/soc-topology.h | 2 +- sound/soc/soc-topology.c | 466 +++++++++++++++++------------------ sound/soc/sof/topology.c | 13 +- 3 files changed, 232 insertions(+), 249 deletions(-) -- 2.24.1

1 2

[RFC PATCH 0/3] Enable using multiple kcontrol types in a widget
by Jaska Uimonen 13 Jan '21

13 Jan '21

Hi, I'm requesting comments to this small series to enable multiple different kcontrol types for a single widget. Currently asoc allows to define and parse multiple same type kcontrols for single widget. So you can have for example two volume controls in a widget but not for example one byte and one enum control. I personally had this kind of use case where I wanted to set filter coefficients with bytes and something else with an enum in one processing widget. I don't have that good perspective on the asoc history on this part so I don't know is there some reason for the existing logic or is it just ok to change it like in this series. I've been told some drivers use virtual widget for this kind of situation, so they define two widget's (one virtual) with separate controls, but in reality bind the controls into the same component in lower level. Or can we just actually use separate types in one widget? First patch is just adding the type field to kcontrol struct. Second patch is a refactoring to take into account the different control types in parsing the topology. So it looks a little bit messy, but is just actually looping through the types in the kcontrol creation. Third patch is me using this in sof driver, so not so directly needed here. I've tested this thing personally up from topology down till the dsp and for me it seems to work, but as said I can't be sure if I'm breaking something else here. Jaska Uimonen (3): ALSA: control: add kcontrol_type to control ASoC: soc-topology: enable use of multiple control types ASoC: SOF: topology: use individual kcontrol types include/sound/control.h | 2 + sound/core/control.c | 2 + sound/soc/soc-topology.c | 462 ++++++++++++++++++--------------------- sound/soc/sof/topology.c | 13 +- 4 files changed, 230 insertions(+), 249 deletions(-) -- 2.24.1

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Re: [Sound-open-firmware] SoF support for 8086:9d71?
by Amadeusz Sławiński 13 Jan '21

13 Jan '21

On 1/13/2021 5:48 AM, Kai-Heng Feng wrote: > On Tue, Jan 12, 2021 at 9:25 PM Amadeusz Sławiński > <amadeuszx.slawinski(a)linux.intel.com> wrote: >> >> On 12/17/2020 9:25 AM, Kai-Heng Feng wrote: >> (...) >>>>> >>>>> Can you try this instead, the dependencies are indeed wrong in the Makefile: >>>> >>>> Ok, compiling a kernel for user to test... >>> >>> Well, this happens: >>> ERROR: modpost: "skl_dsp_set_dma_control" >>> [sound/soc/intel/skylake/snd-soc-skl-ssp-clk.ko] undefined! >>> make[1]: *** [scripts/Makefile.modpost:111: sound/Module.symvers] Error 1 >>> make[1]: *** Deleting file 'sound/Module.symvers' >>> make: *** [Makefile:1709: modules] Error 2 >>> >>> Kai-Heng >>> >>>> >>>> Kai-Heng >>>> >>>>> >>>>> diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile >>>>> index 4e0248d2accc..7c5038803be7 100644 >>>>> --- a/sound/soc/intel/Makefile >>>>> +++ b/sound/soc/intel/Makefile >>>>> @@ -5,7 +5,7 @@ obj-$(CONFIG_SND_SOC) += common/ >>>>> # Platform Support >>>>> obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += atom/ >>>>> obj-$(CONFIG_SND_SOC_INTEL_CATPT) += catpt/ >>>>> -obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += skylake/ >>>>> +obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE_COMMON) += skylake/ >>>>> obj-$(CONFIG_SND_SOC_INTEL_KEEMBAY) += keembay/ >>>>> >>>>> # Machine support >>>>> >>>>> >> >> Hi, >> >> what is the status of this issue? > > Still the same, unresolved. > >> Does it now work with usptream kernel? > > No it doesn't. > Can you try with following patch: diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 4e0248d2accc..7c5038803be7 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -5,7 +5,7 @@ obj-$(CONFIG_SND_SOC) += common/ # Platform Support obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += atom/ obj-$(CONFIG_SND_SOC_INTEL_CATPT) += catpt/ -obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += skylake/ +obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE_COMMON) += skylake/ obj-$(CONFIG_SND_SOC_INTEL_KEEMBAY) += keembay/ # Machine support diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile index dd39149b89b1..1c4649bccec5 100644 --- a/sound/soc/intel/skylake/Makefile +++ b/sound/soc/intel/skylake/Makefile @@ -7,7 +7,7 @@ ifdef CONFIG_DEBUG_FS snd-soc-skl-objs += skl-debug.o endif -obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o +obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE_COMMON) += snd-soc-skl.o #Skylake Clock device support snd-soc-skl-ssp-clk-objs := skl-ssp-clk.o

1 0

[PATCH v5] ASoC: cros_ec_codec: Reset I2S RX when probing
by Yu-Hsuan Hsu 13 Jan '21

13 Jan '21

It is not guaranteed that I2S RX is disabled when the kernel booting. For example, if the kernel crashes while it is enabled, it will keep enabled until the next time EC reboots. Reset I2S RX when probing to fix this issue. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- This patch checks the return value. If it is -ENOPROTOOPT (EC_RES_INVALID_VERSION), it will ask clients to update EC firmware. Previous patches v1: https://patchwork.kernel.org/project/alsa-devel/patch/20200708071117.307070… v2: https://patchwork.kernel.org/project/alsa-devel/patch/20200716170914.362306… v3: https://patchwork.kernel.org/project/alsa-devel/patch/20210106050559.145902… v4: https://patchwork.kernel.org/project/alsa-devel/patch/20210107085942.289152… include/linux/platform_data/cros_ec_commands.h | 1 + sound/soc/codecs/cros_ec_codec.c | 11 +++++++++++ 2 files changed, 12 insertions(+) diff --git a/include/linux/platform_data/cros_ec_commands.h b/include/linux/platform_data/cros_ec_commands.h index 86376779ab31..95889ada83a3 100644 --- a/include/linux/platform_data/cros_ec_commands.h +++ b/include/linux/platform_data/cros_ec_commands.h @@ -4600,6 +4600,7 @@ enum ec_codec_i2s_rx_subcmd { EC_CODEC_I2S_RX_SET_SAMPLE_DEPTH = 0x2, EC_CODEC_I2S_RX_SET_DAIFMT = 0x3, EC_CODEC_I2S_RX_SET_BCLK = 0x4, + EC_CODEC_I2S_RX_RESET = 0x5, EC_CODEC_I2S_RX_SUBCMD_COUNT, }; diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index f33a2a9654e7..d35c57724b45 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -1011,6 +1011,17 @@ static int cros_ec_codec_platform_probe(struct platform_device *pdev) } priv->ec_capabilities = r.capabilities; + /* Reset EC codec i2s rx. */ + p.cmd = EC_CODEC_I2S_RX_RESET; + ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX, + (uint8_t *)&p, sizeof(p), NULL, 0); + if (ret == -ENOPROTOOPT) { + dev_info(dev, + "Command not found. Please update the EC firmware.\n"); + } else if (ret) { + dev_err(dev, "failed to EC_CODEC_I2S_RESET: %d\n", ret); + } + platform_set_drvdata(pdev, priv); ret = devm_snd_soc_register_component(dev, &i2s_rx_component_driver, -- 2.30.0.284.gd98b1dd5eaa7-goog

2 1

[PATCH v2 0/7] PinePhone BT audio bringup
by Samuel Holland 13 Jan '21

13 Jan '21

This series makes use of the additional DAIs recently added to the sun8i-codec driver to add hardware routing for BT SCO (headset) audio on the PinePhone. The BT audio connection is represented by the "dummy" bt-sco codec. The connection to the Quectel EG-25G modem via AIF2 works as well, but I do not include it here because there is no appropriate codec driver in tree. We have been using an out-of-tree "dummy" codec driver for the modem similar to bt-sco, and I'm not sure if such a driver would be desired upstream. Changes from v1: - Fixed DT binding example to follow new binding Arnaud Ferraris (1): arm64: dts: allwinner: pinephone: Set audio card name Samuel Holland (6): ASoC: dt-bindings: sun8i-codec: Increase #sound-dai-cells ARM: dts: sun8i-a33: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Add pinmux nodes for AIF2/AIF3 arm64: dts: allwinner: a64: Allow multiple DAI links arm64: dts: allwinner: pinephone: Add support for Bluetooth audio .../sound/allwinner,sun8i-a33-codec.yaml | 4 +- arch/arm/boot/dts/sun8i-a33.dtsi | 4 +- .../dts/allwinner/sun50i-a64-pinephone.dtsi | 25 +++++++++++++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 37 ++++++++++++++----- 4 files changed, 56 insertions(+), 14 deletions(-) -- 2.26.2

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[PATCH 1/2] cros_ec_commands: Add EC_CODEC_I2S_RX_RESET
by Yu-Hsuan Hsu 13 Jan '21

13 Jan '21

Add the new command EC_CODEC_I2S_RX_RESET in ec_codec_i2s_rx_subcmd, which is used for resetting the EC codec. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- include/linux/platform_data/cros_ec_commands.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/linux/platform_data/cros_ec_commands.h b/include/linux/platform_data/cros_ec_commands.h index 86376779ab31..95889ada83a3 100644 --- a/include/linux/platform_data/cros_ec_commands.h +++ b/include/linux/platform_data/cros_ec_commands.h @@ -4600,6 +4600,7 @@ enum ec_codec_i2s_rx_subcmd { EC_CODEC_I2S_RX_SET_SAMPLE_DEPTH = 0x2, EC_CODEC_I2S_RX_SET_DAIFMT = 0x3, EC_CODEC_I2S_RX_SET_BCLK = 0x4, + EC_CODEC_I2S_RX_RESET = 0x5, EC_CODEC_I2S_RX_SUBCMD_COUNT, }; -- 2.29.2.729.g45daf8777d-goog

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[PATCH v2 2/2] ASoC: rt715:add Mic Mute LED control support
by Perry Yuan 13 Jan '21

13 Jan '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 notifying that SW mute is completed. Then EC will do the HW mute physically. This patch allow codec rt715 driver to ack EC when micmute key pressed through this micphone led control interface like hda_generic provided. Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> Signed-off-by: Limonciello Mario <mario_limonciello(a)dell.com> --- sound/soc/codecs/rt715-sdca.c | 19 ++++++++++++++++++- sound/soc/codecs/rt715-sdca.h | 1 + sound/soc/codecs/rt715.c | 19 ++++++++++++++++++- sound/soc/codecs/rt715.h | 1 + 4 files changed, 38 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index b43ac8559e45..e168ef6efcf5 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> @@ -268,6 +269,7 @@ 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]; @@ -286,7 +288,22 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, if (err < 0) return err; } - +#if IS_ENABLED(CONFIG_DELL_PRIVACY) + /* Privacy LED Trigger State Changed by muted/unmute switch */ + if (mc->invert) { + val0 = ucontrol->value.integer.value[0]; + val1 = ucontrol->value.integer.value[1]; + if (val0 == 1 && val1 == 1) { + rt715->micmute_led = LED_OFF; + ledtrig_audio_set(LED_AUDIO_MICMUTE, + rt715->micmute_led ? LED_ON : LED_OFF); + } else if (val0 == 0 && val1 == 0) { + rt715->micmute_led = LED_ON; + ledtrig_audio_set(LED_AUDIO_MICMUTE, + rt715->micmute_led ? LED_ON : LED_OFF); + } + } +#endif return 0; } diff --git a/sound/soc/codecs/rt715-sdca.h b/sound/soc/codecs/rt715-sdca.h index 840c237895dd..2ab8724ae50b 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; + int micmute_led; }; struct rt715_sdw_stream_data { diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index cdcba70146da..faa4dee6b39a 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> @@ -88,13 +89,29 @@ 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")) loop_cnt = 4; else loop_cnt = 1; - +#if IS_ENABLED(CONFIG_DELL_PRIVACY) + /* Privacy micmute led trigger for muted/unmute switch */ + if (mc->invert) { + val0 = ucontrol->value.integer.value[0]; + val1 = ucontrol->value.integer.value[1]; + if (val0 == 1 && val1 == 1) { + rt715->micmute_led = LED_OFF; + ledtrig_audio_set(LED_AUDIO_MICMUTE, + rt715->micmute_led ? LED_ON : LED_OFF); + } else if (val0 == 0 && val1 == 0) { + rt715->micmute_led = LED_ON; + ledtrig_audio_set(LED_AUDIO_MICMUTE, + rt715->micmute_led ? LED_ON : LED_OFF); + } + } +#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..2d3b5b299514 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; + int micmute_led; }; struct sdw_stream_data { -- 2.25.1

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[PATCH 00/10] Remove support for TX49xx
by Thomas Bogendoerfer 13 Jan '21

13 Jan '21

I couldn't find any buyable product other than reference boards using TX49xx CPUs. And since nobody showed interest in keeping support for it, it's time to remove it. I've split up the removal into seperate parts for different maintainers. So if the patch fits your needs, please take it via your tree or give me an ack so I can apply them the mips-next tree. Thomas Bogendoerfer (10): MIPS: TX49xx: Drop support net: tc35815: Drop support for TX49XX boards net: 8390: Drop support for TX49XX boards spi: txx9: Remove driver dma: tx49 removal mtd: Remove drivers used by TX49xx char: hw_random: Remove tx4939 driver rtc: tx4939: Remove driver ide: tx4938ide: Remove driver ASoC: txx9: Remove driver arch/mips/Kbuild.platforms | 1 - arch/mips/Kconfig | 23 - arch/mips/Makefile | 1 - arch/mips/configs/rbtx49xx_defconfig | 98 --- arch/mips/include/asm/cpu-type.h | 4 - arch/mips/include/asm/cpu.h | 2 +- .../asm/mach-tx49xx/cpu-feature-overrides.h | 26 - arch/mips/include/asm/mach-tx49xx/ioremap.h | 30 - arch/mips/include/asm/mach-tx49xx/kmalloc.h | 7 - .../include/asm/mach-tx49xx/mangle-port.h | 27 - arch/mips/include/asm/mach-tx49xx/spaces.h | 17 - arch/mips/include/asm/mipsregs.h | 6 - arch/mips/include/asm/txx9/boards.h | 10 - arch/mips/include/asm/txx9/generic.h | 9 - arch/mips/include/asm/txx9/rbtx4927.h | 92 --- arch/mips/include/asm/txx9/rbtx4938.h | 145 ---- arch/mips/include/asm/txx9/rbtx4939.h | 142 ---- arch/mips/include/asm/txx9/tx4927.h | 273 -------- arch/mips/include/asm/txx9/tx4927pcic.h | 203 ------ arch/mips/include/asm/txx9/tx4938.h | 312 --------- arch/mips/include/asm/txx9/tx4939.h | 524 --------------- arch/mips/include/asm/vermagic.h | 2 - arch/mips/kernel/cpu-probe.c | 10 - arch/mips/kernel/idle.c | 3 - arch/mips/mm/c-r4k.c | 77 --- arch/mips/mm/page.c | 1 - arch/mips/mm/tlbex.c | 1 - arch/mips/pci/Makefile | 6 - arch/mips/pci/fixup-rbtx4927.c | 73 -- arch/mips/pci/fixup-rbtx4938.c | 53 -- arch/mips/pci/ops-tx4927.c | 524 --------------- arch/mips/pci/pci-tx4927.c | 91 --- arch/mips/pci/pci-tx4938.c | 142 ---- arch/mips/pci/pci-tx4939.c | 107 --- arch/mips/txx9/Kconfig | 89 --- arch/mips/txx9/Makefile | 8 - arch/mips/txx9/Platform | 3 - arch/mips/txx9/generic/7segled.c | 123 ---- arch/mips/txx9/generic/Makefile | 4 - arch/mips/txx9/generic/irq_tx4927.c | 49 -- arch/mips/txx9/generic/irq_tx4938.c | 37 -- arch/mips/txx9/generic/irq_tx4939.c | 216 ------ arch/mips/txx9/generic/mem_tx4927.c | 75 --- arch/mips/txx9/generic/setup.c | 158 +---- arch/mips/txx9/generic/setup_tx4927.c | 337 ---------- arch/mips/txx9/generic/setup_tx4938.c | 485 -------------- arch/mips/txx9/generic/setup_tx4939.c | 568 ---------------- arch/mips/txx9/rbtx4927/Makefile | 2 - arch/mips/txx9/rbtx4927/irq.c | 198 ------ arch/mips/txx9/rbtx4927/prom.c | 42 -- arch/mips/txx9/rbtx4927/setup.c | 380 ----------- arch/mips/txx9/rbtx4938/Makefile | 2 - arch/mips/txx9/rbtx4938/irq.c | 157 ----- arch/mips/txx9/rbtx4938/prom.c | 22 - arch/mips/txx9/rbtx4938/setup.c | 372 ----------- arch/mips/txx9/rbtx4939/Makefile | 2 - arch/mips/txx9/rbtx4939/irq.c | 95 --- arch/mips/txx9/rbtx4939/prom.c | 29 - arch/mips/txx9/rbtx4939/setup.c | 554 --------------- drivers/char/hw_random/Kconfig | 13 - drivers/char/hw_random/Makefile | 1 - drivers/char/hw_random/tx4939-rng.c | 157 ----- drivers/dma/Kconfig | 2 +- drivers/dma/txx9dmac.h | 10 - drivers/ide/Kconfig | 10 - drivers/ide/Makefile | 3 - drivers/ide/tx4938ide.c | 209 ------ drivers/ide/tx4939ide.c | 628 ------------------ drivers/mtd/maps/Kconfig | 6 - drivers/mtd/maps/Makefile | 1 - drivers/mtd/maps/rbtx4939-flash.c | 133 ---- drivers/mtd/nand/raw/Kconfig | 7 - drivers/mtd/nand/raw/Makefile | 1 - drivers/mtd/nand/raw/txx9ndfmc.c | 423 ------------ drivers/net/ethernet/8390/Kconfig | 2 +- drivers/net/ethernet/8390/ne.c | 7 +- drivers/net/ethernet/toshiba/tc35815.c | 29 - drivers/rtc/Kconfig | 7 - drivers/rtc/Makefile | 1 - drivers/rtc/rtc-tx4939.c | 303 --------- drivers/spi/Kconfig | 6 - drivers/spi/Makefile | 1 - drivers/spi/spi-txx9.c | 477 ------------- drivers/watchdog/Kconfig | 2 +- include/linux/platform_data/txx9/ndfmc.h | 28 - sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - 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 -- 93 files changed, 6 insertions(+), 10365 deletions(-) delete mode 100644 arch/mips/configs/rbtx49xx_defconfig delete mode 100644 arch/mips/include/asm/mach-tx49xx/cpu-feature-overrides.h delete mode 100644 arch/mips/include/asm/mach-tx49xx/ioremap.h delete mode 100644 arch/mips/include/asm/mach-tx49xx/kmalloc.h delete mode 100644 arch/mips/include/asm/mach-tx49xx/mangle-port.h delete mode 100644 arch/mips/include/asm/mach-tx49xx/spaces.h delete mode 100644 arch/mips/include/asm/txx9/rbtx4927.h delete mode 100644 arch/mips/include/asm/txx9/rbtx4938.h delete mode 100644 arch/mips/include/asm/txx9/rbtx4939.h delete mode 100644 arch/mips/include/asm/txx9/tx4927.h delete mode 100644 arch/mips/include/asm/txx9/tx4927pcic.h delete mode 100644 arch/mips/include/asm/txx9/tx4938.h delete mode 100644 arch/mips/include/asm/txx9/tx4939.h delete mode 100644 arch/mips/pci/fixup-rbtx4927.c delete mode 100644 arch/mips/pci/fixup-rbtx4938.c delete mode 100644 arch/mips/pci/ops-tx4927.c delete mode 100644 arch/mips/pci/pci-tx4927.c delete mode 100644 arch/mips/pci/pci-tx4938.c delete mode 100644 arch/mips/pci/pci-tx4939.c delete mode 100644 arch/mips/txx9/generic/7segled.c delete mode 100644 arch/mips/txx9/generic/irq_tx4927.c delete mode 100644 arch/mips/txx9/generic/irq_tx4938.c delete mode 100644 arch/mips/txx9/generic/irq_tx4939.c delete mode 100644 arch/mips/txx9/generic/mem_tx4927.c delete mode 100644 arch/mips/txx9/generic/setup_tx4927.c delete mode 100644 arch/mips/txx9/generic/setup_tx4938.c delete mode 100644 arch/mips/txx9/generic/setup_tx4939.c delete mode 100644 arch/mips/txx9/rbtx4927/Makefile delete mode 100644 arch/mips/txx9/rbtx4927/irq.c delete mode 100644 arch/mips/txx9/rbtx4927/prom.c delete mode 100644 arch/mips/txx9/rbtx4927/setup.c delete mode 100644 arch/mips/txx9/rbtx4938/Makefile delete mode 100644 arch/mips/txx9/rbtx4938/irq.c delete mode 100644 arch/mips/txx9/rbtx4938/prom.c delete mode 100644 arch/mips/txx9/rbtx4938/setup.c delete mode 100644 arch/mips/txx9/rbtx4939/Makefile delete mode 100644 arch/mips/txx9/rbtx4939/irq.c delete mode 100644 arch/mips/txx9/rbtx4939/prom.c delete mode 100644 arch/mips/txx9/rbtx4939/setup.c delete mode 100644 drivers/char/hw_random/tx4939-rng.c delete mode 100644 drivers/ide/tx4938ide.c delete mode 100644 drivers/ide/tx4939ide.c delete mode 100644 drivers/mtd/maps/rbtx4939-flash.c delete mode 100644 drivers/mtd/nand/raw/txx9ndfmc.c delete mode 100644 drivers/rtc/rtc-tx4939.c delete mode 100644 drivers/spi/spi-txx9.c delete mode 100644 include/linux/platform_data/txx9/ndfmc.h 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 -- 2.29.2

6 17

[PATCH] ASoC: imx-hdmi: Fix warning of the uninitialized variable ret
by Shengjiu Wang 12 Jan '21

12 Jan '21

From: shengjiu wang <shengjiu.wang(a)nxp.com> When condition ((hdmi_out && hdmi_in) || (!hdmi_out && !hdmi_in)) is true, then goto fail, the uninitialized variable ret will be returned. Signed-off-by: shengjiu wang <shengjiu.wang(a)nxp.com> Reported-by: kernel test robot <lkp(a)intel.com> --- sound/soc/fsl/imx-hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index 2c2a76a71940..ede4a9ad1054 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -164,6 +164,7 @@ static int imx_hdmi_probe(struct platform_device *pdev) if ((hdmi_out && hdmi_in) || (!hdmi_out && !hdmi_in)) { dev_err(&pdev->dev, "Invalid HDMI DAI link\n"); + ret = -EINVAL; goto fail; } -- 2.17.1

5 7

[PATCH] ALSA: hda: fix SND_INTEL_DSP_CONFIG dependency
by Arnd Bergmann 12 Jan '21

12 Jan '21

From: Arnd Bergmann <arnd(a)arndb.de> The sof-pci-dev driver fails to link when built into the kernel and CONFIG_SND_INTEL_DSP_CONFIG is set to =m: arm-linux-gnueabi-ld: sound/soc/sof/sof-pci-dev.o: in function `sof_pci_probe': sof-pci-dev.c:(.text+0x1c): undefined reference to `snd_intel_dsp_driver_probe' All other drivers using this interface already use a 'select SND_INTEL_DSP_CONFIG' statement to force the it to be present, so it seems reasonable to do the same here. The stub implementation in the header makes the problem harder to find, as it avoids the link error when SND_INTEL_DSP_CONFIG is completely disabled, without any obvious upsides. Remove these stubs to make it clearer that the driver is in fact needed here. Fixes: 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- include/sound/intel-dsp-config.h | 17 ----------------- sound/soc/sof/Kconfig | 2 ++ 2 files changed, 2 insertions(+), 17 deletions(-) diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h index d4609077c258..94667e870029 100644 --- a/include/sound/intel-dsp-config.h +++ b/include/sound/intel-dsp-config.h @@ -18,24 +18,7 @@ enum { SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF }; -#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) - int snd_intel_dsp_driver_probe(struct pci_dev *pci); int snd_intel_acpi_dsp_driver_probe(struct device *dev, const u8 acpi_hid[ACPI_ID_LEN]); -#else - -static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) -{ - return SND_INTEL_DSP_DRIVER_ANY; -} - -static inline -int snd_intel_acpi_dsp_driver_probe(struct device *dev, const u8 acpi_hid[ACPI_ID_LEN]) -{ - return SND_INTEL_DSP_DRIVER_ANY; -} - -#endif - #endif diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 031dad5fc4c7..051fd3d27047 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -12,6 +12,7 @@ if SND_SOC_SOF_TOPLEVEL config SND_SOC_SOF_PCI tristate "SOF PCI enumeration support" depends on PCI + select SND_INTEL_DSP_CONFIG select SND_SOC_SOF select SND_SOC_ACPI if ACPI help @@ -23,6 +24,7 @@ config SND_SOC_SOF_PCI config SND_SOC_SOF_ACPI tristate "SOF ACPI enumeration support" depends on ACPI || COMPILE_TEST + select SND_INTEL_DSP_CONFIG select SND_SOC_SOF select SND_SOC_ACPI if ACPI select IOSF_MBI if X86 && PCI -- 2.29.2

6 15

[PATCH v3 1/4] ALSA: hda/realtek: Power up codec when setting LED via COEF and GPIO
by Kai-Heng Feng 12 Jan '21

12 Jan '21

System takes a very long time to suspend after commit 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization"): [ 90.065964] PM: suspend entry (s2idle) [ 90.067337] Filesystems sync: 0.001 seconds [ 90.185758] Freezing user space processes ... (elapsed 0.002 seconds) done. [ 90.188713] OOM killer disabled. [ 90.188714] Freezing remaining freezable tasks ... (elapsed 0.001 seconds) done. [ 90.190024] printk: Suspending console(s) (use no_console_suspend to debug) [ 90.904912] intel_pch_thermal 0000:00:12.0: CPU-PCH is cool [49C], continue to suspend [ 321.262505] snd_hda_codec_realtek ehdaudio0D0: Unable to sync register 0x2b8000. -5 [ 328.426919] snd_hda_codec_realtek ehdaudio0D0: Unable to sync register 0x2b8000. -5 [ 329.490933] ACPI: EC: interrupt blocked That commit keeps the codec suspended during the system suspend. However, led_suspend() for mute and micmute led writes codec register, triggers a pending wake up. The wakeup is then handled in __device_suspend() by the following: - pm_runtime_disable() to handle the wakeup event. - device is no longer is suspended state, direct-complete isn't taken. - pm_runtime_enable() to balance disable_depth. if (dev->power.direct_complete) { if (pm_runtime_status_suspended(dev)) { pm_runtime_disable(dev); if (pm_runtime_status_suspended(dev)) { pm_dev_dbg(dev, state, "direct-complete "); goto Complete; } pm_runtime_enable(dev); } dev->power.direct_complete = false; } Since direct-complete doens't apply anymore, the codec's system suspend routine is used. This doesn't play well with SOF driver. When its runtime resume is called for system suspend, hda_codec_jack_check() schedules jackpoll_work which uses snd_hdac_is_power_on() to check whether codec is suspended. Because of the previous pm_runtime_enable(), snd_hdac_is_power_on() returns false and jackpoll continues to run, and snd_hda_power_up_pm() cannot power up an already suspended codec in multiple attempts, causes the long delay on system suspend. When direct-complete path is taken, snd_hdac_is_power_on() returns true and hda_jackpoll_work() is skipped by accident. This is still not correct, and it will be addressed by later patch. Explicitly runtime resume codec on setting LED to solve the issue. Fixes: 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization") --- v3: New patch. sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3c1d2a3fb1a4..304a7bc89fcd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4164,7 +4164,10 @@ static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask, { if (polarity) enabled = !enabled; + /* temporarily power up/down for setting GPIO */ + snd_hda_power_up_pm(codec); alc_update_gpio_data(codec, mask, !enabled); /* muted -> LED on */ + snd_hda_power_down_pm(codec); } /* turn on/off mute LED via GPIO per vmaster hook */ @@ -4287,8 +4290,10 @@ static void alc_update_coef_led(struct hda_codec *codec, if (polarity) on = !on; /* temporarily power up/down for setting COEF bit */ + snd_hda_power_up_pm(codec); alc_update_coef_idx(codec, led->idx, led->mask, on ? led->on : led->off); + snd_hda_power_down_pm(codec); } /* update mute-LED according to the speaker mute state via COEF bit */ -- 2.29.2

3 6

[PATCH v3 0/3] Dell hardware privacy implementation for dell laptop
by Perry Yuan 12 Jan '21

12 Jan '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 (3): platform/x86: dell-privacy: Add support for Dell hardware privacy x86/platform/dell-privacy-wmi: add document for dell privacy driver ASoC: rt715:add micmute led state control supports .../testing/sysfs-platform-dell-privacy-wmi | 31 ++ drivers/platform/x86/Kconfig | 17 + drivers/platform/x86/Makefile | 4 +- drivers/platform/x86/dell-laptop.c | 25 +- drivers/platform/x86/dell-privacy-acpi.c | 167 +++++++++ drivers/platform/x86/dell-privacy-wmi.c | 320 ++++++++++++++++++ drivers/platform/x86/dell-privacy-wmi.h | 33 ++ drivers/platform/x86/dell-wmi.c | 37 +- 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 + 12 files changed, 650 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] ASoC: SOF: add mutex to protect the dsp_power_state access
by Kai Vehmanen 12 Jan '21

12 Jan '21

From: Keyon Jie <yang.jie(a)linux.intel.com> There could be more than one thread read/write the dsp_power_state simultaneously (e.g. hda_dsp_d0i3_work and sof_ipc_tx_message), add a mutex power_state_access to make sure the access to it is mutually exclusive. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/core.c | 1 + sound/soc/sof/ops.h | 11 ++++++++--- sound/soc/sof/sof-priv.h | 2 ++ 3 files changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index adc7c37145d6..2b85ef5d6092 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -316,6 +316,7 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) INIT_LIST_HEAD(&sdev->route_list); spin_lock_init(&sdev->ipc_lock); spin_lock_init(&sdev->hw_lock); + mutex_init(&sdev->power_state_access); if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) INIT_WORK(&sdev->probe_work, sof_probe_work); diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 95e748b36903..4c1f9daaa6e8 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -208,11 +208,16 @@ static inline int snd_sof_dsp_set_power_state(struct snd_sof_dev *sdev, const struct sof_dsp_power_state *target_state) { + int ret = 0; + + mutex_lock(&sdev->power_state_access); + if (sof_ops(sdev)->set_power_state) - return sof_ops(sdev)->set_power_state(sdev, target_state); + ret = sof_ops(sdev)->set_power_state(sdev, target_state); - /* D0 substate is not supported, do nothing here. */ - return 0; + mutex_unlock(&sdev->power_state_access); + + return ret; } /* debug */ diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 68da8f797403..28d19fa30614 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -375,6 +375,8 @@ struct snd_sof_dev { /* current DSP power state */ struct sof_dsp_power_state dsp_power_state; + /* mutex to protect the dsp_power_state access */ + struct mutex power_state_access; /* Intended power target of system suspend */ enum sof_system_suspend_state system_suspend_target; -- 2.29.2

2 1

[PATCH] ASoC: Intel: cht_bsw_nau8824: Move snd_soc_dai_set_tdm_slot call to cht_codec_fixup
by Hans de Goede 12 Jan '21

12 Jan '21

Move the snd_soc_dai_set_tdm_slot() call from cht_codec_init() to cht_codec_fixup(). There are 2 reasons for doing this: 1. This aligns the cht_bsw_nau8824 with all the other BYT/CHT machine drivers which also do this from their codec_fixup function. 2. When using the SOF driver, things like the TDM info is set from the topology file. Moving the call to the codec_fixup function, which gets skipped when using the SOF driver avoids the call interfering with the settings when using the SOF driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_nau8824.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c index fd5e25ca05f7..da5a5cbc8759 100644 --- a/sound/soc/intel/boards/cht_bsw_nau8824.c +++ b/sound/soc/intel/boards/cht_bsw_nau8824.c @@ -100,13 +100,6 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_component *component = codec_dai->component; int ret, jack_type; - /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0x1, 4, 24); - if (ret < 0) { - dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); - return ret; - } - /* NAU88L24 supports 4 butons headset detection * KEY_PLAYPAUSE * KEY_VOICECOMMAND @@ -141,6 +134,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + int ret; /* The DSP will covert the FE rate to 48k, stereo, 24bits */ rate->min = rate->max = 48000; @@ -150,6 +144,13 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, snd_mask_none(fmt); params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xf, 0x1, 4, 24); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + return 0; } -- 2.28.0

3 2

[PATCH] ASoC: adau17x1: Remove redundant null check before clk_disable_unprepare
by Xu Wang 12 Jan '21

12 Jan '21

Because clk_disable_unprepare() already checked NULL clock parameter, so the additional check is unnecessary, just remove it. Signed-off-by: Xu Wang <vulab(a)iscas.ac.cn> --- sound/soc/codecs/adau17x1.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 30e072c80ac1..546ee8178038 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -1095,8 +1095,7 @@ void adau17x1_remove(struct device *dev) { struct adau *adau = dev_get_drvdata(dev); - if (adau->mclk) - clk_disable_unprepare(adau->mclk); + clk_disable_unprepare(adau->mclk); } EXPORT_SYMBOL_GPL(adau17x1_remove); -- 2.17.1

3 2

[PATCH 0/3] ASoC: Intel: bytcr_rt5640: Add support for devices with only a headphone jack (no speakers/mic)
by Hans de Goede 12 Jan '21

12 Jan '21

Hi All, This series adds support for devices with only a headphone jack (no speakers/internal mic). Specifically this adds support for the Mele PCG03 Mini PC. But the new no-speakers and no-internal-mic quirks will likely be useful on other devices too. Regards, Hans

3 5

[PATCH] ASoC: es8316: Fix possible NULL pointer deref in es8316_disable_jack_detect()
by Hans de Goede 12 Jan '21

12 Jan '21

sound/soc/soc-core.c: soc_remove_component() unconditionally calls snd_soc_component_set_jack(component, NULL, NULL); on any components being removed. This means that on machines where the machine-driver does not provide a jack through snd_soc_component_set_jack() es8316_disable_jack_detect() will still get called and at this time es8316->jack will be NULL and the es8316->jack->status check in es8316_disable_jack_detect() will lead to a NULL pointer deref. Fix this by checking for es8316->jack bein NULL at the start of es8316_disable_jack_detect() and turn the function into a no-op in that case. Cc: russianneuromancer <russianneuromancer(a)ya.ru> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/es8316.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index f9ec5cf82599..1ce84f12a4cf 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -681,6 +681,9 @@ static void es8316_disable_jack_detect(struct snd_soc_component *component) { struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); + if (!es8316->jack) + return; /* Already disabled (or never enabled) */ + disable_irq(es8316->irq); mutex_lock(&es8316->lock); -- 2.28.0

2 1

[PATCH] ALSA: hda/hdmi - enable runtime pm for CI AMD display audio
by Alex Deucher 12 Jan '21

12 Jan '21

We are able to power down the GPU and audio via the GPU driver so flag these asics as supporting runtime pm. Reviewed-by: Evan Quan <evan.quan(a)amd.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/pci/hda/hda_intel.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d539f52009a1..2ea46b5a9a23 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2597,7 +2597,8 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x0002), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | + AZX_DCAPS_PM_RUNTIME }, { PCI_DEVICE(0x1002, 0x1308), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x157a), @@ -2659,9 +2660,11 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xaab0), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaac0), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | + AZX_DCAPS_PM_RUNTIME }, { PCI_DEVICE(0x1002, 0xaac8), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | + AZX_DCAPS_PM_RUNTIME }, { PCI_DEVICE(0x1002, 0xaad8), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, -- 2.29.2

2 3

[PATCH/RFC 0/2] ALSA: firewire: Fix integer overflows on 32-bit
by Geert Uytterhoeven 12 Jan '21

12 Jan '21

Hi all, This patch series fixes two integer overflows on 32-bit platforms when in multiplications with the NSEC_PER_SEC constant, found by code inspection. They are marked "RFC", as I don't know the maximum transfer length of MIDI. Depending on this maximum length, a small tweak may be necessary. Thanks for your comments! Geert Uytterhoeven (2): ALSA: fireface: Fix integer overflow in transmit_midi_msg() ALSA: firewire-tascam: Fix integer overflow in midi_port_work() sound/firewire/fireface/ff-transaction.c | 2 +- sound/firewire/tascam/tascam-transaction.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 2.25.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

4 7

[PATCH v2 0/2] fix tegra-hda on tegra30 devices
by Peter Geis 12 Jan '21

12 Jan '21

The following patches fix tegra-hda on legacy tegra devices. Two issues were discovered preventing tegra-hda from functioning: -The hda clocks on tegra30 were assigned to clk_m and running at too low of a rate to function. -The tegra-hda encounters an input/output error when opening a stream. Since the only mainline device that used tegra-hda until recently was the t124, it is unknown exactly when this was broken. Fortunately a recent patch was submitted that fixed the issue only on t194 devices. We can apply it to the tegra30-hda device to resolve the issue across the board. Note that downstream devices used the spdif device instead of hda for hdmi audio. The spdif device lacks a driver on mainline. -Checkpatch seems to have issues finding [1], but git show has no issue. [1] commit 60019d8c650d ("ALSA: hda/tegra: workaround playback failure on Tegra194") Changelog: v2: -Added ack and reviewed-by from Jon -Updated fix to apply to tegra30-hda vice universally (Thanks Jon) -Updated commit to include comments from hardware team (Thanks Sameer) -Cleaned up commit messages Peter Geis (2): clk: tegra30: Add hda clock default rates to clock driver ALSA: hda/tegra: fix tegra-hda on tegra30 soc drivers/clk/tegra/clk-tegra30.c | 2 ++ sound/pci/hda/hda_tegra.c | 2 +- 2 files changed, 3 insertions(+), 1 deletion(-) -- 2.25.1

4 9

SoF support for 8086:9d71?
by Kai-Heng Feng 12 Jan '21

12 Jan '21

Hi, On more recent kernels legacy HDA stops working for device 8086:9d71: [ 47.211381] snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 [ 47.211383] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SST driver However, only legacy HDA supports it: 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Sunrise Point-LP HD Audio [8086:9d71] (rev 21) .. Kernel modules: snd_hda_intel Ubuntu kernel doesn't select SND_SOC_INTEL_SKYLAKE, which builds snd-soc-skl, because based on the Kconfig description SOF should support it. I wonder if SND_SOC_INTEL_SKYLAKE should be select again? BugLink: <https://bugs.launchpad.net/bugs/1902457>

3 9

alsa-lib: snd_pcm_plugin_status() recently broken
by Mark Hills 12 Jan '21

12 Jan '21

The following assertion: relay: pcm_plugin.c:570: snd_pcm_plugin_status: Assertion `status->appl_ptr == *pcm->appl.ptr' failed. It bisects to 28cc099, the introduction of that assertion. #4 0x00007ffff7d0dd03 in snd_pcm_plugin_status (pcm=0x633560, status=0x7fffffffe1e0) at pcm_plugin.c:570 570 assert(status->appl_ptr == *pcm->appl.ptr); (gdb) print status->appl_ptr $1 = 0 (gdb) print pcm->appl.ptr $2 = (volatile snd_pcm_uframes_t *) 0x6334f8 (gdb) print *pcm->appl.ptr $3 = 47936 The calling code is single threaded on a dmix device, with these operations in order: snd_pcm_avail_update(); snd_pcm_status(); snd_pcm_mmap_begin(); Commenting out the assertion leaves me with a credible result from the function as it did previously; snd_pcm_status_get_delay() == 64. -- Mark >From 28cc099d9ea3962b033cb1cb9c3e07db828d9ff7 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela <perex(a)perex.cz> Date: Sun, 3 Jan 2021 16:41:29 +0100 Subject: [PATCH] pcm: plugin - optimize sync in snd_pcm_plugin_status() Do hw_ptr sync only once after the status call. This avoids double update. Also, the application pointer must not change when the status is called. Add assert() call for this condition. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- src/pcm/pcm_plugin.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index 83793397..6a815145 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -553,14 +553,11 @@ static int snd_pcm_plugin_status(snd_pcm_t *pcm, snd_pcm_status_t * status) snd_pcm_plugin_t *plugin = pcm->private_data; snd_pcm_sframes_t err; - /* sync with the latest hw and appl ptrs */ - snd_pcm_plugin_avail_update(pcm); - err = snd_pcm_status(plugin->gen.slave, status); if (err < 0) return err; - status->appl_ptr = *pcm->appl.ptr; - status->hw_ptr = *pcm->hw.ptr; + assert(status->appl_ptr == *pcm->appl.ptr); + snd_pcm_plugin_sync_hw_ptr(pcm, status->hw_ptr, status->avail); return 0; } -- 2.17.5

1 0

[PATCH][next] ASoC: soc-pcm: Fix uninitialised return value in variable ret
by Colin King 12 Jan '21

12 Jan '21

From: Colin Ian King <colin.king(a)canonical.com> Currently when attempting to start the BE fails because the FE is not started the error return variable ret is not initialized and garbage is returned. Fix this by setting it to 0 so the caller does not report the error "ASoC: failed to shutdown some BEs" and because this failure path has already reported the reason for the early return. Addresses-Coverity: ("Uninitialized scalar variable") Fixes: 2c1382840c19 ("ASoC: soc-pcm: disconnect BEs if the FE is not ready") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/soc-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 481a4a25acb0..b787ce4ceb5a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2443,6 +2443,7 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) fe->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE) { dev_err(fe->dev, "ASoC: FE %s is not ready %d\n", fe->dai_link->name, fe->dpcm[stream].state); + ret = 0; goto disconnect; } -- 2.29.2

5 8

[PATCH] docs: ALSA: Fix reference to mixart.rst
by Jonathan Neuschäfer 12 Jan '21

12 Jan '21

MIXART.txt has been converted to ReST and renamed. Fix the reference in alsa-configuration.rst. Fixes: 3d8e81862ce4 ("ALSA: doc: ReSTize MIXART.txt") Signed-off-by: Jonathan Neuschäfer <j.neuschaefer(a)gmx.net> --- Documentation/sound/alsa-configuration.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index fe52c314b7639..b36af65a08edf 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -1501,7 +1501,7 @@ Module for Digigram miXart8 sound cards. This module supports multiple cards. Note: One miXart8 board will be represented as 4 alsa cards. -See MIXART.txt for details. +See Documentation/sound/cards/mixart.rst for details. When the driver is compiled as a module and the hotplug firmware is supported, the firmware data is loaded via hotplug automatically. -- 2.29.2

3 2

[PATCH] ASoC: Intel: bytcr_rt5640: Add quirks for DeeQ X5-Z8300
by Chris Chiu 11 Jan '21

11 Jan '21

DeeQ X5-Z8300 is a laptop empowered by Intel Atom Cherry Trail. Add quirks to select the correct input map, jack-detect options to enable jack sensing and internal/headset microphones. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> --- sound/soc/intel/boards/bytcr_rt5651.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 688b5e0a49e3..9df09e26b05a 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -507,6 +507,14 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { BYT_RT5651_SSP0_AIF1 | BYT_RT5651_MONO_SPEAKER), }, + { + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "DeeQ"), + DMI_MATCH(DMI_PRODUCT_NAME, "X5-Z8300"), + }, + .driver_data = (void *)(BYT_RT5651_IN1_IN2_MAP | BYT_RT5651_JD1_1), + }, {} }; -- 2.20.1

3 3

[PATCH alsa-ucm-conf 1/2] bytcr-rt5640: Add support for devices without speakers and/or an internal mic
by Hans de Goede 11 Jan '21

11 Jan '21

There are kernel patches pending upstream for the bytcr-rt5640 machine-driver to support devices without speakers and/or an internal mic. Since the UCM profile already conditionally loads the speaker and internal-mic codec include files, these will simply get skipped in this case, so this almost works with the current UCM profile without changes. The only troublesome part is the ConflictingDevice sections in the codec HeadPhones.conf and HeadsetMic.conf files, which refer to resp. a "Speaker" and a "Mic" device. Without any of the speaker or mic codec conf files being included there will not by any "Speaker" / "Mic" devices leading to an error while parsing the HeadPhones.conf / HeadsetMic.conf files. This commit makes the ConflictingDevice section conditional, fixing this. Cc: Rasmus Porsager <rasmus(a)beat.dk> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/bytcr-rt5640/HiFi-Components.conf | 28 +++++++++++++++++++++----- ucm2/bytcr-rt5640/HiFi-LongName.conf | 28 +++++++++++++++++++++----- ucm2/codecs/rt5640/HeadPhones.conf | 14 ++++++++++--- ucm2/codecs/rt5640/HeadsetMic.conf | 14 ++++++++++--- 4 files changed, 68 insertions(+), 16 deletions(-) diff --git a/ucm2/bytcr-rt5640/HiFi-Components.conf b/ucm2/bytcr-rt5640/HiFi-Components.conf index 2ce4f59..a76b42d 100644 --- a/ucm2/bytcr-rt5640/HiFi-Components.conf +++ b/ucm2/bytcr-rt5640/HiFi-Components.conf @@ -1,10 +1,16 @@ +Define.HaveSpeaker "" +Define.HaveInternalMic "" + If.spk { Condition { Type String Haystack "${CardComponents}" Needle "cfg-spk:2" } - True.Include.spk.File "/codecs/rt5640/Speaker.conf" + True { + Include.spk.File "/codecs/rt5640/Speaker.conf" + Define.HaveSpeaker "yes" + } } If.mono { @@ -13,7 +19,10 @@ If.mono { Haystack "${CardComponents}" Needle "cfg-spk:1" } - True.Include.mspk.File "/codecs/rt5640/MonoSpeaker.conf" + True { + Include.mspk.File "/codecs/rt5640/MonoSpeaker.conf" + Define.HaveSpeaker "yes" + } } Include.hs.File "/codecs/rt5640/HeadPhones.conf" @@ -24,7 +33,10 @@ If.dmic1 { Haystack "${CardComponents}" Needle "cfg-mic:dmic1" } - True.Include.dmic.File "/codecs/rt5640/DigitalMics.conf" + True { + Include.dmic.File "/codecs/rt5640/DigitalMics.conf" + Define.HaveInternalMic "yes" + } } If.in1 { @@ -33,7 +45,10 @@ If.in1 { Haystack "${CardComponents}" Needle "cfg-mic:in1" } - True.Include.mic1.File "/codecs/rt5640/IN1-InternalMic.conf" + True { + Include.mic1.File "/codecs/rt5640/IN1-InternalMic.conf" + Define.HaveInternalMic "yes" + } } If.in3 { @@ -42,7 +57,10 @@ If.in3 { Haystack "${CardComponents}" Needle "cfg-mic:in3" } - True.Include.mic3.File "/codecs/rt5640/IN3-InternalMic.conf" + True { + Include.mic3.File "/codecs/rt5640/IN3-InternalMic.conf" + Define.HaveInternalMic "yes" + } } Include.hsmic.File "/codecs/rt5640/HeadsetMic.conf" diff --git a/ucm2/bytcr-rt5640/HiFi-LongName.conf b/ucm2/bytcr-rt5640/HiFi-LongName.conf index c07f153..6cb5556 100644 --- a/ucm2/bytcr-rt5640/HiFi-LongName.conf +++ b/ucm2/bytcr-rt5640/HiFi-LongName.conf @@ -1,10 +1,16 @@ +Define.HaveSpeaker "" +Define.HaveInternalMic "" + If.spk { Condition { Type String Haystack "${CardLongName}" Needle "-stereo-spk" } - True.Include.spk.File "/codecs/rt5640/Speaker.conf" + True { + Include.spk.File "/codecs/rt5640/Speaker.conf" + Define.HaveSpeaker "yes" + } } If.mono { @@ -13,7 +19,10 @@ If.mono { Haystack "${CardLongName}" Needle "-mono-spk" } - True.Include.mspk.File "/codecs/rt5640/MonoSpeaker.conf" + True { + Include.mspk.File "/codecs/rt5640/MonoSpeaker.conf" + Define.HaveSpeaker "yes" + } } Include.hs.File "/codecs/rt5640/HeadPhones.conf" @@ -24,7 +33,10 @@ If.dmic1 { Haystack "${CardLongName}" Needle "-dmic1-mic" } - True.Include.dmic.File "/codecs/rt5640/DigitalMics.conf" + True { + Include.dmic.File "/codecs/rt5640/DigitalMics.conf" + Define.HaveInternalMic "yes" + } } If.in1 { @@ -33,7 +45,10 @@ If.in1 { Haystack "${CardLongName}" Needle "-in1-mic" } - True.Include.mic1.File "/codecs/rt5640/IN1-InternalMic.conf" + True { + Include.mic1.File "/codecs/rt5640/IN1-InternalMic.conf" + Define.HaveInternalMic "yes" + } } If.in3 { @@ -42,7 +57,10 @@ If.in3 { Haystack "${CardLongName}" Needle "-in3-mic" } - True.Include.mic3.File "/codecs/rt5640/IN3-InternalMic.conf" + True { + Include.mic3.File "/codecs/rt5640/IN3-InternalMic.conf" + Define.HaveInternalMic "yes" + } } Include.hsmic.File "/codecs/rt5640/HeadsetMic.conf" diff --git a/ucm2/codecs/rt5640/HeadPhones.conf b/ucm2/codecs/rt5640/HeadPhones.conf index 42151d6..b7707e5 100644 --- a/ucm2/codecs/rt5640/HeadPhones.conf +++ b/ucm2/codecs/rt5640/HeadPhones.conf @@ -1,9 +1,17 @@ SectionDevice."Headphones" { Comment "Headphones" - ConflictingDevice [ - "Speaker" - ] + If.have-spk { + Condition { + Type String + Empty "${var:HaveSpeaker}" + } + False { + ConflictingDevice [ + "Speaker" + ] + } + } EnableSequence [ cset "name='DAC MIXL INF1 Switch' on" diff --git a/ucm2/codecs/rt5640/HeadsetMic.conf b/ucm2/codecs/rt5640/HeadsetMic.conf index 7a8dfb8..f88b611 100644 --- a/ucm2/codecs/rt5640/HeadsetMic.conf +++ b/ucm2/codecs/rt5640/HeadsetMic.conf @@ -1,9 +1,17 @@ SectionDevice."Headset" { Comment "Headset Microphone" - ConflictingDevice [ - "Mic" - ] + If.have-mic { + Condition { + Type String + Empty "${var:HaveInternalMic}" + } + False { + ConflictingDevice [ + "Mic" + ] + } + } EnableSequence [ cset "name='Headset Mic Switch' on" -- 2.28.0

2 3

[PATCH] ASoC: soc-pcm: return correct -ERRNO in failure path
by Souptick Joarder 11 Jan '21

11 Jan '21

Kernel test robot throws below error -> sound/soc/soc-pcm.c:2523 dpcm_run_update_startup() error: uninitialized symbol 'ret'. Initializing ret = 0 and returning correct -ERRNO in failure path. Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Souptick Joarder <jrdr.linux(a)gmail.com> --- sound/soc/soc-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 481a4a2..29328ce 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2432,7 +2432,7 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) snd_soc_dpcm_get_substream(fe, stream); struct snd_soc_dpcm *dpcm; enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; - int ret; + int ret = 0; unsigned long flags; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", @@ -2441,6 +2441,7 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) /* Only start the BE if the FE is ready */ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_FREE || fe->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE) { + ret = -EINVAL; dev_err(fe->dev, "ASoC: FE %s is not ready %d\n", fe->dai_link->name, fe->dpcm[stream].state); goto disconnect; -- 1.9.1

2 1

[PATCH v3 0/4] ASoC: rt5645: Enable internal mic and headset on ECS EF20
by Chris Chiu 11 Jan '21

11 Jan '21

These patches are trying to fix the jack detection and internal microphone problems on ECS EF20 series laptops which are empowered by Intel Atom x5-Z8350 CPU (CherryTrail) with Realtek rt5645 audio codec. --- v2 -> v3: Restore the accidentally removed terminator of the dmi_platform_data[]. v1 -> v2: Invoke callback() of the DMI quirk if it exists, because the dmi_first_match() doesn't. --- 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 include/sound/rt5645.h | 2 ++ sound/soc/codecs/rt5645.c | 45 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 47 insertions(+) -- 2.20.1

2 5

[PATCH 1/2] ASoC: rt5682: enable fast discharge for headset unplugging
by shumingf＠realtek.com 11 Jan '21

11 Jan '21

From: Shuming Fan <shumingf(a)realtek.com> To avoid the pop noise, this patch enables the feature of fast discharge. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 4d865edadd7e..42ad5bea86b5 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -953,6 +953,8 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) case 0x1: case 0x2: rt5682->jack_type = SND_JACK_HEADSET; + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_FAST_OFF_MASK, RT5682_FAST_OFF_EN); rt5682_enable_push_button_irq(component, true); break; default: @@ -982,6 +984,8 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) snd_soc_component_update_bits(component, RT5682_MICBIAS_2, RT5682_PWR_CLK25M_MASK | RT5682_PWR_CLK1M_MASK, RT5682_PWR_CLK25M_PD | RT5682_PWR_CLK1M_PD); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_FAST_OFF_MASK, RT5682_FAST_OFF_DIS); rt5682->jack_type = 0; } @@ -1011,11 +1015,13 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, if (!rt5682->is_sdw) { switch (rt5682->pdata.jd_src) { case RT5682_JD1: + snd_soc_component_update_bits(component, + RT5682_CBJ_CTRL_5, 0x0700, 0x0600); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2, RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); snd_soc_component_write(component, RT5682_CBJ_CTRL_1, - 0xd042); + 0xd142); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_3, RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); -- 2.29.0

2 1

[PATCH 2/2] ASoC: rt5682: fix getting the wrong device id when the suspend_stress_test
by shumingf＠realtek.com 11 Jan '21

11 Jan '21

From: Shuming Fan <shumingf(a)realtek.com> This patch will be the workaround to fix getting the wrong device ID on the rare chance. It seems like something unstable when the system resumes. e.g. the bus clock This patch tries to read the device ID to check several times. After the test, the driver will get the correct device ID the second time. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682-sdw.c | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 4d707e854875..69f046bba67a 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -375,18 +375,12 @@ static int rt5682_sdw_init(struct device *dev, struct regmap *regmap, static int rt5682_io_init(struct device *dev, struct sdw_slave *slave) { struct rt5682_priv *rt5682 = dev_get_drvdata(dev); - int ret = 0; + int ret = 0, loop = 10; unsigned int val; if (rt5682->hw_init) return 0; - regmap_read(rt5682->regmap, RT5682_DEVICE_ID, &val); - if (val != DEVICE_ID) { - dev_err(dev, "Device with ID register %x is not rt5682\n", val); - return -ENODEV; - } - /* * PM runtime is only enabled when a Slave reports as Attached */ @@ -406,6 +400,19 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave) pm_runtime_get_noresume(&slave->dev); + while (loop > 0) { + regmap_read(rt5682->regmap, RT5682_DEVICE_ID, &val); + if (val == DEVICE_ID) + break; + dev_warn(dev, "Device with ID register %x is not rt5682\n", val); + usleep_range(30000, 30005); + loop--; + } + if (val != DEVICE_ID) { + dev_err(dev, "Device with ID register %x is not rt5682\n", val); + return -ENODEV; + } + if (rt5682->first_hw_init) { regcache_cache_only(rt5682->regmap, false); regcache_cache_bypass(rt5682->regmap, true); -- 2.29.0

2 1

[PATCH] ASoC: soc-pcm: Fix an uninitialized error code
by Dan Carpenter 11 Jan '21

11 Jan '21

The error path here doesn't set "ret" so it returns uninitialized data instead of a negative error code. Fixes: 2c1382840c19 ("ASoC: soc-pcm: disconnect BEs if the FE is not ready") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/soc-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 481a4a25acb0..489697963dd3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2443,6 +2443,7 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) fe->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE) { dev_err(fe->dev, "ASoC: FE %s is not ready %d\n", fe->dai_link->name, fe->dpcm[stream].state); + ret = -EINVAL; goto disconnect; } -- 2.29.2

2 1

[PATCH] ASoC: wm_adsp: Fix uninitialized variable warnings
by Richard Fitzgerald 11 Jan '21

11 Jan '21

wm_adsp_read_data_word() used if (ret) to check for an error from wm_adsp_read_raw_data_block(). While this is perfectly valid, wm_adsp_read_raw_data_block() itself uses if (ret < 0) and three calls to wm_adsp_read_data_word() also use if (ret < 0). This creates an error check chain like this: 1st) if (ret < 0) return ret; 2nd) if (ret) return ret; 3rd) if (ret < 0) ... This can confuse the compiler into thinking that there are possible returns > 0 from the middle if() that are not handled by the final if(). If this was true it would lead to using uninitialized variables later in the outer function. Fix this by changing the test in wm_adsp_read_data_word() to be if (ret < 0). Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f8ad768364c2..1fc7bc1970ea 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3694,7 +3694,7 @@ static inline int wm_adsp_read_data_word(struct wm_adsp *dsp, int mem_type, int ret; ret = wm_adsp_read_raw_data_block(dsp, mem_type, mem_addr, 1, &raw); - if (ret) + if (ret < 0) return ret; *data = be32_to_cpu(raw) & 0x00ffffffu; -- 2.20.1

2 1

[PATCH] ASoC: audio-graph-card: Drop remote-endpoint as required property
by Sameer Pujar 11 Jan '21

11 Jan '21

The remote-endpoint may not be available if it is part of some pluggable module. One such example would be an audio card, the Codec endpoint will not be available until it is plugged in. Hence drop 'remote-endpoint' as a required property. Cc: Rob Herring <robh(a)kernel.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- Documentation/devicetree/bindings/sound/audio-graph-port.yaml | 3 --- 1 file changed, 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index 2005014..766e910 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -71,9 +71,6 @@ properties: description: CPU to Codec rate channels. $ref: /schemas/types.yaml#/definitions/uint32 - required: - - remote-endpoint - ports: description: multi OF-Graph subnode type: object -- 2.7.4

2 4

[PATCH v2] ASoC: hdmi-codec: Fix return value in hdmi_codec_set_jack()
by Stephan Gerhold 08 Jan '21

08 Jan '21

Sound is broken on the DragonBoard 410c (apq8016_sbc) since 5.10: hdmi-audio-codec hdmi-audio-codec.1.auto: ASoC: error at snd_soc_component_set_jack on hdmi-audio-codec.1.auto: -95 qcom-apq8016-sbc 7702000.sound: Failed to set jack: -95 ADV7533: ASoC: error at snd_soc_link_init on ADV7533: -95 hdmi-audio-codec hdmi-audio-codec.1.auto: ASoC: error at snd_soc_component_set_jack on hdmi-audio-codec.1.auto: -95 qcom-apq8016-sbc: probe of 7702000.sound failed with error -95 This happens because apq8016_sbc calls snd_soc_component_set_jack() on all codec DAIs and attempts to ignore failures with return code -ENOTSUPP. -ENOTSUPP is also excluded from error logging in soc_component_ret(). However, hdmi_codec_set_jack() returns -E*OP*NOTSUPP if jack detection is not supported, which is not handled in apq8016_sbc and soc_component_ret(). Make it return -ENOTSUPP instead to fix sound and silence the errors. Cc: Cheng-Yi Chiang <cychiang(a)chromium.org> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Fixes: 55c5cc63ab32 ("ASoC: hdmi-codec: Use set_jack ops to set jack") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- Changes in v2: - Replace -EOPNOTSUPP check with -ENOTSUPP in imx-hdmi.c --- sound/soc/codecs/hdmi-codec.c | 2 +- sound/soc/fsl/imx-hdmi.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index d5fcc4db8284..0f3ac22f2cf8 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -717,7 +717,7 @@ static int hdmi_codec_set_jack(struct snd_soc_component *component, void *data) { struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); - int ret = -EOPNOTSUPP; + int ret = -ENOTSUPP; if (hcp->hcd.ops->hook_plugged_cb) { hcp->jack = jack; diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index ede4a9ad1054..dbbb7618351c 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -90,7 +90,7 @@ static int imx_hdmi_init(struct snd_soc_pcm_runtime *rtd) } ret = snd_soc_component_set_jack(component, &data->hdmi_jack, NULL); - if (ret && ret != -EOPNOTSUPP) { + if (ret && ret != -ENOTSUPP) { dev_err(card->dev, "Can't set HDMI Jack %d\n", ret); return ret; } -- 2.30.0

3 2

[PATCH] ASoC: es8328: Remove redundant null check before clk_disable_unprepare
by Xu Wang 08 Jan '21

08 Jan '21

Because clk_disable_unprepare() already checked NULL clock parameter, so the additional check is unnecessary, just remove it. Signed-off-by: Xu Wang <vulab(a)iscas.ac.cn> --- sound/soc/codecs/es8328.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 7e26231a596a..b537300d0ce8 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -809,8 +809,7 @@ static void es8328_remove(struct snd_soc_component *component) es8328 = snd_soc_component_get_drvdata(component); - if (es8328->clk) - clk_disable_unprepare(es8328->clk); + clk_disable_unprepare(es8328->clk); regulator_bulk_disable(ARRAY_SIZE(es8328->supplies), es8328->supplies); -- 2.17.1

2 1

[PATCH 0/2] fix tegra-hda on tegra30 devices
by Peter Geis 08 Jan '21

08 Jan '21

The following patches fix tegra-hda on legacy tegra devices. Two issues were discovered preventing tegra-hda from functioning: The hda clocks on tegra30 were assigned to clk_m and running at too low of a rate to function. The tegra-hda encounters an input/output error when opening a stream. Since the only mainline device that used tegra-hda previously was the t124, it is unknown exactly when this was broken. Fortunately a recent patch was submitted that fixed the issue only on t194 devices. We can apply it universally to the tegra-hda device to resolve the issues across the board. Note that downstream devices used the spdif device instead of hda for hdmi audio. The spdif device lacks a driver on mainline. Peter Geis (2): clk: tegra30: Add hda clock default rates to clock driver ALSA: hda/tegra: fix tegra-hda on tegra30 soc drivers/clk/tegra/clk-tegra30.c | 2 ++ sound/pci/hda/hda_tegra.c | 3 +-- 2 files changed, 3 insertions(+), 2 deletions(-) -- 2.25.1

4 10

[PATCH] ASoC: codecs: da*: Remove redundant null check before clk_prepare_enable/clk_disable_unprepare
by Xu Wang 08 Jan '21

08 Jan '21

Because clk_prepare_enable() and clk_disable_unprepare() already checked NULL clock parameter, so the additional checks are unnecessary, just remove them. Signed-off-by: Xu Wang <vulab(a)iscas.ac.cn> --- sound/soc/codecs/da7218.c | 13 +++++-------- sound/soc/codecs/da7219-aad.c | 19 ++++++++----------- sound/soc/codecs/da7219.c | 3 +-- 3 files changed, 14 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index 2bfafbe9e3dc..324d95e3b48a 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2582,12 +2582,10 @@ static int da7218_set_bias_level(struct snd_soc_component *component, case SND_SOC_BIAS_PREPARE: /* Enable MCLK for transition to ON state */ if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_STANDBY) { - if (da7218->mclk) { - ret = clk_prepare_enable(da7218->mclk); - if (ret) { - dev_err(component->dev, "Failed to enable mclk\n"); - return ret; - } + ret = clk_prepare_enable(da7218->mclk); + if (ret) { + dev_err(component->dev, "Failed to enable mclk\n"); + return ret; } } @@ -2605,8 +2603,7 @@ static int da7218_set_bias_level(struct snd_soc_component *component, DA7218_LDO_EN_MASK); } else { /* Remove MCLK */ - if (da7218->mclk) - clk_disable_unprepare(da7218->mclk); + clk_disable_unprepare(da7218->mclk); } break; case SND_SOC_BIAS_OFF: diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 48081d71c22c..fa61cfc8e50c 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -123,15 +123,13 @@ static void da7219_aad_hptest_work(struct work_struct *work) mutex_lock(&da7219->pll_lock); /* Ensure MCLK is available for HP test procedure */ - if (da7219->mclk) { - ret = clk_prepare_enable(da7219->mclk); - if (ret) { - dev_err(component->dev, "Failed to enable mclk - %d\n", ret); - mutex_unlock(&da7219->pll_lock); - mutex_unlock(&da7219->ctrl_lock); - snd_soc_dapm_mutex_unlock(dapm); - return; - } + ret = clk_prepare_enable(da7219->mclk); + if (ret) { + dev_err(component->dev, "Failed to enable mclk - %d\n", ret); + mutex_unlock(&da7219->pll_lock); + mutex_unlock(&da7219->ctrl_lock); + snd_soc_dapm_mutex_unlock(dapm); + return; } /* @@ -318,8 +316,7 @@ static void da7219_aad_hptest_work(struct work_struct *work) da7219_set_pll(component, DA7219_SYSCLK_MCLK, 0); /* Remove MCLK, if previously enabled */ - if (da7219->mclk) - clk_disable_unprepare(da7219->mclk); + clk_disable_unprepare(da7219->mclk); mutex_unlock(&da7219->pll_lock); mutex_unlock(&da7219->ctrl_lock); diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index e9b45daec0ca..50307cc9d640 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1826,8 +1826,7 @@ static int da7219_set_bias_level(struct snd_soc_component *component, if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_PREPARE) { /* Remove MCLK */ - if (da7219->mclk) - clk_disable_unprepare(da7219->mclk); + clk_disable_unprepare(da7219->mclk); } break; case SND_SOC_BIAS_OFF: -- 2.17.1

2 1

[PATCH 0/4] ASoC: rt5645: Enable internal mic and headset on ECS EF20
by Chris Chiu 08 Jan '21

08 Jan '21

These patches are trying to fix the jack detection and internal microphone problems on ECS EF20 series laptops which are empowered by Intel Atom x5-Z8350 CPU (CherryTrail) with Realtek rt5645 audio codec. 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 include/sound/rt5645.h | 2 ++ sound/soc/codecs/rt5645.c | 46 ++++++++++++++++++++++++++++++++++++++- 2 files changed, 47 insertions(+), 1 deletion(-) -- 2.20.1

2 6

Add support for WAVE file format in cplay
by GitHub pull_request - opened 07 Jan '21

07 Jan '21

alsa-project/tinycompress pull request #9 was opened from dbaluta: This will allow us to use PCM codec type with cplay. This patchseries moves some WAVE related functions and structures to their own file in order to be used by both cplay/crecord. Request URL : https://github.com/alsa-project/tinycompress/pull/9 Patch URL : https://github.com/alsa-project/tinycompress/pull/9.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

ASoC: soc-pcm: disconnect BEs if the FE is not ready
by 朱灿灿 07 Jan '21

07 Jan '21

FE is connected to two BEs, BE1 is active, BE2 is deactive. When closing BE1, FE/BE1 is in HW_FREE state, then BE2 is startup by mixer runtime update. For FE is in HW_FREE state, dpcm_run_update_startup() will skip BE2's startup because FE's state is HW_FREE, BE2 stays in FE's be_clients list. During FE's closed, the dpcm_fe_dai_close() will close all related BEs, BE2 will be closed. This will lead to BE2's dpcm[stream].users mismatch. We need disconnet all pending BEs in the corner case. Signed-off-by: zhucancan <zhucancan(a)vivo.com> --- sound/soc/soc-pcm.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..04ad53f0ae4f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2451,8 +2451,11 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) /* Only start the BE if the FE is ready */ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_FREE || - fe->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE) - return -EINVAL; + fe->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE) { + dev_err(fe->dev, "ASoC: FE %s is not ready %d\n", + fe->dai_link->name, fe->dpcm[stream].state); + goto disconnect; + } /* startup must always be called for new BEs */ ret = dpcm_be_dai_startup(fe, stream); @@ -2513,12 +2516,18 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) close: dpcm_be_dai_shutdown(fe, stream); disconnect: - /* disconnect any closed BEs */ + /* disconnect any pending BEs */ spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; - if (be->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE) - dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + if (be->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE || + be->dpcm[stream].state == SND_SOC_DPCM_STATE_NEW) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; } spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); -- 2.21.0

2 1

[PATCH] ASoC: Intel: bytcht_es8316: Remove comment about SSP0 being untested
by Hans de Goede 07 Jan '21

07 Jan '21

Remove a stale comment about SSP0 being untested, the bytcht_es8316 has supported SSP0 for a while now and this has been successfully tested on a GP electrinic T701 tablet. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcht_es8316.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 892cf684216e..06df2d46d910 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -331,9 +331,6 @@ static struct snd_soc_dai_link byt_cht_es8316_dais[] = { /* back ends */ { - /* Only SSP2 has been tested here, so BYT-CR platforms that - * require SSP0 will not work. - */ .name = "SSP2-Codec", .id = 0, .no_pcm = 1, -- 2.28.0

2 1

[PATCH resend] sound: Convert strlcpy to strscpy when return value is unused
by Joe Perches 07 Jan '21

07 Jan '21

strlcpy is deprecated. see: Documentation/process/deprecated.rst Change the calls that do not use the strlcpy return value to the preferred strscpy. Done with cocci script: @@ expression e1, e2, e3; @@ - strlcpy( + strscpy( e1, e2, e3); This cocci script leaves the instances where the return value is used unchanged. After this patch, sound/ has 3 uses of strlcpy() that need to be manually inspected for conversion and changed one day. $ git grep -w strlcpy sound/ sound/usb/card.c: len = strlcpy(card->longname, s, sizeof(card->longname)); sound/usb/mixer.c: return strlcpy(buf, p->name, buflen); sound/usb/mixer.c: return strlcpy(buf, p->names[index], buflen); Miscellenea: o Remove trailing whitespace in conversion of sound/core/hwdep.c Link: https://lore.kernel.org/lkml/CAHk-=wgfRnXz0W3D37d01q3JFkr_i_uTL=V6A6G1oUZcp… Signed-off-by: Joe Perches <joe(a)perches.com> --- Resend adding alsa-devel as requested by Takashi Iwai <tiwai(a)suse.de> sound/aoa/codecs/onyx.c | 2 +- sound/aoa/codecs/tas.c | 2 +- sound/aoa/codecs/toonie.c | 2 +- sound/aoa/core/alsa.c | 8 ++++---- sound/aoa/fabrics/layout.c | 6 +++--- sound/aoa/soundbus/sysfs.c | 2 +- sound/arm/aaci.c | 6 +++--- sound/arm/pxa2xx-ac97.c | 2 +- sound/core/compress_offload.c | 2 +- sound/core/control.c | 16 ++++++++-------- sound/core/ctljack.c | 2 +- sound/core/hwdep.c | 6 +++--- sound/core/init.c | 4 ++-- sound/core/oss/mixer_oss.c | 12 ++++++------ sound/core/pcm.c | 2 +- sound/core/pcm_native.c | 6 +++--- sound/core/rawmidi.c | 2 +- sound/core/seq/oss/seq_oss_midi.c | 4 ++-- sound/core/seq/oss/seq_oss_synth.c | 6 +++--- sound/core/seq/seq_clientmgr.c | 2 +- sound/core/seq/seq_ports.c | 6 +++--- sound/core/timer.c | 10 +++++----- sound/core/timer_compat.c | 4 ++-- sound/drivers/opl3/opl3_oss.c | 2 +- sound/drivers/opl3/opl3_synth.c | 2 +- sound/firewire/bebob/bebob_hwdep.c | 2 +- sound/firewire/dice/dice-hwdep.c | 2 +- sound/firewire/digi00x/digi00x-hwdep.c | 2 +- sound/firewire/fireface/ff-hwdep.c | 2 +- sound/firewire/fireworks/fireworks_hwdep.c | 2 +- sound/firewire/motu/motu-hwdep.c | 2 +- sound/firewire/oxfw/oxfw-hwdep.c | 2 +- sound/firewire/tascam/tascam-hwdep.c | 2 +- sound/i2c/i2c.c | 4 ++-- sound/isa/ad1848/ad1848.c | 4 ++-- sound/isa/cs423x/cs4231.c | 4 ++-- sound/isa/cs423x/cs4236.c | 4 ++-- sound/isa/es1688/es1688.c | 4 ++-- sound/isa/sb/sb16_csp.c | 2 +- sound/isa/sb/sb_mixer.c | 2 +- sound/oss/dmasound/dmasound_core.c | 4 ++-- sound/pci/cs5535audio/cs5535audio_olpc.c | 4 ++-- sound/pci/ctxfi/ctpcm.c | 2 +- sound/pci/emu10k1/emu10k1.c | 4 ++-- sound/pci/emu10k1/emu10k1_main.c | 2 +- sound/pci/emu10k1/emufx.c | 6 +++--- sound/pci/es1968.c | 2 +- sound/pci/fm801.c | 2 +- sound/pci/hda/hda_auto_parser.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_controller.c | 2 +- sound/pci/hda/hda_eld.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/hda/hda_jack.c | 2 +- sound/pci/ice1712/juli.c | 2 +- sound/pci/ice1712/psc724.c | 4 ++-- sound/pci/ice1712/quartet.c | 2 +- sound/pci/ice1712/wm8776.c | 2 +- sound/pci/lola/lola.c | 2 +- sound/pci/lola/lola_pcm.c | 2 +- sound/pci/rme9652/hdspm.c | 2 +- sound/ppc/keywest.c | 2 +- sound/soc/qcom/qdsp6/q6afe.c | 2 +- sound/soc/sh/rcar/core.c | 2 +- sound/usb/bcd2000/bcd2000.c | 2 +- sound/usb/caiaq/audio.c | 2 +- sound/usb/caiaq/device.c | 6 +++--- sound/usb/caiaq/midi.c | 2 +- sound/usb/card.c | 4 ++-- sound/usb/hiface/chip.c | 6 +++--- sound/usb/hiface/pcm.c | 2 +- sound/usb/mixer.c | 12 ++++++------ sound/usb/mixer_quirks.c | 2 +- sound/usb/mixer_scarlett.c | 2 +- sound/usb/mixer_scarlett_gen2.c | 2 +- sound/usb/mixer_us16x08.c | 2 +- sound/x86/intel_hdmi_audio.c | 2 +- sound/xen/xen_snd_front_cfg.c | 2 +- 79 files changed, 134 insertions(+), 134 deletions(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 12028b3e2eee..1abee841cc45 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -1013,7 +1013,7 @@ static int onyx_i2c_probe(struct i2c_client *client, goto fail; } - strlcpy(onyx->codec.name, "onyx", MAX_CODEC_NAME_LEN); + strscpy(onyx->codec.name, "onyx", MAX_CODEC_NAME_LEN); onyx->codec.owner = THIS_MODULE; onyx->codec.init = onyx_init_codec; onyx->codec.exit = onyx_exit_codec; diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index d3e37577b529..ac246dd3ab49 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -894,7 +894,7 @@ static int tas_i2c_probe(struct i2c_client *client, /* seems that half is a saner default */ tas->drc_range = TAS3004_DRC_MAX / 2; - strlcpy(tas->codec.name, "tas", MAX_CODEC_NAME_LEN); + strscpy(tas->codec.name, "tas", MAX_CODEC_NAME_LEN); tas->codec.owner = THIS_MODULE; tas->codec.init = tas_init_codec; tas->codec.exit = tas_exit_codec; diff --git a/sound/aoa/codecs/toonie.c b/sound/aoa/codecs/toonie.c index c2d014486c33..0da5af129492 100644 --- a/sound/aoa/codecs/toonie.c +++ b/sound/aoa/codecs/toonie.c @@ -126,7 +126,7 @@ static int __init toonie_init(void) if (!toonie) return -ENOMEM; - strlcpy(toonie->codec.name, "toonie", sizeof(toonie->codec.name)); + strscpy(toonie->codec.name, "toonie", sizeof(toonie->codec.name)); toonie->codec.owner = THIS_MODULE; toonie->codec.init = toonie_init_codec; toonie->codec.exit = toonie_exit_codec; diff --git a/sound/aoa/core/alsa.c b/sound/aoa/core/alsa.c index b61081342266..7fce8581ddbd 100644 --- a/sound/aoa/core/alsa.c +++ b/sound/aoa/core/alsa.c @@ -28,10 +28,10 @@ int aoa_alsa_init(char *name, struct module *mod, struct device *dev) return err; aoa_card = alsa_card->private_data; aoa_card->alsa_card = alsa_card; - strlcpy(alsa_card->driver, "AppleOnbdAudio", sizeof(alsa_card->driver)); - strlcpy(alsa_card->shortname, name, sizeof(alsa_card->shortname)); - strlcpy(alsa_card->longname, name, sizeof(alsa_card->longname)); - strlcpy(alsa_card->mixername, name, sizeof(alsa_card->mixername)); + strscpy(alsa_card->driver, "AppleOnbdAudio", sizeof(alsa_card->driver)); + strscpy(alsa_card->shortname, name, sizeof(alsa_card->shortname)); + strscpy(alsa_card->longname, name, sizeof(alsa_card->longname)); + strscpy(alsa_card->mixername, name, sizeof(alsa_card->mixername)); err = snd_card_register(aoa_card->alsa_card); if (err < 0) { printk(KERN_ERR "snd-aoa: couldn't register alsa card\n"); diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index d2e85b83f7ed..ec4ef18555bc 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -948,7 +948,7 @@ static void layout_attached_codec(struct aoa_codec *codec) ldev->gpio.methods->set_lineout(codec->gpio, 1); ctl = snd_ctl_new1(&lineout_ctl, codec->gpio); if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE) - strlcpy(ctl->id.name, + strscpy(ctl->id.name, "Headphone Switch", sizeof(ctl->id.name)); ldev->lineout_ctrl = ctl; aoa_snd_ctl_add(ctl); @@ -962,14 +962,14 @@ static void layout_attached_codec(struct aoa_codec *codec) ctl = snd_ctl_new1(&lineout_detect_choice, ldev); if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE) - strlcpy(ctl->id.name, + strscpy(ctl->id.name, "Headphone Detect Autoswitch", sizeof(ctl->id.name)); aoa_snd_ctl_add(ctl); ctl = snd_ctl_new1(&lineout_detected, ldev); if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE) - strlcpy(ctl->id.name, + strscpy(ctl->id.name, "Headphone Detected", sizeof(ctl->id.name)); ldev->lineout_detected_ctrl = ctl; diff --git a/sound/aoa/soundbus/sysfs.c b/sound/aoa/soundbus/sysfs.c index a2d55e15afbb..dead3105689b 100644 --- a/sound/aoa/soundbus/sysfs.c +++ b/sound/aoa/soundbus/sysfs.c @@ -13,7 +13,7 @@ static ssize_t modalias_show(struct device *dev, struct device_attribute *attr, int length; if (*sdev->modalias) { - strlcpy(buf, sdev->modalias, sizeof(sdev->modalias) + 1); + strscpy(buf, sdev->modalias, sizeof(sdev->modalias) + 1); strcat(buf, "\n"); length = strlen(buf); } else { diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index a0996c47e58f..f02a91bdaa97 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -890,8 +890,8 @@ static struct aaci *aaci_init_card(struct amba_device *dev) card->private_free = aaci_free_card; - strlcpy(card->driver, DRIVER_NAME, sizeof(card->driver)); - strlcpy(card->shortname, "ARM AC'97 Interface", sizeof(card->shortname)); + strscpy(card->driver, DRIVER_NAME, sizeof(card->driver)); + strscpy(card->shortname, "ARM AC'97 Interface", sizeof(card->shortname)); snprintf(card->longname, sizeof(card->longname), "%s PL%03x rev%u at 0x%08llx, irq %d", card->shortname, amba_part(dev), amba_rev(dev), @@ -921,7 +921,7 @@ static int aaci_init_pcm(struct aaci *aaci) pcm->private_data = aaci; pcm->info_flags = 0; - strlcpy(pcm->name, DRIVER_NAME, sizeof(pcm->name)); + strscpy(pcm->name, DRIVER_NAME, sizeof(pcm->name)); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &aaci_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &aaci_capture_ops); diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index ea8e233150c8..6322e6392594 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -235,7 +235,7 @@ static int pxa2xx_ac97_probe(struct platform_device *dev) if (ret < 0) goto err; - strlcpy(card->driver, dev->dev.driver->name, sizeof(card->driver)); + strscpy(card->driver, dev->dev.driver->name, sizeof(card->driver)); ret = pxa2xx_ac97_pcm_new(card); if (ret) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index debc30fcf5b3..21ce4c056a92 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -1132,7 +1132,7 @@ static void snd_compress_proc_done(struct snd_compr *compr) static inline void snd_compress_set_id(struct snd_compr *compr, const char *id) { - strlcpy(compr->id, id, sizeof(compr->id)); + strscpy(compr->id, id, sizeof(compr->id)); } #else static inline int snd_compress_proc_init(struct snd_compr *compr) diff --git a/sound/core/control.c b/sound/core/control.c index 3b44378b9dec..8f41aba6acac 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -261,7 +261,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, kctl->id.device = ncontrol->device; kctl->id.subdevice = ncontrol->subdevice; if (ncontrol->name) { - strlcpy(kctl->id.name, ncontrol->name, sizeof(kctl->id.name)); + strscpy(kctl->id.name, ncontrol->name, sizeof(kctl->id.name)); if (strcmp(ncontrol->name, kctl->id.name) != 0) pr_warn("ALSA: Control name '%s' truncated to '%s'\n", ncontrol->name, kctl->id.name); @@ -701,12 +701,12 @@ static int snd_ctl_card_info(struct snd_card *card, struct snd_ctl_file * ctl, return -ENOMEM; down_read(&snd_ioctl_rwsem); info->card = card->number; - strlcpy(info->id, card->id, sizeof(info->id)); - strlcpy(info->driver, card->driver, sizeof(info->driver)); - strlcpy(info->name, card->shortname, sizeof(info->name)); - strlcpy(info->longname, card->longname, sizeof(info->longname)); - strlcpy(info->mixername, card->mixername, sizeof(info->mixername)); - strlcpy(info->components, card->components, sizeof(info->components)); + strscpy(info->id, card->id, sizeof(info->id)); + strscpy(info->driver, card->driver, sizeof(info->driver)); + strscpy(info->name, card->shortname, sizeof(info->name)); + strscpy(info->longname, card->longname, sizeof(info->longname)); + strscpy(info->mixername, card->mixername, sizeof(info->mixername)); + strscpy(info->components, card->components, sizeof(info->components)); up_read(&snd_ioctl_rwsem); if (copy_to_user(arg, info, sizeof(struct snd_ctl_card_info))) { kfree(info); @@ -2137,7 +2137,7 @@ int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels, WARN(strlen(names[info->value.enumerated.item]) >= sizeof(info->value.enumerated.name), "ALSA: too long item name '%s'\n", names[info->value.enumerated.item]); - strlcpy(info->value.enumerated.name, + strscpy(info->value.enumerated.name, names[info->value.enumerated.item], sizeof(info->value.enumerated.name)); return 0; diff --git a/sound/core/ctljack.c b/sound/core/ctljack.c index 9be4e282f2e0..709b1a9c2caa 100644 --- a/sound/core/ctljack.c +++ b/sound/core/ctljack.c @@ -35,7 +35,7 @@ static int get_available_index(struct snd_card *card, const char *name) sid.index = 0; sid.iface = SNDRV_CTL_ELEM_IFACE_CARD; - strlcpy(sid.name, name, sizeof(sid.name)); + strscpy(sid.name, name, sizeof(sid.name)); while (snd_ctl_find_id(card, &sid)) { sid.index++; diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 0c029892880a..e01602bcd674 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -177,8 +177,8 @@ static int snd_hwdep_info(struct snd_hwdep *hw, memset(&info, 0, sizeof(info)); info.card = hw->card->number; - strlcpy(info.id, hw->id, sizeof(info.id)); - strlcpy(info.name, hw->name, sizeof(info.name)); + strscpy(info.id, hw->id, sizeof(info.id)); + strscpy(info.name, hw->name, sizeof(info.name)); info.iface = hw->iface; if (copy_to_user(_info, &info, sizeof(info))) return -EFAULT; @@ -379,7 +379,7 @@ int snd_hwdep_new(struct snd_card *card, char *id, int device, hwdep->card = card; hwdep->device = device; if (id) - strlcpy(hwdep->id, id, sizeof(hwdep->id)); + strscpy(hwdep->id, id, sizeof(hwdep->id)); snd_device_initialize(&hwdep->dev, card); hwdep->dev.release = release_hwdep_device; diff --git a/sound/core/init.c b/sound/core/init.c index 75aec71c48a8..56834febc7a4 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -174,7 +174,7 @@ int snd_card_new(struct device *parent, int idx, const char *xid, if (extra_size > 0) card->private_data = (char *)card + sizeof(struct snd_card); if (xid) - strlcpy(card->id, xid, sizeof(card->id)); + strscpy(card->id, xid, sizeof(card->id)); err = 0; mutex_lock(&snd_card_mutex); if (idx < 0) /* first check the matching module-name slot */ @@ -623,7 +623,7 @@ static void snd_card_set_id_no_lock(struct snd_card *card, const char *src, /* last resort... */ dev_err(card->dev, "unable to set card id (%s)\n", id); if (card->proc_root->name) - strlcpy(card->id, card->proc_root->name, sizeof(card->id)); + strscpy(card->id, card->proc_root->name, sizeof(card->id)); } /** diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index f702c96a7478..2f53750a9875 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -87,8 +87,8 @@ static int snd_mixer_oss_info(struct snd_mixer_oss_file *fmixer, struct mixer_info info; memset(&info, 0, sizeof(info)); - strlcpy(info.id, mixer && mixer->id[0] ? mixer->id : card->driver, sizeof(info.id)); - strlcpy(info.name, mixer && mixer->name[0] ? mixer->name : card->mixername, sizeof(info.name)); + strscpy(info.id, mixer && mixer->id[0] ? mixer->id : card->driver, sizeof(info.id)); + strscpy(info.name, mixer && mixer->name[0] ? mixer->name : card->mixername, sizeof(info.name)); info.modify_counter = card->mixer_oss_change_count; if (copy_to_user(_info, &info, sizeof(info))) return -EFAULT; @@ -103,8 +103,8 @@ static int snd_mixer_oss_info_obsolete(struct snd_mixer_oss_file *fmixer, _old_mixer_info info; memset(&info, 0, sizeof(info)); - strlcpy(info.id, mixer && mixer->id[0] ? mixer->id : card->driver, sizeof(info.id)); - strlcpy(info.name, mixer && mixer->name[0] ? mixer->name : card->mixername, sizeof(info.name)); + strscpy(info.id, mixer && mixer->id[0] ? mixer->id : card->driver, sizeof(info.id)); + strscpy(info.name, mixer && mixer->name[0] ? mixer->name : card->mixername, sizeof(info.name)); if (copy_to_user(_info, &info, sizeof(info))) return -EFAULT; return 0; @@ -499,7 +499,7 @@ static struct snd_kcontrol *snd_mixer_oss_test_id(struct snd_mixer_oss *mixer, c memset(&id, 0, sizeof(id)); id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strlcpy(id.name, name, sizeof(id.name)); + strscpy(id.name, name, sizeof(id.name)); id.index = index; return snd_ctl_find_id(card, &id); } @@ -1355,7 +1355,7 @@ static int snd_mixer_oss_notify_handler(struct snd_card *card, int cmd) mixer->oss_dev_alloc = 1; mixer->card = card; if (*card->mixername) - strlcpy(mixer->name, card->mixername, sizeof(mixer->name)); + strscpy(mixer->name, card->mixername, sizeof(mixer->name)); else snprintf(mixer->name, sizeof(mixer->name), "mixer%i", card->number); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index be5714f1bb58..e5947281e5fc 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -729,7 +729,7 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device, init_waitqueue_head(&pcm->open_wait); INIT_LIST_HEAD(&pcm->list); if (id) - strlcpy(pcm->id, id, sizeof(pcm->id)); + strscpy(pcm->id, id, sizeof(pcm->id)); err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 9f3f8e953ff0..66ae1e248103 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -209,13 +209,13 @@ int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info) info->device = pcm->device; info->stream = substream->stream; info->subdevice = substream->number; - strlcpy(info->id, pcm->id, sizeof(info->id)); - strlcpy(info->name, pcm->name, sizeof(info->name)); + strscpy(info->id, pcm->id, sizeof(info->id)); + strscpy(info->name, pcm->name, sizeof(info->name)); info->dev_class = pcm->dev_class; info->dev_subclass = pcm->dev_subclass; info->subdevices_count = pstr->substream_count; info->subdevices_avail = pstr->substream_count - pstr->substream_opened; - strlcpy(info->subname, substream->name, sizeof(info->subname)); + strscpy(info->subname, substream->name, sizeof(info->subname)); return 0; } diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 257ad5206240..aca00af93afe 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1686,7 +1686,7 @@ int snd_rawmidi_new(struct snd_card *card, char *id, int device, INIT_LIST_HEAD(&rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams); if (id != NULL) - strlcpy(rmidi->id, id, sizeof(rmidi->id)); + strscpy(rmidi->id, id, sizeof(rmidi->id)); snd_device_initialize(&rmidi->dev, card); rmidi->dev.release = release_rawmidi_device; diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 2ddfe2226651..3f82c196de46 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -173,7 +173,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) snd_use_lock_init(&mdev->use_lock); /* copy and truncate the name of synth device */ - strlcpy(mdev->name, pinfo->name, sizeof(mdev->name)); + strscpy(mdev->name, pinfo->name, sizeof(mdev->name)); /* create MIDI coder */ if (snd_midi_event_new(MAX_MIDI_EVENT_BUF, &mdev->coder) < 0) { @@ -647,7 +647,7 @@ snd_seq_oss_midi_make_info(struct seq_oss_devinfo *dp, int dev, struct midi_info inf->device = dev; inf->dev_type = 0; /* FIXME: ?? */ inf->capabilities = 0; /* FIXME: ?? */ - strlcpy(inf->name, mdev->name, sizeof(inf->name)); + strscpy(inf->name, mdev->name, sizeof(inf->name)); snd_use_lock_free(&mdev->use_lock); return 0; } diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 11554d0412f0..136dc663887a 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -107,7 +107,7 @@ snd_seq_oss_synth_probe(struct device *_dev) snd_use_lock_init(&rec->use_lock); /* copy and truncate the name of synth device */ - strlcpy(rec->name, dev->name, sizeof(rec->name)); + strscpy(rec->name, dev->name, sizeof(rec->name)); /* registration */ spin_lock_irqsave(&register_lock, flags); @@ -616,7 +616,7 @@ snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_in inf->synth_subtype = 0; inf->nr_voices = 16; inf->device = dev; - strlcpy(inf->name, minf.name, sizeof(inf->name)); + strscpy(inf->name, minf.name, sizeof(inf->name)); } else { if ((rec = get_synthdev(dp, dev)) == NULL) return -ENXIO; @@ -624,7 +624,7 @@ snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_in inf->synth_subtype = rec->synth_subtype; inf->nr_voices = rec->nr_voices; inf->device = dev; - strlcpy(inf->name, rec->name, sizeof(inf->name)); + strscpy(inf->name, rec->name, sizeof(inf->name)); snd_use_lock_free(&rec->use_lock); } return 0; diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index f9f2fea58b32..b6a24fb5e76b 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1584,7 +1584,7 @@ static int snd_seq_ioctl_get_queue_info(struct snd_seq_client *client, info->queue = q->queue; info->owner = q->owner; info->locked = q->locked; - strlcpy(info->name, q->name, sizeof(info->name)); + strscpy(info->name, q->name, sizeof(info->name)); queuefree(q); return 0; diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 83be6b982a87..b9c2ce2b8d5a 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -327,7 +327,7 @@ int snd_seq_set_port_info(struct snd_seq_client_port * port, /* set port name */ if (info->name[0]) - strlcpy(port->name, info->name, sizeof(port->name)); + strscpy(port->name, info->name, sizeof(port->name)); /* set capabilities */ port->capability = info->capability; @@ -356,7 +356,7 @@ int snd_seq_get_port_info(struct snd_seq_client_port * port, return -EINVAL; /* get port name */ - strlcpy(info->name, port->name, sizeof(info->name)); + strscpy(info->name, port->name, sizeof(info->name)); /* get capabilities */ info->capability = port->capability; @@ -654,7 +654,7 @@ int snd_seq_event_port_attach(int client, /* Set up the port */ memset(&portinfo, 0, sizeof(portinfo)); portinfo.addr.client = client; - strlcpy(portinfo.name, portname ? portname : "Unnamed port", + strscpy(portinfo.name, portname ? portname : "Unnamed port", sizeof(portinfo.name)); portinfo.capability = cap; diff --git a/sound/core/timer.c b/sound/core/timer.c index 765ea66665a8..6898b1ac0d7f 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -959,7 +959,7 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid, timer->tmr_device = tid->device; timer->tmr_subdevice = tid->subdevice; if (id) - strlcpy(timer->id, id, sizeof(timer->id)); + strscpy(timer->id, id, sizeof(timer->id)); timer->sticks = 1; INIT_LIST_HEAD(&timer->device_list); INIT_LIST_HEAD(&timer->open_list_head); @@ -1659,8 +1659,8 @@ static int snd_timer_user_ginfo(struct file *file, ginfo->card = t->card ? t->card->number : -1; if (t->hw.flags & SNDRV_TIMER_HW_SLAVE) ginfo->flags |= SNDRV_TIMER_FLG_SLAVE; - strlcpy(ginfo->id, t->id, sizeof(ginfo->id)); - strlcpy(ginfo->name, t->name, sizeof(ginfo->name)); + strscpy(ginfo->id, t->id, sizeof(ginfo->id)); + strscpy(ginfo->name, t->name, sizeof(ginfo->name)); ginfo->resolution = t->hw.resolution; if (t->hw.resolution_min > 0) { ginfo->resolution_min = t->hw.resolution_min; @@ -1814,8 +1814,8 @@ static int snd_timer_user_info(struct file *file, info->card = t->card ? t->card->number : -1; if (t->hw.flags & SNDRV_TIMER_HW_SLAVE) info->flags |= SNDRV_TIMER_FLG_SLAVE; - strlcpy(info->id, t->id, sizeof(info->id)); - strlcpy(info->name, t->name, sizeof(info->name)); + strscpy(info->id, t->id, sizeof(info->id)); + strscpy(info->name, t->name, sizeof(info->name)); info->resolution = t->hw.resolution; if (copy_to_user(_info, info, sizeof(*_info))) err = -EFAULT; diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c index 0103d16f6f9f..ee973b7b8044 100644 --- a/sound/core/timer_compat.c +++ b/sound/core/timer_compat.c @@ -61,8 +61,8 @@ static int snd_timer_user_info_compat(struct file *file, info.card = t->card ? t->card->number : -1; if (t->hw.flags & SNDRV_TIMER_HW_SLAVE) info.flags |= SNDRV_TIMER_FLG_SLAVE; - strlcpy(info.id, t->id, sizeof(info.id)); - strlcpy(info.name, t->name, sizeof(info.name)); + strscpy(info.id, t->id, sizeof(info.id)); + strscpy(info.name, t->name, sizeof(info.name)); info.resolution = t->hw.resolution; if (copy_to_user(_info, &info, sizeof(*_info))) return -EFAULT; diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index 7bf0d5f3fedd..c82c7c1c0714 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -97,7 +97,7 @@ void snd_opl3_init_seq_oss(struct snd_opl3 *opl3, char *name) return; opl3->oss_seq_dev = dev; - strlcpy(dev->name, name, sizeof(dev->name)); + strscpy(dev->name, name, sizeof(dev->name)); arg = SNDRV_SEQ_DEVICE_ARGPTR(dev); arg->type = SYNTH_TYPE_FM; if (opl3->hardware < OPL3_HW_OPL3) { diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 08c10ac9d6c8..97d30a833ac8 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -290,7 +290,7 @@ int snd_opl3_load_patch(struct snd_opl3 *opl3, } if (name) - strlcpy(patch->name, name, sizeof(patch->name)); + strscpy(patch->name, name, sizeof(patch->name)); return 0; } diff --git a/sound/firewire/bebob/bebob_hwdep.c b/sound/firewire/bebob/bebob_hwdep.c index c362eb38ab90..8677e3ec8d14 100644 --- a/sound/firewire/bebob/bebob_hwdep.c +++ b/sound/firewire/bebob/bebob_hwdep.c @@ -80,7 +80,7 @@ hwdep_get_info(struct snd_bebob *bebob, void __user *arg) info.card = dev->card->index; *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); - strlcpy(info.device_name, dev_name(&dev->device), + strscpy(info.device_name, dev_name(&dev->device), sizeof(info.device_name)); if (copy_to_user(arg, &info, sizeof(info))) diff --git a/sound/firewire/dice/dice-hwdep.c b/sound/firewire/dice/dice-hwdep.c index f69f7996762f..ffc0b97782d6 100644 --- a/sound/firewire/dice/dice-hwdep.c +++ b/sound/firewire/dice/dice-hwdep.c @@ -79,7 +79,7 @@ static int hwdep_get_info(struct snd_dice *dice, void __user *arg) info.card = dev->card->index; *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); - strlcpy(info.device_name, dev_name(&dev->device), + strscpy(info.device_name, dev_name(&dev->device), sizeof(info.device_name)); if (copy_to_user(arg, &info, sizeof(info))) diff --git a/sound/firewire/digi00x/digi00x-hwdep.c b/sound/firewire/digi00x/digi00x-hwdep.c index 41c5857c612e..aadf7d724856 100644 --- a/sound/firewire/digi00x/digi00x-hwdep.c +++ b/sound/firewire/digi00x/digi00x-hwdep.c @@ -87,7 +87,7 @@ static int hwdep_get_info(struct snd_dg00x *dg00x, void __user *arg) info.card = dev->card->index; *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); - strlcpy(info.device_name, dev_name(&dev->device), + strscpy(info.device_name, dev_name(&dev->device), sizeof(info.device_name)); if (copy_to_user(arg, &info, sizeof(info))) diff --git a/sound/firewire/fireface/ff-hwdep.c b/sound/firewire/fireface/ff-hwdep.c index e73e8d2865a5..4b2e0dff5ddb 100644 --- a/sound/firewire/fireface/ff-hwdep.c +++ b/sound/firewire/fireface/ff-hwdep.c @@ -79,7 +79,7 @@ static int hwdep_get_info(struct snd_ff *ff, void __user *arg) info.card = dev->card->index; *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); - strlcpy(info.device_name, dev_name(&dev->device), + strscpy(info.device_name, dev_name(&dev->device), sizeof(info.device_name)); if (copy_to_user(arg, &info, sizeof(info))) diff --git a/sound/firewire/fireworks/fireworks_hwdep.c b/sound/firewire/fireworks/fireworks_hwdep.c index e93eb4616c5f..626c0c34b0b6 100644 --- a/sound/firewire/fireworks/fireworks_hwdep.c +++ b/sound/firewire/fireworks/fireworks_hwdep.c @@ -212,7 +212,7 @@ hwdep_get_info(struct snd_efw *efw, void __user *arg) info.card = dev->card->index; *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); - strlcpy(info.device_name, dev_name(&dev->device), + strscpy(info.device_name, dev_name(&dev->device), sizeof(info.device_name)); if (copy_to_user(arg, &info, sizeof(info))) diff --git a/sound/firewire/motu/motu-hwdep.c b/sound/firewire/motu/motu-hwdep.c index 0764a477052a..b5ced5d27758 100644 --- a/sound/firewire/motu/motu-hwdep.c +++ b/sound/firewire/motu/motu-hwdep.c @@ -86,7 +86,7 @@ static int hwdep_get_info(struct snd_motu *motu, void __user *arg) info.card = dev->card->index; *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); - strlcpy(info.device_name, dev_name(&dev->device), + strscpy(info.device_name, dev_name(&dev->device), sizeof(info.device_name)); if (copy_to_user(arg, &info, sizeof(info))) diff --git a/sound/firewire/oxfw/oxfw-hwdep.c b/sound/firewire/oxfw/oxfw-hwdep.c index eba33d050060..9e1b3e151bad 100644 --- a/sound/firewire/oxfw/oxfw-hwdep.c +++ b/sound/firewire/oxfw/oxfw-hwdep.c @@ -79,7 +79,7 @@ static int hwdep_get_info(struct snd_oxfw *oxfw, void __user *arg) info.card = dev->card->index; *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); - strlcpy(info.device_name, dev_name(&dev->device), + strscpy(info.device_name, dev_name(&dev->device), sizeof(info.device_name)); if (copy_to_user(arg, &info, sizeof(info))) diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c index 6f38335fe10b..74eed9505665 100644 --- a/sound/firewire/tascam/tascam-hwdep.c +++ b/sound/firewire/tascam/tascam-hwdep.c @@ -154,7 +154,7 @@ static int hwdep_get_info(struct snd_tscm *tscm, void __user *arg) info.card = dev->card->index; *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); - strlcpy(info.device_name, dev_name(&dev->device), + strscpy(info.device_name, dev_name(&dev->device), sizeof(info.device_name)); if (copy_to_user(arg, &info, sizeof(info))) diff --git a/sound/i2c/i2c.c b/sound/i2c/i2c.c index a684faa771ef..847e3b6ca601 100644 --- a/sound/i2c/i2c.c +++ b/sound/i2c/i2c.c @@ -84,7 +84,7 @@ int snd_i2c_bus_create(struct snd_card *card, const char *name, list_add_tail(&bus->buses, &master->buses); bus->master = master; } - strlcpy(bus->name, name, sizeof(bus->name)); + strscpy(bus->name, name, sizeof(bus->name)); err = snd_device_new(card, SNDRV_DEV_BUS, bus, &ops); if (err < 0) { snd_i2c_bus_free(bus); @@ -108,7 +108,7 @@ int snd_i2c_device_create(struct snd_i2c_bus *bus, const char *name, if (device == NULL) return -ENOMEM; device->addr = addr; - strlcpy(device->name, name, sizeof(device->name)); + strscpy(device->name, name, sizeof(device->name)); list_add_tail(&device->list, &bus->devices); device->bus = bus; *rdevice = device; diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c index 593c6e959afe..47bffe623105 100644 --- a/sound/isa/ad1848/ad1848.c +++ b/sound/isa/ad1848/ad1848.c @@ -95,8 +95,8 @@ static int snd_ad1848_probe(struct device *dev, unsigned int n) if (error < 0) goto out; - strlcpy(card->driver, "AD1848", sizeof(card->driver)); - strlcpy(card->shortname, chip->pcm->name, sizeof(card->shortname)); + strscpy(card->driver, "AD1848", sizeof(card->driver)); + strscpy(card->shortname, chip->pcm->name, sizeof(card->shortname)); if (!thinkpad[n]) snprintf(card->longname, sizeof(card->longname), diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index 2135963eba78..bcbea6962d7e 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -95,8 +95,8 @@ static int snd_cs4231_probe(struct device *dev, unsigned int n) if (error < 0) goto out; - strlcpy(card->driver, "CS4231", sizeof(card->driver)); - strlcpy(card->shortname, chip->pcm->name, sizeof(card->shortname)); + strscpy(card->driver, "CS4231", sizeof(card->driver)); + strscpy(card->shortname, chip->pcm->name, sizeof(card->shortname)); if (dma2[n] < 0) snprintf(card->longname, sizeof(card->longname), diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index fa3c39cff5f8..fb9d8a4b7084 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -405,8 +405,8 @@ static int snd_cs423x_probe(struct snd_card *card, int dev) if (err < 0) return err; } - strlcpy(card->driver, chip->pcm->name, sizeof(card->driver)); - strlcpy(card->shortname, chip->pcm->name, sizeof(card->shortname)); + strscpy(card->driver, chip->pcm->name, sizeof(card->driver)); + strscpy(card->shortname, chip->pcm->name, sizeof(card->shortname)); if (dma2[dev] < 0) snprintf(card->longname, sizeof(card->longname), "%s at 0x%lx, irq %i, dma %i", diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 64610571a5e1..766ab43aaf77 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -133,8 +133,8 @@ static int snd_es1688_probe(struct snd_card *card, unsigned int n) if (error < 0) return error; - strlcpy(card->driver, "ES1688", sizeof(card->driver)); - strlcpy(card->shortname, chip->pcm->name, sizeof(card->shortname)); + strscpy(card->driver, "ES1688", sizeof(card->driver)); + strscpy(card->shortname, chip->pcm->name, sizeof(card->shortname)); snprintf(card->longname, sizeof(card->longname), "%s at 0x%lx, irq %i, dma %i", chip->pcm->name, chip->port, chip->irq, chip->dma8); diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index 270af863e198..8635a2b6b36b 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -388,7 +388,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, return err; /* fill in codec header */ - strlcpy(p->codec_name, info.codec_name, sizeof(p->codec_name)); + strscpy(p->codec_name, info.codec_name, sizeof(p->codec_name)); p->func_nr = func_nr; p->mode = le16_to_cpu(funcdesc_h.flags_play_rec); switch (le16_to_cpu(funcdesc_h.VOC_type)) { diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 3f703b4a304d..5de5506e7e60 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -482,7 +482,7 @@ int snd_sbmixer_add_ctl(struct snd_sb *chip, const char *name, int index, int ty ctl = snd_ctl_new1(&newctls[type], chip); if (! ctl) return -ENOMEM; - strlcpy(ctl->id.name, name, sizeof(ctl->id.name)); + strscpy(ctl->id.name, name, sizeof(ctl->id.name)); ctl->id.index = index; ctl->private_value = value; if ((err = snd_ctl_add(chip->card, ctl)) < 0) diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 38f25e97538f..49679aa8631d 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -355,8 +355,8 @@ static int mixer_ioctl(struct file *file, u_int cmd, u_long arg) { mixer_info info; memset(&info, 0, sizeof(info)); - strlcpy(info.id, dmasound.mach.name2, sizeof(info.id)); - strlcpy(info.name, dmasound.mach.name2, sizeof(info.name)); + strscpy(info.id, dmasound.mach.name2, sizeof(info.id)); + strscpy(info.name, dmasound.mach.name2, sizeof(info.name)); info.modify_counter = mixer.modify_counter; if (copy_to_user((void __user *)arg, &info, sizeof(info))) return -EFAULT; diff --git a/sound/pci/cs5535audio/cs5535audio_olpc.c b/sound/pci/cs5535audio/cs5535audio_olpc.c index 4e295303b041..110d3209441b 100644 --- a/sound/pci/cs5535audio/cs5535audio_olpc.c +++ b/sound/pci/cs5535audio/cs5535audio_olpc.c @@ -158,13 +158,13 @@ int olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97) /* drop the original AD1888 HPF control */ memset(&elem, 0, sizeof(elem)); elem.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strlcpy(elem.name, "High Pass Filter Enable", sizeof(elem.name)); + strscpy(elem.name, "High Pass Filter Enable", sizeof(elem.name)); snd_ctl_remove_id(card, &elem); /* drop the original V_REFOUT control */ memset(&elem, 0, sizeof(elem)); elem.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strlcpy(elem.name, "V_REFOUT Enable", sizeof(elem.name)); + strscpy(elem.name, "V_REFOUT Enable", sizeof(elem.name)); snd_ctl_remove_id(card, &elem); /* add the OLPC-specific controls */ diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 3f48ad0e27e7..81dfc6a76b18 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -433,7 +433,7 @@ int ct_alsa_pcm_create(struct ct_atc *atc, pcm->private_data = atc; pcm->info_flags = 0; pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX; - strlcpy(pcm->name, device_name, sizeof(pcm->name)); + strscpy(pcm->name, device_name, sizeof(pcm->name)); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ct_pcm_playback_ops); diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 29b7720d7961..353934c88cbd 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -168,9 +168,9 @@ static int snd_card_emu10k1_probe(struct pci_dev *pci, } #endif - strlcpy(card->driver, emu->card_capabilities->driver, + strscpy(card->driver, emu->card_capabilities->driver, sizeof(card->driver)); - strlcpy(card->shortname, emu->card_capabilities->name, + strscpy(card->shortname, emu->card_capabilities->name, sizeof(card->shortname)); snprintf(card->longname, sizeof(card->longname), "%s (rev.%d, serial:0x%x) at 0x%lx, irq %i", diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index bd70e112ffd7..24a2fd706d69 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1869,7 +1869,7 @@ int snd_emu10k1_create(struct snd_card *card, emu->serial); if (!*card->id && c->id) - strlcpy(card->id, c->id, sizeof(card->id)); + strscpy(card->id, c->id, sizeof(card->id)); is_audigy = emu->audigy = c->emu10k2_chip; diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 4e76ed0e91d5..80ef62a4a7c0 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -940,7 +940,7 @@ static int snd_emu10k1_list_controls(struct snd_emu10k1 *emu, memset(gctl, 0, sizeof(*gctl)); id = &ctl->kcontrol->id; gctl->id.iface = (__force int)id->iface; - strlcpy(gctl->id.name, id->name, sizeof(gctl->id.name)); + strscpy(gctl->id.name, id->name, sizeof(gctl->id.name)); gctl->id.index = id->index; gctl->id.device = id->device; gctl->id.subdevice = id->subdevice; @@ -976,7 +976,7 @@ static int snd_emu10k1_icode_poke(struct snd_emu10k1 *emu, err = snd_emu10k1_verify_controls(emu, icode, in_kernel); if (err < 0) goto __error; - strlcpy(emu->fx8010.name, icode->name, sizeof(emu->fx8010.name)); + strscpy(emu->fx8010.name, icode->name, sizeof(emu->fx8010.name)); /* stop FX processor - this may be dangerous, but it's better to miss some samples than generate wrong ones - [jk] */ if (emu->audigy) @@ -1015,7 +1015,7 @@ static int snd_emu10k1_icode_peek(struct snd_emu10k1 *emu, int err; mutex_lock(&emu->fx8010.lock); - strlcpy(icode->name, emu->fx8010.name, sizeof(icode->name)); + strscpy(icode->name, emu->fx8010.name, sizeof(icode->name)); /* ok, do the main job */ err = snd_emu10k1_gpr_peek(emu, icode); if (err >= 0) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 34332d008b27..c6be14c4f311 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2768,7 +2768,7 @@ static int snd_es1968_create(struct snd_card *card, if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) { dev_info(card->dev, "detected TEA575x radio type %s\n", get_tea575x_gpio(chip)->name); - strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name, + strscpy(chip->tea.card, get_tea575x_gpio(chip)->name, sizeof(chip->tea.card)); break; } diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 0a95032fd297..c6ad6235a669 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1300,7 +1300,7 @@ static int snd_fm801_create(struct snd_card *card, chip->tea575x_tuner |= tuner_only; } if (!(chip->tea575x_tuner & TUNER_DISABLED)) { - strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name, + strscpy(chip->tea.card, get_tea575x_gpio(chip)->name, sizeof(chip->tea.card)); } #endif diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 4dc01647753c..1a001ecf7f63 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -764,7 +764,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, } if (!name) return 0; - strlcpy(label, name, maxlen); + strscpy(label, name, maxlen); return 1; } EXPORT_SYMBOL_GPL(snd_hda_get_pin_label); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 687216e74526..bc3b51eb8390 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4009,7 +4009,7 @@ int snd_hda_add_imux_item(struct hda_codec *codec, sizeof(imux->items[imux->num_items].label), "%s %d", label, label_idx); else - strlcpy(imux->items[imux->num_items].label, label, + strscpy(imux->items[imux->num_items].label, label, sizeof(imux->items[imux->num_items].label)); imux->items[imux->num_items].index = index; imux->num_items++; diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 80016b7b6849..9087981cd1f7 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -735,7 +735,7 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec, &pcm); if (err < 0) return err; - strlcpy(pcm->name, cpcm->name, sizeof(pcm->name)); + strscpy(pcm->name, cpcm->name, sizeof(pcm->name)); apcm = kzalloc(sizeof(*apcm), GFP_KERNEL); if (apcm == NULL) { snd_device_free(chip->card, pcm); diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 136477ed46ae..9e97443795f8 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -260,7 +260,7 @@ int snd_hdmi_parse_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e, codec_info(codec, "HDMI: out of range MNL %d\n", mnl); goto out_fail; } else - strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl + 1); + strscpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl + 1); for (i = 0; i < e->sad_count; i++) { if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) { diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8060cc86dfea..5e40944e7342 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5721,7 +5721,7 @@ static void fill_pcm_stream_name(char *str, size_t len, const char *sfx, if (*str) return; - strlcpy(str, chip_name, len); + strscpy(str, chip_name, len); /* drop non-alnum chars after a space */ for (p = strchr(str, ' '); p; p = strchr(p + 1, ' ')) { diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 18cdd67bcb86..84c8e3e06ca8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2037,7 +2037,7 @@ static int azx_first_init(struct azx *chip) return -EBUSY; strcpy(card->driver, "HDA-Intel"); - strlcpy(card->shortname, driver_short_names[chip->driver_type], + strscpy(card->shortname, driver_short_names[chip->driver_type], sizeof(card->shortname)); snprintf(card->longname, sizeof(card->longname), "%s at 0x%lx irq %i", diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 588059428d8f..b8b568046592 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -530,7 +530,7 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, !is_jack_detectable(codec, nid); if (base_name) - strlcpy(name, base_name, sizeof(name)); + strscpy(name, base_name, sizeof(name)); else snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), NULL); if (phantom_jack) diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index e57a55cebc5a..f0f8324b08b6 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -413,7 +413,7 @@ static struct snd_kcontrol *ctl_find(struct snd_card *card, { struct snd_ctl_elem_id sid = {0}; - strlcpy(sid.name, name, sizeof(sid.name)); + strscpy(sid.name, name, sizeof(sid.name)); sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; return snd_ctl_find_id(card, &sid); } diff --git a/sound/pci/ice1712/psc724.c b/sound/pci/ice1712/psc724.c index 7aa3f92040d0..82cf365cda10 100644 --- a/sound/pci/ice1712/psc724.c +++ b/sound/pci/ice1712/psc724.c @@ -189,12 +189,12 @@ static void psc724_set_jack_state(struct snd_ice1712 *ice, bool hp_connected) /* notify about master speaker mute change */ memset(&elem_id, 0, sizeof(elem_id)); elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strlcpy(elem_id.name, "Master Speakers Playback Switch", + strscpy(elem_id.name, "Master Speakers Playback Switch", sizeof(elem_id.name)); kctl = snd_ctl_find_id(ice->card, &elem_id); snd_ctl_notify(ice->card, SNDRV_CTL_EVENT_MASK_VALUE, &kctl->id); /* and headphone mute change */ - strlcpy(elem_id.name, spec->wm8776.ctl[WM8776_CTL_HP_SW].name, + strscpy(elem_id.name, spec->wm8776.ctl[WM8776_CTL_HP_SW].name, sizeof(elem_id.name)); kctl = snd_ctl_find_id(ice->card, &elem_id); snd_ctl_notify(ice->card, SNDRV_CTL_EVENT_MASK_VALUE, &kctl->id); diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index 0e3e04aa9faf..0dfa093f7dca 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -771,7 +771,7 @@ static struct snd_kcontrol *ctl_find(struct snd_card *card, { struct snd_ctl_elem_id sid = {0}; - strlcpy(sid.name, name, sizeof(sid.name)); + strscpy(sid.name, name, sizeof(sid.name)); sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; return snd_ctl_find_id(card, &sid); } diff --git a/sound/pci/ice1712/wm8776.c b/sound/pci/ice1712/wm8776.c index d96008df880d..6eda86119dff 100644 --- a/sound/pci/ice1712/wm8776.c +++ b/sound/pci/ice1712/wm8776.c @@ -38,7 +38,7 @@ static void snd_wm8776_activate_ctl(struct snd_wm8776 *wm, unsigned int index_offset; memset(&elem_id, 0, sizeof(elem_id)); - strlcpy(elem_id.name, ctl_name, sizeof(elem_id.name)); + strscpy(elem_id.name, ctl_name, sizeof(elem_id.name)); elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; kctl = snd_ctl_find_id(card, &elem_id); if (!kctl) diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index cdd8db79bcfa..491c90f83fbc 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -669,7 +669,7 @@ static int lola_create(struct snd_card *card, struct pci_dev *pci, } strcpy(card->driver, "Lola"); - strlcpy(card->shortname, "Digigram Lola", sizeof(card->shortname)); + strscpy(card->shortname, "Digigram Lola", sizeof(card->shortname)); snprintf(card->longname, sizeof(card->longname), "%s at 0x%lx irq %i", card->shortname, chip->bar[0].addr, chip->irq); diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index f647c7ed00c4..684faaf40f31 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -601,7 +601,7 @@ int lola_create_pcm(struct lola *chip) &pcm); if (err < 0) return err; - strlcpy(pcm->name, "Digigram Lola", sizeof(pcm->name)); + strscpy(pcm->name, "Digigram Lola", sizeof(pcm->name)); pcm->private_data = chip; for (i = 0; i < 2; i++) { if (chip->pcm[i].num_streams) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 04e878a0f773..b66711574b1a 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6329,7 +6329,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, memset(&hdspm_version, 0, sizeof(hdspm_version)); hdspm_version.card_type = hdspm->io_type; - strlcpy(hdspm_version.cardname, hdspm->card_name, + strscpy(hdspm_version.cardname, hdspm->card_name, sizeof(hdspm_version.cardname)); hdspm_version.serial = hdspm->serial; hdspm_version.firmware_rev = hdspm->firmware_rev; diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index 9554a0c506af..a6c1905039de 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -49,7 +49,7 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter) return -EINVAL; /* ignored */ memset(&info, 0, sizeof(struct i2c_board_info)); - strlcpy(info.type, "keywest", I2C_NAME_SIZE); + strscpy(info.type, "keywest", I2C_NAME_SIZE); info.addr = keywest_ctx->addr; client = i2c_new_client_device(adapter, &info); if (IS_ERR(client)) diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index daa58b5f941e..cad1cd1bfdf0 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -1707,7 +1707,7 @@ int q6afe_vote_lpass_core_hw(struct device *dev, uint32_t hw_block_id, pkt->hdr.token = hw_block_id; pkt->hdr.opcode = AFE_CMD_REMOTE_LPASS_CORE_HW_VOTE_REQUEST; vote_cfg->hw_block_id = hw_block_id; - strlcpy(vote_cfg->client_name, client_name, + strscpy(vote_cfg->client_name, client_name, sizeof(vote_cfg->client_name)); ret = afe_apr_send_pkt(afe, pkt, NULL, diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6e670b3e92a0..6dd5659db44c 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1472,7 +1472,7 @@ static int rsnd_kctrl_info(struct snd_kcontrol *kctrl, uinfo->value.enumerated.items = cfg->max; if (uinfo->value.enumerated.item >= cfg->max) uinfo->value.enumerated.item = cfg->max - 1; - strlcpy(uinfo->value.enumerated.name, + strscpy(uinfo->value.enumerated.name, cfg->texts[uinfo->value.enumerated.item], sizeof(uinfo->value.enumerated.name)); } else { diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c index 010976d9ceb2..cd4a0bc6d278 100644 --- a/sound/usb/bcd2000/bcd2000.c +++ b/sound/usb/bcd2000/bcd2000.c @@ -300,7 +300,7 @@ static int bcd2000_init_midi(struct bcd2000 *bcd2k) if (ret < 0) return ret; - strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + strscpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; rmidi->private_data = bcd2k; diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 3b6bb2cbe886..4981753652a7 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -804,7 +804,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) } cdev->pcm->private_data = cdev; - strlcpy(cdev->pcm->name, cdev->product_name, sizeof(cdev->pcm->name)); + strscpy(cdev->pcm->name, cdev->product_name, sizeof(cdev->pcm->name)); memset(cdev->sub_playback, 0, sizeof(cdev->sub_playback)); memset(cdev->sub_capture, 0, sizeof(cdev->sub_capture)); diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 2af3b7eb0a88..e03481caf7f6 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -477,9 +477,9 @@ static int init_card(struct snd_usb_caiaqdev *cdev) usb_string(usb_dev, usb_dev->descriptor.iProduct, cdev->product_name, CAIAQ_USB_STR_LEN); - strlcpy(card->driver, MODNAME, sizeof(card->driver)); - strlcpy(card->shortname, cdev->product_name, sizeof(card->shortname)); - strlcpy(card->mixername, cdev->product_name, sizeof(card->mixername)); + strscpy(card->driver, MODNAME, sizeof(card->driver)); + strscpy(card->shortname, cdev->product_name, sizeof(card->shortname)); + strscpy(card->mixername, cdev->product_name, sizeof(card->mixername)); /* if the id was not passed as module option, fill it with a shortened * version of the product string which does not contain any diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 512fbb3ee604..c656d0162432 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -125,7 +125,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) if (ret < 0) return ret; - strlcpy(rmidi->name, device->product_name, sizeof(rmidi->name)); + strscpy(rmidi->name, device->product_name, sizeof(rmidi->name)); rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; rmidi->private_data = device; diff --git a/sound/usb/card.c b/sound/usb/card.c index d731ca62d599..85e79b9ecb08 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -474,7 +474,7 @@ static void usb_audio_make_shortname(struct usb_device *dev, else if (quirk && quirk->product_name) s = quirk->product_name; if (s && *s) { - strlcpy(card->shortname, s, sizeof(card->shortname)); + strscpy(card->shortname, s, sizeof(card->shortname)); return; } @@ -506,7 +506,7 @@ static void usb_audio_make_longname(struct usb_device *dev, if (preset && preset->profile_name) s = preset->profile_name; if (s && *s) { - strlcpy(card->longname, s, sizeof(card->longname)); + strscpy(card->longname, s, sizeof(card->longname)); return; } diff --git a/sound/usb/hiface/chip.c b/sound/usb/hiface/chip.c index b2d9623e9934..c2824188d142 100644 --- a/sound/usb/hiface/chip.c +++ b/sound/usb/hiface/chip.c @@ -80,12 +80,12 @@ static int hiface_chip_create(struct usb_interface *intf, return ret; } - strlcpy(card->driver, DRIVER_NAME, sizeof(card->driver)); + strscpy(card->driver, DRIVER_NAME, sizeof(card->driver)); if (quirk && quirk->device_name) - strlcpy(card->shortname, quirk->device_name, sizeof(card->shortname)); + strscpy(card->shortname, quirk->device_name, sizeof(card->shortname)); else - strlcpy(card->shortname, "M2Tech generic audio", sizeof(card->shortname)); + strscpy(card->shortname, "M2Tech generic audio", sizeof(card->shortname)); strlcat(card->longname, card->shortname, sizeof(card->longname)); len = strlcat(card->longname, " at ", sizeof(card->longname)); diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index d942179ca095..71f17f02f341 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -594,7 +594,7 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) pcm->private_data = rt; pcm->private_free = hiface_pcm_free; - strlcpy(pcm->name, "USB-SPDIF Audio", sizeof(pcm->name)); + strscpy(pcm->name, "USB-SPDIF Audio", sizeof(pcm->name)); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_ops); snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC, NULL, 0, 0); diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 12b15ed59eaa..60569d64c990 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1556,7 +1556,7 @@ static void check_no_speaker_on_headset(struct snd_kcontrol *kctl, if (!found) return; - strlcpy(kctl->id.name, "Headphone", sizeof(kctl->id.name)); + strscpy(kctl->id.name, "Headphone", sizeof(kctl->id.name)); } static const struct usb_feature_control_info *get_feature_control_info(int control) @@ -1691,7 +1691,7 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, break; default: if (!len) - strlcpy(kctl->id.name, audio_feature_info[control-1].name, + strscpy(kctl->id.name, audio_feature_info[control-1].name, sizeof(kctl->id.name)); break; } @@ -1770,7 +1770,7 @@ static void get_connector_control_name(struct usb_mixer_interface *mixer, int name_len = get_term_name(mixer->chip, term, name, name_size, 0); if (name_len == 0) - strlcpy(name, "Unknown", name_size); + strscpy(name, "Unknown", name_size); /* * sound/core/ctljack.c has a convention of naming jack controls @@ -2490,7 +2490,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, if (check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name))) { /* nothing */ ; } else if (info->name) { - strlcpy(kctl->id.name, info->name, sizeof(kctl->id.name)); + strscpy(kctl->id.name, info->name, sizeof(kctl->id.name)); } else { if (extension_unit) nameid = uac_extension_unit_iExtension(desc, state->mixer->protocol); @@ -2503,7 +2503,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, kctl->id.name, sizeof(kctl->id.name)); if (!len) - strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); + strscpy(kctl->id.name, name, sizeof(kctl->id.name)); } append_ctl_name(kctl, " "); append_ctl_name(kctl, valinfo->suffix); @@ -2743,7 +2743,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, kctl->id.name, sizeof(kctl->id.name), 0); /* ... or use the fixed string "USB" as the last resort */ if (!len) - strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); + strscpy(kctl->id.name, "USB", sizeof(kctl->id.name)); /* and add the proper suffix */ if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR || diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index df036a359f2f..abad1d61a536 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2725,7 +2725,7 @@ static int snd_pioneer_djm_controls_info(struct snd_kcontrol *kctl, struct snd_c if (info->value.enumerated.item >= count) info->value.enumerated.item = count - 1; name = group->options[info->value.enumerated.item].name; - strlcpy(info->value.enumerated.name, name, sizeof(info->value.enumerated.name)); + strscpy(info->value.enumerated.name, name, sizeof(info->value.enumerated.name)); info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; info->count = 1; info->value.enumerated.items = count; diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 49fcd2505443..691b95466d0f 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -569,7 +569,7 @@ static int add_new_ctl(struct usb_mixer_interface *mixer, } kctl->private_free = snd_usb_mixer_elem_free; - strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); + strscpy(kctl->id.name, name, sizeof(kctl->id.name)); err = snd_usb_mixer_add_control(&elem->head, kctl); if (err < 0) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 4bbec56c7df3..560c2ade829d 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -961,7 +961,7 @@ static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer, } kctl->private_free = snd_usb_mixer_elem_free; - strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); + strscpy(kctl->id.name, name, sizeof(kctl->id.name)); err = snd_usb_mixer_add_control(&elem->head, kctl); if (err < 0) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index bd63a9ce6a70..b7b6f3834ed5 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -1076,7 +1076,7 @@ static int add_new_ctl(struct usb_mixer_interface *mixer, else kctl->private_free = snd_usb_mixer_elem_free; - strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); + strscpy(kctl->id.name, name, sizeof(kctl->id.name)); err = snd_usb_mixer_add_control(&elem->head, kctl); if (err < 0) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 9f9fcd2749f2..5b351f4ca543 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1790,7 +1790,7 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) /* setup private data which can be retrieved when required */ pcm->private_data = ctx; pcm->info_flags = 0; - strlcpy(pcm->name, card->shortname, strlen(card->shortname)); + strscpy(pcm->name, card->shortname, strlen(card->shortname)); /* setup the ops for playabck */ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &had_pcm_ops); diff --git a/sound/xen/xen_snd_front_cfg.c b/sound/xen/xen_snd_front_cfg.c index eda077c8087a..63b0398c3276 100644 --- a/sound/xen/xen_snd_front_cfg.c +++ b/sound/xen/xen_snd_front_cfg.c @@ -398,7 +398,7 @@ static int cfg_device(struct xen_snd_front_info *front_info, str = xenbus_read(XBT_NIL, device_path, XENSND_FIELD_DEVICE_NAME, NULL); if (!IS_ERR(str)) { - strlcpy(pcm_instance->name, str, sizeof(pcm_instance->name)); + strscpy(pcm_instance->name, str, sizeof(pcm_instance->name)); kfree(str); }

3 3

[GIT PULL] ASoC fixes for v5.11-rc2
by Mark Brown 07 Jan '21

07 Jan '21

The following changes since commit 5c8fe583cce542aa0b84adc939ce85293de36e5e: Linux 5.11-rc1 (2020-12-27 15:30:22 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.11-rc2 for you to fetch changes up to 1f092d1c8819679d78a7d9c62a46d4939d217a9d: ASoC: AMD Renoir - add DMI entry for Lenovo ThinkPad X395 (2020-12-28 14:22:44 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.11 A collection of mostly driver specific fixes, plus a maintainership update for TI and a fix for DAPM driver removal paths. ---------------------------------------------------------------- Bard Liao (1): ASoC: max98373: don't access volatile registers in bias level off Cezary Rojewski (1): ASoC: Intel: haswell: Add missing pm_ops Colin Ian King (3): ASoC: codecs: fix spelling mistake in Kconfig "comunicate" -> "communicate" ASoC: SOF: Fix spelling mistake in Kconfig "ond" -> "and" ASoC: atmel: fix spelling mistake in Kconfig "programable" -> "programmable" Dan Carpenter (1): ASoC: Intel: fix error code cnl_set_dsp_D0() Jaroslav Kysela (2): ASoC: AMD Renoir - add DMI entry for Lenovo ThinkPad E14 Gen 2 ASoC: AMD Renoir - add DMI entry for Lenovo ThinkPad X395 Jerome Brunet (2): ASoC: meson: axg-tdmin: fix axg skew offset ASoC: meson: axg-tdm-interface: fix loopback Kuninori Morimoto (1): ASoC: rsnd: don't call clk_disable_unprepare() if can't use Mark Brown (3): Merge series "ASoC: ti: Maintainer mail address change" from Peter Ujfalusi <peter.ujfalusi(a)ti.com>: Merge series "Platform driver update to support playback recover after resume" from Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org>: Merge tag 'v5.11-rc1' into asoc-5.11 Peter Ujfalusi (2): MAINTAINERS: Update email address for TI ASoC and twl4030 codec drivers ASoC: dt-bindings: ti, j721e: Update maintainer and author information Ravulapati Vishnu vardhan rao (1): ASoC: amd: Replacing MSI with Legacy IRQ model Shuming Fan (1): ASoC: rt711: mutex between calibration and power state changes Srinivasa Rao Mandadapu (2): ASoC: qcom: Fix incorrect volatile registers ASoC: qcom: Add support for playback recover after resume Thomas Hebb (1): ASoC: dapm: remove widget from dirty list on free shengjiu wang (1): ASoC: imx-hdmi: Fix warning of the uninitialized variable ret .../bindings/sound/ti,j721e-cpb-audio.yaml | 4 +- .../bindings/sound/ti,j721e-cpb-ivi-audio.yaml | 4 +- MAINTAINERS | 6 +-- sound/soc/amd/raven/pci-acp3x.c | 16 ++----- sound/soc/amd/renoir/rn-pci-acp3x.c | 14 ++++++ sound/soc/atmel/Kconfig | 4 +- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/max98373-i2c.c | 20 +++++++++ sound/soc/codecs/max98373-sdw.c | 20 +++++++++ sound/soc/codecs/max98373.c | 34 +++++++++++++-- sound/soc/codecs/max98373.h | 8 ++++ sound/soc/codecs/rt711.c | 6 +++ sound/soc/fsl/imx-hdmi.c | 1 + sound/soc/intel/boards/haswell.c | 1 + sound/soc/intel/skylake/cnl-sst.c | 1 + sound/soc/meson/axg-tdm-interface.c | 14 +++++- sound/soc/meson/axg-tdmin.c | 13 +----- sound/soc/qcom/lpass-cpu.c | 20 +-------- sound/soc/qcom/lpass-platform.c | 50 +++++++++++++++------- sound/soc/sh/rcar/adg.c | 18 ++++---- sound/soc/soc-dapm.c | 1 + sound/soc/sof/Kconfig | 2 +- 22 files changed, 181 insertions(+), 78 deletions(-)

2 1

[PATCH] ASoC: hdmi-codec: Fix return value in hdmi_codec_set_jack()
by Stephan Gerhold 07 Jan '21

07 Jan '21

Sound is broken on the DragonBoard 410c (apq8016_sbc) since 5.10: hdmi-audio-codec hdmi-audio-codec.1.auto: ASoC: error at snd_soc_component_set_jack on hdmi-audio-codec.1.auto: -95 qcom-apq8016-sbc 7702000.sound: Failed to set jack: -95 ADV7533: ASoC: error at snd_soc_link_init on ADV7533: -95 hdmi-audio-codec hdmi-audio-codec.1.auto: ASoC: error at snd_soc_component_set_jack on hdmi-audio-codec.1.auto: -95 qcom-apq8016-sbc: probe of 7702000.sound failed with error -95 This happens because apq8016_sbc calls snd_soc_component_set_jack() on all codec DAIs and attempts to ignore failures with return code -ENOTSUPP. -ENOTSUPP is also excluded from error logging in soc_component_ret(). However, hdmi_codec_set_jack() returns -E*OP*NOTSUPP if jack detection is not supported, which is not handled in apq8016_sbc and soc_component_ret(). Make it return -ENOTSUPP instead to fix sound and silence the errors. Cc: Cheng-Yi Chiang <cychiang(a)chromium.org> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Fixes: 55c5cc63ab32 ("ASoC: hdmi-codec: Use set_jack ops to set jack") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index d5fcc4db8284..0f3ac22f2cf8 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -717,7 +717,7 @@ static int hdmi_codec_set_jack(struct snd_soc_component *component, void *data) { struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); - int ret = -EOPNOTSUPP; + int ret = -ENOTSUPP; if (hcp->hcd.ops->hook_plugged_cb) { hcp->jack = jack; -- 2.30.0

2 2

[PATCH v4 0/5] ALSA: usb-audio: Fix regression for Pioneer devices
by Takashi Iwai 07 Jan '21

07 Jan '21

Hi, here is the revised patch set for addressing the regression reported for Pioneer devices with the implicit feedback mode. Takashi === Takashi Iwai (5): ALSA: usb-audio: Fix the missing endpoints creations for quirks ALSA: usb-audio: Choose audioformat of a counter-part substream ALSA: usb-audio: Avoid unnecessary interface re-setup ALSA: usb-audio: Annotate the endpoint index in audioformat ALSA: usb-audio: Fix implicit feedback sync setup for Pioneer devices sound/usb/card.c | 5 +-- sound/usb/card.h | 3 ++ sound/usb/endpoint.c | 84 ++++++++++++++++++++++++++++++++++------ sound/usb/endpoint.h | 2 +- sound/usb/implicit.c | 55 +++++++++++++++++++------- sound/usb/quirks-table.h | 6 +++ sound/usb/quirks.c | 54 +++++++++++++++++++++----- sound/usb/usbaudio.h | 1 + 8 files changed, 169 insertions(+), 41 deletions(-) -- 2.26.2

1 5

[PATCH 00/41] USB audio refactoring for better implicit feedback support
by Takashi Iwai 07 Jan '21

07 Jan '21

Hi, here is a largish patch set for USB-audio. Its main goal is to address the long-standing issues wrt the implicit feedback mode. With the current driver, when the implicit fb is used, the full duplex doesn't work properly unless you start both streams *very* carefully. This patch set tries to ease such the configuration mismatch and fix many other bugs. In short, with this patch set applied, USB-audio driver will try to restrict the hw_params to match with the already opened counter-part stream automatically. So individual aplay and arecord invocation should work now, as well as PulseAudio. Also, the patch set includes more fixes, e.g. the generic detection of the implicit feedback mode on UAC2/3, and the fix for the sample rate selection on UAC2/3, as well as a few more quirks to follow the new implicit fb model. The latest patches are available in topic/usb-audio-refactoring branch. I specially thank Keith Milner and Dylan Robinson for their time spent for lots of testing. Without their patient help, I couldn't finish those patches, as I myself don't own such devices. Takashi === Dylan Robinson (1): ALSA: usb-audio: Fix MOTU M-Series quirks Takashi Iwai (40): ALSA: usb-audio: Handle discrete rates properly in hw constraints ALSA: usb-audio: Don't call usb_set_interface() at trigger callback ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3 ALSA: usb-audio: Check implicit feedback EP generically for UAC2 ALSA: usb-audio: Add snd_usb_get_endpoint() helper ALSA: usb-audio: Set and clear sync EP link properly ALSA: usb-audio: Improve some debug prints ALSA: usb-audio: Track implicit fb sync endpoint in audioformat list ALSA: usb-audio: Move snd_usb_autoresume() call out of setup_hw_info() ALSA: usb-audio: Add hw constraint for implicit fb sync ALSA: usb-audio: Simplify hw_params rules ALSA: usb-audio: Drop debug.h ALSA: usb-audio: Avoid doubly initialization for implicit fb ALSA: usb-audio: Create endpoint objects at parsing phase ALSA: usb-audio: Drop keep_interface flag again ALSA: usb-audio: Add snd_usb_get_host_interface() helper ALSA: usb-audio: Don't set altsetting before initializing sample rate ALSA: usb-audio: Pass snd_usb_audio object to quirk functions ALSA: usb-audio: Simplify snd_usb_init_sample_rate() arguments ALSA: usb-audio: Simplify snd_usb_init_pitch() arguments ALSA: usb-audio: Stop both endpoints properly at error ALSA: usb-audio: Set callbacks via snd_usb_endpoint_set_callback() ALSA: usb-audio: Always set up the parameters after resume ALSA: usb-audio: Fix EP matching for continuous rates ALSA: usb-audio: Refactor endpoint management ALSA: usb-audio: Fix possible stall of implicit fb packet ring-buffer ALSA: usb-audio: Constify audioformat pointer references ALSA: usb-audio: Use atomic_t for endpoint use_count ALSA: usb-audio: Refactoring endpoint URB deactivation ALSA: usb-audio: Drop unneeded snd_usb_substream fields ALSA: usb-audio: Unify the code for the next packet size calculation ALSA: usb-audio: Simplify rate_min/max and rates set up ALSA: usb-audio: Replace slave/master terms ALSA: usb-audio: Use unsigned char for iface and altsettings fields ALSA: usb-audio: Show sync endpoint information in proc outputs ALSA: usb-audio: Quirk for BOSS GT-001 ALSA: usb-audio: Factor out the implicit feedback quirk code ALSA: usb-audio: Add generic implicit fb parsing ALSA: usb-audio: Add implicit_fb module option ALSA: usb-audio: Fix quirks for other BOSS devices Documentation/sound/alsa-configuration.rst | 5 + sound/usb/Makefile | 1 + sound/usb/card.c | 14 +- sound/usb/card.h | 53 +- sound/usb/clock.c | 152 ++-- sound/usb/clock.h | 11 +- sound/usb/debug.h | 16 - sound/usb/endpoint.c | 943 +++++++++++++---------- sound/usb/endpoint.h | 57 +- sound/usb/format.c | 125 +++- sound/usb/helper.c | 10 + sound/usb/helper.h | 3 + sound/usb/implicit.c | 402 ++++++++++ sound/usb/implicit.h | 14 + sound/usb/mixer.c | 46 -- sound/usb/pcm.c | 1117 ++++++++++------------------ sound/usb/pcm.h | 7 +- sound/usb/proc.c | 35 +- sound/usb/quirks-table.h | 8 - sound/usb/quirks.c | 60 +- sound/usb/quirks.h | 10 +- sound/usb/stream.c | 24 +- sound/usb/usbaudio.h | 5 +- 23 files changed, 1696 insertions(+), 1422 deletions(-) delete mode 100644 sound/usb/debug.h create mode 100644 sound/usb/implicit.c create mode 100644 sound/usb/implicit.h -- 2.16.4

3 51

[PATCH v2 0/4] ASoC: rt5645: Enable internal mic and headset on ECS EF20
by Chris Chiu 07 Jan '21

07 Jan '21

These patches are trying to fix the jack detection and internal microphone problems on ECS EF20 series laptops which are empowered by Intel Atom x5-Z8350 CPU (CherryTrail) with Realtek rt5645 audio codec. --- v1 -> v2: Invoke callback() of the DMI quirk if it exists, because the dmi_first_match() doesn't. --- 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 include/sound/rt5645.h | 2 ++ sound/soc/codecs/rt5645.c | 48 ++++++++++++++++++++++++++++++++++++++- 2 files changed, 49 insertions(+), 1 deletion(-) -- 2.20.1

1 4

[PATCH] ALSA: hda/realtek: fix right sounds and mute/micmute LEDs for HP machines
by Jeremy Szu 07 Jan '21

07 Jan '21

* The HP ZBook Fury 15/17 G7 Mobile Workstation are using ALC285 codec which is using 0x04 to control mute LED and 0x01 to control micmute LED. * The right channel speaker is no sound and it needs to expose GPIO1 for initialing AMP. Add quirks to support them. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3c1d2a3fb1a4..dd82ff2bd5d6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7970,6 +7970,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x8780, "HP ZBook Fury 17 G7 Mobile Workstation", + ALC285_FIXUP_HP_GPIO_AMP_INIT), + SND_PCI_QUIRK(0x103c, 0x8783, "HP ZBook Fury 15 G7 Mobile Workstation", + ALC285_FIXUP_HP_GPIO_AMP_INIT), 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), -- 2.29.2

2 1

[PATCH] ucm: fix typo in docs
by Curtis Malainey 07 Jan '21

07 Jan '21

Do you know the tstaus of this fix? Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- alsaucm/alsaucm.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/alsaucm/alsaucm.rst b/alsaucm/alsaucm.rst index 7890ba5..3098672 100644 --- a/alsaucm/alsaucm.rst +++ b/alsaucm/alsaucm.rst @@ -122,7 +122,7 @@ Available commands: the value of the `IDENTIFIER` argument can can be: - ``_devstatus/{device}`` - - ``_modtstaus/{device}`` + - ``_modstatus/{device}`` ``set`` `IDENTIFIER` `VALUE` set string value -- 2.29.2.729.g45daf8777d-goog

2 1

[PATCH v2 1/3] ASoC: SOF: Intel: hda: Resume codec to do jack detection
by Kai-Heng Feng 07 Jan '21

07 Jan '21

Instead of queueing jackpoll_work, runtime resume the codec to let it use different jack detection methods based on jackpoll_interval. This matches SOF driver's behavior with commit a6e7d0a4bdb0 ("ALSA: hda: fix jack detection with Realtek codecs when in D3"). Since SOF only uses Realtek codec, we don't need any additional check for the resume. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v2: No change. sound/soc/sof/intel/hda-codec.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 6875fa570c2c..df59c79cfdfc 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -93,8 +93,7 @@ void hda_codec_jack_check(struct snd_sof_dev *sdev) * has been recorded in STATESTS */ if (codec->jacktbl.used) - schedule_delayed_work(&codec->jackpoll_work, - codec->jackpoll_interval); + pm_request_resume(&codec->core.dev); } #else void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) {} -- 2.29.2

2 7

[PATCH] dt-bindings: Add missing array size constraints
by Rob Herring 07 Jan '21

07 Jan '21

DT properties which can have multiple entries need to specify what the entries are and define how many entries there can be. In the case of only a single entry, just 'maxItems: 1' is sufficient. Add the missing entry constraints. These were found with a modified meta-schema. Unfortunately, there are a few cases where the size constraints are not defined such as common bindings, so the meta-schema can't be part of the normal checks. Cc: Jens Axboe <axboe(a)kernel.dk> Cc: Stephen Boyd <sboyd(a)kernel.org> Cc: Thierry Reding <thierry.reding(a)gmail.com> Cc: MyungJoo Ham <myungjoo.ham(a)samsung.com> Cc: Chanwoo Choi <cw00.choi(a)samsung.com> Cc: Linus Walleij <linus.walleij(a)linaro.org> Cc: Bartosz Golaszewski <bgolaszewski(a)baylibre.com> Cc: Jonathan Cameron <jic23(a)kernel.org> Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: Thomas Gleixner <tglx(a)linutronix.de> Cc: Marc Zyngier <maz(a)kernel.org> Cc: Mauro Carvalho Chehab <mchehab(a)kernel.org> Cc: Chen-Yu Tsai <wens(a)csie.org> Cc: Ulf Hansson <ulf.hansson(a)linaro.org> Cc: "David S. Miller" <davem(a)davemloft.net> Cc: Jakub Kicinski <kuba(a)kernel.org> Cc: Sebastian Reichel <sre(a)kernel.org> Cc: Ohad Ben-Cohen <ohad(a)wizery.com> Cc: Bjorn Andersson <bjorn.andersson(a)linaro.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Rob Herring <robh(a)kernel.org> --- Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-ide(a)vger.kernel.org Cc: linux-clk(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-gpio(a)vger.kernel.org Cc: linux-iio(a)vger.kernel.org Cc: linux-input(a)vger.kernel.org Cc: linux-media(a)vger.kernel.org Cc: linux-mmc(a)vger.kernel.org Cc: netdev(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linux-remoteproc(a)vger.kernel.org Cc: linux-riscv(a)lists.infradead.org Cc: linux-serial(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-spi(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org --- .../socionext,uniphier-system-cache.yaml | 4 ++-- .../bindings/ata/sata_highbank.yaml | 1 + .../bindings/clock/canaan,k210-clk.yaml | 1 + .../bindings/display/brcm,bcm2711-hdmi.yaml | 1 + .../bindings/display/brcm,bcm2835-hdmi.yaml | 1 + .../display/panel/jdi,lt070me05000.yaml | 1 + .../display/panel/mantix,mlaf057we51-x.yaml | 3 ++- .../display/panel/novatek,nt36672a.yaml | 1 + .../devicetree/bindings/dsp/fsl,dsp.yaml | 2 +- .../devicetree/bindings/eeprom/at25.yaml | 3 +-- .../bindings/extcon/extcon-ptn5150.yaml | 2 ++ .../bindings/gpio/gpio-pca95xx.yaml | 1 + .../bindings/iio/adc/adi,ad7768-1.yaml | 2 ++ .../bindings/iio/adc/aspeed,ast2400-adc.yaml | 1 + .../bindings/iio/adc/lltc,ltc2496.yaml | 2 +- .../bindings/iio/adc/qcom,spmi-vadc.yaml | 1 + .../bindings/iio/adc/st,stm32-adc.yaml | 2 ++ .../iio/magnetometer/asahi-kasei,ak8975.yaml | 1 + .../iio/potentiometer/adi,ad5272.yaml | 1 + .../input/touchscreen/elan,elants_i2c.yaml | 1 + .../interrupt-controller/fsl,intmux.yaml | 2 +- .../interrupt-controller/st,stm32-exti.yaml | 2 ++ .../allwinner,sun4i-a10-video-engine.yaml | 1 + .../devicetree/bindings/media/i2c/imx219.yaml | 1 + .../memory-controllers/exynos-srom.yaml | 2 ++ .../bindings/misc/fsl,dpaa2-console.yaml | 1 + .../bindings/mmc/mmc-controller.yaml | 2 ++ .../bindings/net/ti,k3-am654-cpsw-nuss.yaml | 1 + .../bindings/net/ti,k3-am654-cpts.yaml | 1 + .../phy/allwinner,sun4i-a10-usb-phy.yaml | 2 ++ .../phy/allwinner,sun50i-a64-usb-phy.yaml | 2 ++ .../phy/allwinner,sun50i-h6-usb-phy.yaml | 2 ++ .../phy/allwinner,sun5i-a13-usb-phy.yaml | 2 ++ .../phy/allwinner,sun6i-a31-usb-phy.yaml | 2 ++ .../phy/allwinner,sun8i-a23-usb-phy.yaml | 2 ++ .../phy/allwinner,sun8i-a83t-usb-phy.yaml | 2 ++ .../phy/allwinner,sun8i-h3-usb-phy.yaml | 2 ++ .../phy/allwinner,sun8i-r40-usb-phy.yaml | 2 ++ .../phy/allwinner,sun8i-v3s-usb-phy.yaml | 2 ++ .../phy/allwinner,sun9i-a80-usb-phy.yaml | 19 ++++++++----------- .../phy/socionext,uniphier-ahci-phy.yaml | 2 +- .../phy/socionext,uniphier-pcie-phy.yaml | 2 +- .../phy/socionext,uniphier-usb3hs-phy.yaml | 2 +- .../phy/socionext,uniphier-usb3ss-phy.yaml | 2 +- .../bindings/phy/ti,phy-gmii-sel.yaml | 2 +- .../pinctrl/aspeed,ast2400-pinctrl.yaml | 3 +-- .../pinctrl/aspeed,ast2500-pinctrl.yaml | 4 ++-- .../bindings/power/supply/bq25980.yaml | 1 + .../bindings/remoteproc/ingenic,vpu.yaml | 2 +- .../remoteproc/ti,omap-remoteproc.yaml | 3 +++ .../bindings/riscv/sifive-l2-cache.yaml | 1 + .../bindings/serial/renesas,hscif.yaml | 2 ++ .../bindings/serial/renesas,scif.yaml | 2 ++ .../bindings/serial/renesas,scifa.yaml | 2 ++ .../bindings/serial/renesas,scifb.yaml | 2 ++ .../sound/allwinner,sun4i-a10-codec.yaml | 1 + .../bindings/sound/google,sc7180-trogdor.yaml | 1 + .../bindings/sound/samsung,aries-wm8994.yaml | 3 +++ .../bindings/sound/samsung,midas-audio.yaml | 2 ++ .../devicetree/bindings/sound/tas2562.yaml | 2 ++ .../devicetree/bindings/sound/tas2770.yaml | 2 ++ .../bindings/sound/tlv320adcx140.yaml | 1 + .../devicetree/bindings/spi/renesas,rspi.yaml | 2 ++ .../devicetree/bindings/sram/sram.yaml | 2 ++ .../timer/allwinner,sun4i-a10-timer.yaml | 2 ++ .../bindings/timer/intel,ixp4xx-timer.yaml | 2 +- .../usb/allwinner,sun4i-a10-musb.yaml | 2 +- .../bindings/usb/brcm,usb-pinmap.yaml | 3 +++ .../devicetree/bindings/usb/generic-ehci.yaml | 1 + .../devicetree/bindings/usb/generic-ohci.yaml | 1 + .../devicetree/bindings/usb/ingenic,musb.yaml | 2 +- .../bindings/usb/renesas,usbhs.yaml | 1 + .../devicetree/bindings/usb/ti,j721e-usb.yaml | 3 ++- .../bindings/usb/ti,keystone-dwc3.yaml | 2 ++ 74 files changed, 118 insertions(+), 33 deletions(-) diff --git a/Documentation/devicetree/bindings/arm/socionext/socionext,uniphier-system-cache.yaml b/Documentation/devicetree/bindings/arm/socionext/socionext,uniphier-system-cache.yaml index 2e765bb3e6f6..7ca5375f278f 100644 --- a/Documentation/devicetree/bindings/arm/socionext/socionext,uniphier-system-cache.yaml +++ b/Documentation/devicetree/bindings/arm/socionext/socionext,uniphier-system-cache.yaml @@ -30,8 +30,8 @@ properties: Interrupts can be used to notify the completion of cache operations. The number of interrupts should match to the number of CPU cores. The specified interrupts correspond to CPU0, CPU1, ... in this order. - minItems: 1 - maxItems: 4 + minItems: 1 + maxItems: 4 cache-unified: true diff --git a/Documentation/devicetree/bindings/ata/sata_highbank.yaml b/Documentation/devicetree/bindings/ata/sata_highbank.yaml index 5e2a2394e600..ce75d77e9289 100644 --- a/Documentation/devicetree/bindings/ata/sata_highbank.yaml +++ b/Documentation/devicetree/bindings/ata/sata_highbank.yaml @@ -61,6 +61,7 @@ properties: maxItems: 8 calxeda,sgpio-gpio: + maxItems: 3 description: | phandle-gpio bank, bit offset, and default on or off, which indicates that the driver supports SGPIO indicator lights using the indicated diff --git a/Documentation/devicetree/bindings/clock/canaan,k210-clk.yaml b/Documentation/devicetree/bindings/clock/canaan,k210-clk.yaml index 565ca468cb44..7f5cf4001f76 100644 --- a/Documentation/devicetree/bindings/clock/canaan,k210-clk.yaml +++ b/Documentation/devicetree/bindings/clock/canaan,k210-clk.yaml @@ -22,6 +22,7 @@ properties: const: canaan,k210-clk clocks: + maxItems: 1 description: Phandle of the SoC 26MHz fixed-rate oscillator clock. diff --git a/Documentation/devicetree/bindings/display/brcm,bcm2711-hdmi.yaml b/Documentation/devicetree/bindings/display/brcm,bcm2711-hdmi.yaml index 7ce06f9f9f8e..767edc0a7978 100644 --- a/Documentation/devicetree/bindings/display/brcm,bcm2711-hdmi.yaml +++ b/Documentation/devicetree/bindings/display/brcm,bcm2711-hdmi.yaml @@ -60,6 +60,7 @@ properties: Phandle of the I2C controller used for DDC EDID probing hpd-gpios: + maxItems: 1 description: > The GPIO pin for the HDMI hotplug detect (if it doesn't appear as an interrupt/status bit in the HDMI controller itself) diff --git a/Documentation/devicetree/bindings/display/brcm,bcm2835-hdmi.yaml b/Documentation/devicetree/bindings/display/brcm,bcm2835-hdmi.yaml index f54b4e4808f0..031e35e76db2 100644 --- a/Documentation/devicetree/bindings/display/brcm,bcm2835-hdmi.yaml +++ b/Documentation/devicetree/bindings/display/brcm,bcm2835-hdmi.yaml @@ -37,6 +37,7 @@ properties: Phandle of the I2C controller used for DDC EDID probing hpd-gpios: + maxItems: 1 description: > The GPIO pin for the HDMI hotplug detect (if it doesn't appear as an interrupt/status bit in the HDMI controller itself) diff --git a/Documentation/devicetree/bindings/display/panel/jdi,lt070me05000.yaml b/Documentation/devicetree/bindings/display/panel/jdi,lt070me05000.yaml index b8b9435e464c..4f92365e888a 100644 --- a/Documentation/devicetree/bindings/display/panel/jdi,lt070me05000.yaml +++ b/Documentation/devicetree/bindings/display/panel/jdi,lt070me05000.yaml @@ -30,6 +30,7 @@ properties: power supply for LCM (1.8V) dcdc-en-gpios: + maxItems: 1 description: | phandle of the gpio for power ic line Power IC supply enable, High active diff --git a/Documentation/devicetree/bindings/display/panel/mantix,mlaf057we51-x.yaml b/Documentation/devicetree/bindings/display/panel/mantix,mlaf057we51-x.yaml index 51f423297ec8..aa5a0dc391a4 100644 --- a/Documentation/devicetree/bindings/display/panel/mantix,mlaf057we51-x.yaml +++ b/Documentation/devicetree/bindings/display/panel/mantix,mlaf057we51-x.yaml @@ -37,7 +37,8 @@ properties: reset-gpios: true - 'mantix,tp-rstn-gpios': + mantix,tp-rstn-gpios: + maxItems: 1 description: second reset line that triggers DSI config load backlight: true diff --git a/Documentation/devicetree/bindings/display/panel/novatek,nt36672a.yaml b/Documentation/devicetree/bindings/display/panel/novatek,nt36672a.yaml index 2f5df1d235ae..ef4c0a24512d 100644 --- a/Documentation/devicetree/bindings/display/panel/novatek,nt36672a.yaml +++ b/Documentation/devicetree/bindings/display/panel/novatek,nt36672a.yaml @@ -30,6 +30,7 @@ properties: panel. The novatek,nt36672a compatible shall always be provided as a fallback. reset-gpios: + maxItems: 1 description: phandle of gpio for reset line - This should be 8mA, gpio can be configured using mux, pinctrl, pinctrl-names (active high) diff --git a/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml b/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml index 4cc011230153..7afc9f2be13a 100644 --- a/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml +++ b/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml @@ -21,7 +21,7 @@ properties: - fsl,imx8mp-dsp reg: - description: Should contain register location and length + maxItems: 1 clocks: items: diff --git a/Documentation/devicetree/bindings/eeprom/at25.yaml b/Documentation/devicetree/bindings/eeprom/at25.yaml index 121a601db22e..6a2dc8b3ed14 100644 --- a/Documentation/devicetree/bindings/eeprom/at25.yaml +++ b/Documentation/devicetree/bindings/eeprom/at25.yaml @@ -39,8 +39,7 @@ properties: - const: atmel,at25 reg: - description: - Chip select number. + maxItems: 1 spi-max-frequency: true diff --git a/Documentation/devicetree/bindings/extcon/extcon-ptn5150.yaml b/Documentation/devicetree/bindings/extcon/extcon-ptn5150.yaml index 4b0f414486d2..d5cfa32ea52d 100644 --- a/Documentation/devicetree/bindings/extcon/extcon-ptn5150.yaml +++ b/Documentation/devicetree/bindings/extcon/extcon-ptn5150.yaml @@ -19,6 +19,7 @@ properties: const: nxp,ptn5150 int-gpios: + maxItems: 1 deprecated: true description: GPIO pin (input) connected to the PTN5150's INTB pin. @@ -31,6 +32,7 @@ properties: maxItems: 1 vbus-gpios: + maxItems: 1 description: GPIO pin (output) used to control VBUS. If skipped, no such control takes place. diff --git a/Documentation/devicetree/bindings/gpio/gpio-pca95xx.yaml b/Documentation/devicetree/bindings/gpio/gpio-pca95xx.yaml index f5ee23c2df60..57cdcfd4ff3c 100644 --- a/Documentation/devicetree/bindings/gpio/gpio-pca95xx.yaml +++ b/Documentation/devicetree/bindings/gpio/gpio-pca95xx.yaml @@ -81,6 +81,7 @@ properties: const: 2 reset-gpios: + maxItems: 1 description: GPIO specification for the RESET input. This is an active low signal to the PCA953x. Not valid for Maxim MAX732x devices. diff --git a/Documentation/devicetree/bindings/iio/adc/adi,ad7768-1.yaml b/Documentation/devicetree/bindings/iio/adc/adi,ad7768-1.yaml index 924477dfb833..a85a28145ef6 100644 --- a/Documentation/devicetree/bindings/iio/adc/adi,ad7768-1.yaml +++ b/Documentation/devicetree/bindings/iio/adc/adi,ad7768-1.yaml @@ -40,6 +40,7 @@ properties: ADC reference voltage supply adi,sync-in-gpios: + maxItems: 1 description: Enables synchronization of multiple devices that require simultaneous sampling. A pulse is always required if the configuration is changed @@ -76,6 +77,7 @@ patternProperties: properties: reg: + maxItems: 1 description: | The channel number. diff --git a/Documentation/devicetree/bindings/iio/adc/aspeed,ast2400-adc.yaml b/Documentation/devicetree/bindings/iio/adc/aspeed,ast2400-adc.yaml index 7f534a933e92..a726b6c2ab65 100644 --- a/Documentation/devicetree/bindings/iio/adc/aspeed,ast2400-adc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/aspeed,ast2400-adc.yaml @@ -23,6 +23,7 @@ properties: maxItems: 1 clocks: + maxItems: 1 description: Input clock used to derive the sample clock. Expected to be the SoC's APB clock. diff --git a/Documentation/devicetree/bindings/iio/adc/lltc,ltc2496.yaml b/Documentation/devicetree/bindings/iio/adc/lltc,ltc2496.yaml index 2716d4e95329..0bd2fc0356c8 100644 --- a/Documentation/devicetree/bindings/iio/adc/lltc,ltc2496.yaml +++ b/Documentation/devicetree/bindings/iio/adc/lltc,ltc2496.yaml @@ -20,7 +20,7 @@ properties: description: Power supply for the reference voltage reg: - description: spi chipselect number according to the usual spi bindings + maxItems: 1 spi-max-frequency: description: maximal spi bus frequency supported diff --git a/Documentation/devicetree/bindings/iio/adc/qcom,spmi-vadc.yaml b/Documentation/devicetree/bindings/iio/adc/qcom,spmi-vadc.yaml index 95cc705b961b..74a4a9d95798 100644 --- a/Documentation/devicetree/bindings/iio/adc/qcom,spmi-vadc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/qcom,spmi-vadc.yaml @@ -68,6 +68,7 @@ patternProperties: properties: reg: + maxItems: 1 description: | ADC channel number. See include/dt-bindings/iio/qcom,spmi-vadc.h diff --git a/Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml b/Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml index 28417b31b558..6364ede9bb5f 100644 --- a/Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml @@ -41,6 +41,8 @@ properties: maxItems: 2 clocks: + minItems: 1 + maxItems: 2 description: | Core can use up to two clocks, depending on part used: - "adc" clock: for the analog circuitry, common to all ADCs. diff --git a/Documentation/devicetree/bindings/iio/magnetometer/asahi-kasei,ak8975.yaml b/Documentation/devicetree/bindings/iio/magnetometer/asahi-kasei,ak8975.yaml index a25590a16ba7..a0a1ffe017df 100644 --- a/Documentation/devicetree/bindings/iio/magnetometer/asahi-kasei,ak8975.yaml +++ b/Documentation/devicetree/bindings/iio/magnetometer/asahi-kasei,ak8975.yaml @@ -47,6 +47,7 @@ properties: description: an optional 3x3 mounting rotation matrix. reset-gpios: + maxItems: 1 description: | an optional pin needed for AK09911 to set the reset state. This should be usually active low diff --git a/Documentation/devicetree/bindings/iio/potentiometer/adi,ad5272.yaml b/Documentation/devicetree/bindings/iio/potentiometer/adi,ad5272.yaml index 1aee9f9be951..0ebb6725a1af 100644 --- a/Documentation/devicetree/bindings/iio/potentiometer/adi,ad5272.yaml +++ b/Documentation/devicetree/bindings/iio/potentiometer/adi,ad5272.yaml @@ -25,6 +25,7 @@ properties: maxItems: 1 reset-gpios: + maxItems: 1 description: Active low signal to the AD5272 RESET input. diff --git a/Documentation/devicetree/bindings/input/touchscreen/elan,elants_i2c.yaml b/Documentation/devicetree/bindings/input/touchscreen/elan,elants_i2c.yaml index a792d6377b1d..a9b53c2e6f0a 100644 --- a/Documentation/devicetree/bindings/input/touchscreen/elan,elants_i2c.yaml +++ b/Documentation/devicetree/bindings/input/touchscreen/elan,elants_i2c.yaml @@ -29,6 +29,7 @@ properties: description: touchscreen can be used as a wakeup source. reset-gpios: + maxItems: 1 description: reset gpio the chip is connected to. vcc33-supply: diff --git a/Documentation/devicetree/bindings/interrupt-controller/fsl,intmux.yaml b/Documentation/devicetree/bindings/interrupt-controller/fsl,intmux.yaml index 43c6effbb5bd..1d6e0f64a807 100644 --- a/Documentation/devicetree/bindings/interrupt-controller/fsl,intmux.yaml +++ b/Documentation/devicetree/bindings/interrupt-controller/fsl,intmux.yaml @@ -31,7 +31,7 @@ properties: The 1st cell is hw interrupt number, the 2nd cell is channel index. clocks: - description: ipg clock. + maxItems: 1 clock-names: const: ipg diff --git a/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml b/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml index 2a5b29567926..6d3e68eb2e8b 100644 --- a/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml +++ b/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml @@ -36,6 +36,8 @@ properties: Reference to a phandle of a hardware spinlock provider node. interrupts: + minItems: 1 + maxItems: 96 description: Interrupts references to primary interrupt controller diff --git a/Documentation/devicetree/bindings/media/allwinner,sun4i-a10-video-engine.yaml b/Documentation/devicetree/bindings/media/allwinner,sun4i-a10-video-engine.yaml index 2f7058f7760c..c34303b87a5b 100644 --- a/Documentation/devicetree/bindings/media/allwinner,sun4i-a10-video-engine.yaml +++ b/Documentation/devicetree/bindings/media/allwinner,sun4i-a10-video-engine.yaml @@ -53,6 +53,7 @@ properties: maxItems: 1 memory-region: + maxItems: 1 description: CMA pool to use for buffers allocation instead of the default CMA pool. diff --git a/Documentation/devicetree/bindings/media/i2c/imx219.yaml b/Documentation/devicetree/bindings/media/i2c/imx219.yaml index dfc4d29a4f04..184d33bd3828 100644 --- a/Documentation/devicetree/bindings/media/i2c/imx219.yaml +++ b/Documentation/devicetree/bindings/media/i2c/imx219.yaml @@ -40,6 +40,7 @@ properties: Digital core voltage supply, 1.2 volts reset-gpios: + maxItems: 1 description: |- Reference to the GPIO connected to the xclr pin, if any. Must be released (set high) after all supplies are applied. diff --git a/Documentation/devicetree/bindings/memory-controllers/exynos-srom.yaml b/Documentation/devicetree/bindings/memory-controllers/exynos-srom.yaml index 637e24f0f73b..c6e44f47ce7c 100644 --- a/Documentation/devicetree/bindings/memory-controllers/exynos-srom.yaml +++ b/Documentation/devicetree/bindings/memory-controllers/exynos-srom.yaml @@ -28,6 +28,8 @@ properties: const: 1 ranges: + minItems: 1 + maxItems: 4 description: | Reflects the memory layout with four integer values per bank. Format: <bank-number> 0 <parent address of bank> <size> diff --git a/Documentation/devicetree/bindings/misc/fsl,dpaa2-console.yaml b/Documentation/devicetree/bindings/misc/fsl,dpaa2-console.yaml index 271a3eafc054..8cc951feb7df 100644 --- a/Documentation/devicetree/bindings/misc/fsl,dpaa2-console.yaml +++ b/Documentation/devicetree/bindings/misc/fsl,dpaa2-console.yaml @@ -15,6 +15,7 @@ properties: const: "fsl,dpaa2-console" reg: + maxItems: 1 description: A standard property. Specifies the region where the MCFBA (MC firmware base address) register can be found. diff --git a/Documentation/devicetree/bindings/mmc/mmc-controller.yaml b/Documentation/devicetree/bindings/mmc/mmc-controller.yaml index 186f04ba9357..df4ee4c778ae 100644 --- a/Documentation/devicetree/bindings/mmc/mmc-controller.yaml +++ b/Documentation/devicetree/bindings/mmc/mmc-controller.yaml @@ -40,6 +40,7 @@ properties: There is no card detection available; polling must be used. cd-gpios: + maxItems: 1 description: The card detection will be done using the GPIO provided. @@ -104,6 +105,7 @@ properties: line. Not used in combination with eMMC or SDIO. wp-gpios: + maxItems: 1 description: GPIO to use for the write-protect detection. diff --git a/Documentation/devicetree/bindings/net/ti,k3-am654-cpsw-nuss.yaml b/Documentation/devicetree/bindings/net/ti,k3-am654-cpsw-nuss.yaml index c47b58f3e3f6..097c5cc6c853 100644 --- a/Documentation/devicetree/bindings/net/ti,k3-am654-cpsw-nuss.yaml +++ b/Documentation/devicetree/bindings/net/ti,k3-am654-cpsw-nuss.yaml @@ -66,6 +66,7 @@ properties: dma-coherent: true clocks: + maxItems: 1 description: CPSW2G NUSS functional clock clock-names: diff --git a/Documentation/devicetree/bindings/net/ti,k3-am654-cpts.yaml b/Documentation/devicetree/bindings/net/ti,k3-am654-cpts.yaml index 9b7117920d90..2a42a27fb911 100644 --- a/Documentation/devicetree/bindings/net/ti,k3-am654-cpts.yaml +++ b/Documentation/devicetree/bindings/net/ti,k3-am654-cpts.yaml @@ -59,6 +59,7 @@ properties: - const: cpts clocks: + maxItems: 1 description: CPTS reference clock clock-names: diff --git a/Documentation/devicetree/bindings/phy/allwinner,sun4i-a10-usb-phy.yaml b/Documentation/devicetree/bindings/phy/allwinner,sun4i-a10-usb-phy.yaml index 94ac23687b7e..77606c899fe2 100644 --- a/Documentation/devicetree/bindings/phy/allwinner,sun4i-a10-usb-phy.yaml +++ b/Documentation/devicetree/bindings/phy/allwinner,sun4i-a10-usb-phy.yaml @@ -51,9 +51,11 @@ properties: - const: usb2_reset usb0_id_det-gpios: + maxItems: 1 description: GPIO to the USB OTG ID pin usb0_vbus_det-gpios: + maxItems: 1 description: GPIO to the USB OTG VBUS detect pin usb0_vbus_power-supply: diff --git a/Documentation/devicetree/bindings/phy/allwinner,sun50i-a64-usb-phy.yaml b/Documentation/devicetree/bindings/phy/allwinner,sun50i-a64-usb-phy.yaml index fd6e126fcf18..078af52b16ed 100644 --- a/Documentation/devicetree/bindings/phy/allwinner,sun50i-a64-usb-phy.yaml +++ b/Documentation/devicetree/bindings/phy/allwinner,sun50i-a64-usb-phy.yaml @@ -50,9 +50,11 @@ properties: - const: usb1_reset usb0_id_det-gpios: + maxItems: 1 description: GPIO to the USB OTG ID pin usb0_vbus_det-gpios: + maxItems: 1 description: GPIO to the USB OTG VBUS detect pin usb0_vbus_power-supply: diff --git a/Documentation/devicetree/bindings/phy/allwinner,sun50i-h6-usb-phy.yaml b/Documentation/devicetree/bindings/phy/allwinner,sun50i-h6-usb-phy.yaml index 7670411002c9..e632140722a2 100644 --- a/Documentation/devicetree/bindings/phy/allwinner,sun50i-h6-usb-phy.yaml +++ b/Documentation/devicetree/bindings/phy/allwinner,sun50i-h6-usb-phy.yaml @@ -50,9 +50,11 @@ properties: - const: usb3_reset usb0_id_det-gpios: + maxItems: 1 description: GPIO to the USB OTG ID pin usb0_vbus_det-gpios: + maxItems: 1 description: GPIO to the USB OTG VBUS detect pin usb0_vbus_power-supply: diff --git a/Documentation/devicetree/bindings/phy/allwinner,sun5i-a13-usb-phy.yaml b/Documentation/devicetree/bindings/phy/allwinner,sun5i-a13-usb-phy.yaml index 9b319381d1ad..5bad9b06e2e7 100644 --- a/Documentation/devicetree/bindings/phy/allwinner,sun5i-a13-usb-phy.yaml +++ b/Documentation/devicetree/bindings/phy/allwinner,sun5i-a13-usb-phy.yaml @@ -45,9 +45,11 @@ properties: - const: usb1_reset usb0_id_det-gpios: + maxItems: 1 description: GPIO to the USB OTG ID pin usb0_vbus_det-gpios: + maxItems: 1 description: GPIO to the USB OTG VBUS detect pin usb0_vbus_power-supply: diff --git a/Documentation/devicetree/bindings/phy/allwinner,sun6i-a31-usb-phy.yaml b/Documentation/devicetree/bindings/phy/allwinner,sun6i-a31-usb-phy.yaml index b0ed01bbf3db..922b4665e00d 100644 --- a/Documentation/devicetree/bindings/phy/allwinner,sun6i-a31-usb-phy.yaml +++ b/Documentation/devicetree/bindings/phy/allwinner,sun6i-a31-usb-phy.yaml @@ -54,9 +54,11 @@ properties: - const: usb2_reset usb0_id_det-gpios: + maxItems: 1 description: GPIO to the USB OTG ID pin usb0_vbus_det-gpios: + maxItems: 1 description: GPIO to the USB OTG VBUS detect pin usb0_vbus_power-supply: diff --git a/Documentation/devicetree/bindings/phy/allwinner,sun8i-a23-usb-phy.yaml b/Documentation/devicetree/bindings/phy/allwinner,sun8i-a23-usb-phy.yaml index b0674406f8aa..a94019efc2f3 100644 --- a/Documentation/devicetree/bindings/phy/allwinner,sun8i-a23-usb-phy.yaml +++ b/Documentation/devicetree/bindings/phy/allwinner,sun8i-a23-usb-phy.yaml @@ -50,9 +50,11 @@ properties: - const: usb1_reset usb0_id_det-gpios: + maxItems: 1 description: GPIO to the USB OTG ID pin usb0_vbus_det-gpios: + maxItems: 1 description: GPIO to the USB OTG VBUS detect pin usb0_vbus_power-supply: diff --git a/Documentation/devicetree/bindings/phy/allwinner,sun8i-a83t-usb-phy.yaml b/Documentation/devicetree/bindings/phy/allwinner,sun8i-a83t-usb-phy.yaml index 48dc9c834a9b..33f3ddc0492d 100644 --- a/Documentation/devicetree/bindings/phy/allwinner,sun8i-a83t-usb-phy.yaml +++ b/Documentation/devicetree/bindings/phy/allwinner,sun8i-a83t-usb-phy.yaml @@ -56,9 +56,11 @@ properties: - const: usb2_reset usb0_id_det-gpios: + maxItems: 1 description: GPIO to the USB OTG ID pin usb0_vbus_det-gpios: + maxItems: 1 description: GPIO to the USB OTG VBUS detect pin usb0_vbus_power-supply: diff --git a/Documentation/devicetree/bindings/phy/allwinner,sun8i-h3-usb-phy.yaml b/Documentation/devicetree/bindings/phy/allwinner,sun8i-h3-usb-phy.yaml index 60c344585276..f80431060803 100644 --- a/Documentation/devicetree/bindings/phy/allwinner,sun8i-h3-usb-phy.yaml +++ b/Documentation/devicetree/bindings/phy/allwinner,sun8i-h3-usb-phy.yaml @@ -62,9 +62,11 @@ properties: - const: usb3_reset usb0_id_det-gpios: + maxItems: 1 description: GPIO to the USB OTG ID pin usb0_vbus_det-gpios: + maxItems: 1 description: GPIO to the USB OTG VBUS detect pin usb0_vbus_power-supply: diff --git a/Documentation/devicetree/bindings/phy/allwinner,sun8i-r40-usb-phy.yaml b/Documentation/devicetree/bindings/phy/allwinner,sun8i-r40-usb-phy.yaml index a2bb36790fbd..d947e50a49d2 100644 --- a/Documentation/devicetree/bindings/phy/allwinner,sun8i-r40-usb-phy.yaml +++ b/Documentation/devicetree/bindings/phy/allwinner,sun8i-r40-usb-phy.yaml @@ -56,9 +56,11 @@ properties: - const: usb2_reset usb0_id_det-gpios: + maxItems: 1 description: GPIO to the USB OTG ID pin usb0_vbus_det-gpios: + maxItems: 1 description: GPIO to the USB OTG VBUS detect pin usb0_vbus_power-supply: diff --git a/Documentation/devicetree/bindings/phy/allwinner,sun8i-v3s-usb-phy.yaml b/Documentation/devicetree/bindings/phy/allwinner,sun8i-v3s-usb-phy.yaml index eadfd0c9493c..a2836c296cc4 100644 --- a/Documentation/devicetree/bindings/phy/allwinner,sun8i-v3s-usb-phy.yaml +++ b/Documentation/devicetree/bindings/phy/allwinner,sun8i-v3s-usb-phy.yaml @@ -42,9 +42,11 @@ properties: const: usb0_reset usb0_id_det-gpios: + maxItems: 1 description: GPIO to the USB OTG ID pin usb0_vbus_det-gpios: + maxItems: 1 description: GPIO to the USB OTG VBUS detect pin usb0_vbus_power-supply: diff --git a/Documentation/devicetree/bindings/phy/allwinner,sun9i-a80-usb-phy.yaml b/Documentation/devicetree/bindings/phy/allwinner,sun9i-a80-usb-phy.yaml index ded7d6f0a119..2eb493fa64fd 100644 --- a/Documentation/devicetree/bindings/phy/allwinner,sun9i-a80-usb-phy.yaml +++ b/Documentation/devicetree/bindings/phy/allwinner,sun9i-a80-usb-phy.yaml @@ -22,7 +22,8 @@ properties: clocks: anyOf: - - description: Main PHY Clock + - maxItems: 1 + description: Main PHY Clock - items: - description: Main PHY clock @@ -39,20 +40,16 @@ properties: - const: hsic_480M resets: - anyOf: + minItems: 1 + items: - description: Normal USB PHY reset - - - items: - - description: Normal USB PHY reset - - description: HSIC Reset + - description: HSIC Reset reset-names: - oneOf: + minItems: 1 + items: - const: phy - - - items: - - const: phy - - const: hsic + - const: hsic phy_type: const: hsic diff --git a/Documentation/devicetree/bindings/phy/socionext,uniphier-ahci-phy.yaml b/Documentation/devicetree/bindings/phy/socionext,uniphier-ahci-phy.yaml index 34756347a14e..745c525ce6b9 100644 --- a/Documentation/devicetree/bindings/phy/socionext,uniphier-ahci-phy.yaml +++ b/Documentation/devicetree/bindings/phy/socionext,uniphier-ahci-phy.yaml @@ -20,7 +20,7 @@ properties: - socionext,uniphier-pxs3-ahci-phy reg: - description: PHY register region (offset and length) + maxItems: 1 "#phy-cells": const: 0 diff --git a/Documentation/devicetree/bindings/phy/socionext,uniphier-pcie-phy.yaml b/Documentation/devicetree/bindings/phy/socionext,uniphier-pcie-phy.yaml index a06831fd64b9..3e0566899041 100644 --- a/Documentation/devicetree/bindings/phy/socionext,uniphier-pcie-phy.yaml +++ b/Documentation/devicetree/bindings/phy/socionext,uniphier-pcie-phy.yaml @@ -21,7 +21,7 @@ properties: - socionext,uniphier-pxs3-pcie-phy reg: - description: PHY register region (offset and length) + maxItems: 1 "#phy-cells": const: 0 diff --git a/Documentation/devicetree/bindings/phy/socionext,uniphier-usb3hs-phy.yaml b/Documentation/devicetree/bindings/phy/socionext,uniphier-usb3hs-phy.yaml index 6fa5caab1487..a681cbc3b4ef 100644 --- a/Documentation/devicetree/bindings/phy/socionext,uniphier-usb3hs-phy.yaml +++ b/Documentation/devicetree/bindings/phy/socionext,uniphier-usb3hs-phy.yaml @@ -24,7 +24,7 @@ properties: - socionext,uniphier-pxs3-usb3-hsphy reg: - description: PHY register region (offset and length) + maxItems: 1 "#phy-cells": const: 0 diff --git a/Documentation/devicetree/bindings/phy/socionext,uniphier-usb3ss-phy.yaml b/Documentation/devicetree/bindings/phy/socionext,uniphier-usb3ss-phy.yaml index 9d46715ed036..41c0dd68ee25 100644 --- a/Documentation/devicetree/bindings/phy/socionext,uniphier-usb3ss-phy.yaml +++ b/Documentation/devicetree/bindings/phy/socionext,uniphier-usb3ss-phy.yaml @@ -25,7 +25,7 @@ properties: - socionext,uniphier-pxs3-usb3-ssphy reg: - description: PHY register region (offset and length) + maxItems: 1 "#phy-cells": const: 0 diff --git a/Documentation/devicetree/bindings/phy/ti,phy-gmii-sel.yaml b/Documentation/devicetree/bindings/phy/ti,phy-gmii-sel.yaml index bcec422d7734..ff8a6d9eb153 100644 --- a/Documentation/devicetree/bindings/phy/ti,phy-gmii-sel.yaml +++ b/Documentation/devicetree/bindings/phy/ti,phy-gmii-sel.yaml @@ -55,7 +55,7 @@ properties: - ti,am654-phy-gmii-sel reg: - description: Address and length of the register set for the device + maxItems: 1 '#phy-cells': true diff --git a/Documentation/devicetree/bindings/pinctrl/aspeed,ast2400-pinctrl.yaml b/Documentation/devicetree/bindings/pinctrl/aspeed,ast2400-pinctrl.yaml index 54631dc1adb0..91be5720d094 100644 --- a/Documentation/devicetree/bindings/pinctrl/aspeed,ast2400-pinctrl.yaml +++ b/Documentation/devicetree/bindings/pinctrl/aspeed,ast2400-pinctrl.yaml @@ -23,8 +23,7 @@ properties: compatible: const: aspeed,ast2400-pinctrl reg: - description: | - A hint for the memory regions associated with the pin-controller + maxItems: 2 patternProperties: '^.*$': diff --git a/Documentation/devicetree/bindings/pinctrl/aspeed,ast2500-pinctrl.yaml b/Documentation/devicetree/bindings/pinctrl/aspeed,ast2500-pinctrl.yaml index a90c0fe0495f..40e9e8d4be5a 100644 --- a/Documentation/devicetree/bindings/pinctrl/aspeed,ast2500-pinctrl.yaml +++ b/Documentation/devicetree/bindings/pinctrl/aspeed,ast2500-pinctrl.yaml @@ -24,8 +24,8 @@ properties: compatible: const: aspeed,ast2500-pinctrl reg: - description: | - A hint for the memory regions associated with the pin-controller + maxItems: 2 + aspeed,external-nodes: minItems: 2 maxItems: 2 diff --git a/Documentation/devicetree/bindings/power/supply/bq25980.yaml b/Documentation/devicetree/bindings/power/supply/bq25980.yaml index f6b3dd4093ca..06eca6667f67 100644 --- a/Documentation/devicetree/bindings/power/supply/bq25980.yaml +++ b/Documentation/devicetree/bindings/power/supply/bq25980.yaml @@ -70,6 +70,7 @@ properties: description: Enables bypass mode at boot time interrupts: + maxItems: 1 description: | Indicates that the device state has changed. diff --git a/Documentation/devicetree/bindings/remoteproc/ingenic,vpu.yaml b/Documentation/devicetree/bindings/remoteproc/ingenic,vpu.yaml index c019f9fbe916..d0aa91bbf5e5 100644 --- a/Documentation/devicetree/bindings/remoteproc/ingenic,vpu.yaml +++ b/Documentation/devicetree/bindings/remoteproc/ingenic,vpu.yaml @@ -44,7 +44,7 @@ properties: - const: vpu interrupts: - description: VPU hardware interrupt + maxItems: 1 required: - compatible diff --git a/Documentation/devicetree/bindings/remoteproc/ti,omap-remoteproc.yaml b/Documentation/devicetree/bindings/remoteproc/ti,omap-remoteproc.yaml index 084960a8f17a..1a1159097a2a 100644 --- a/Documentation/devicetree/bindings/remoteproc/ti,omap-remoteproc.yaml +++ b/Documentation/devicetree/bindings/remoteproc/ti,omap-remoteproc.yaml @@ -70,10 +70,13 @@ properties: the firmware image. clocks: + maxItems: 1 description: | Main functional clock for the remote processor resets: + minItems: 1 + maxItems: 2 description: | Reset handles for the remote processor diff --git a/Documentation/devicetree/bindings/riscv/sifive-l2-cache.yaml b/Documentation/devicetree/bindings/riscv/sifive-l2-cache.yaml index efc0198eeb74..2ece8630dc68 100644 --- a/Documentation/devicetree/bindings/riscv/sifive-l2-cache.yaml +++ b/Documentation/devicetree/bindings/riscv/sifive-l2-cache.yaml @@ -63,6 +63,7 @@ properties: next-level-cache: true memory-region: + maxItems: 1 description: | The reference to the reserved-memory for the L2 Loosely Integrated Memory region. The reserved memory node should be defined as per the bindings in reserved-memory.txt. diff --git a/Documentation/devicetree/bindings/serial/renesas,hscif.yaml b/Documentation/devicetree/bindings/serial/renesas,hscif.yaml index c139c5edb93e..69533347aa57 100644 --- a/Documentation/devicetree/bindings/serial/renesas,hscif.yaml +++ b/Documentation/devicetree/bindings/serial/renesas,hscif.yaml @@ -81,6 +81,8 @@ properties: maxItems: 1 dmas: + minItems: 2 + maxItems: 4 description: Must contain a list of pairs of references to DMA specifiers, one for transmission, and one for reception. diff --git a/Documentation/devicetree/bindings/serial/renesas,scif.yaml b/Documentation/devicetree/bindings/serial/renesas,scif.yaml index 672158906c33..22d76829f7ae 100644 --- a/Documentation/devicetree/bindings/serial/renesas,scif.yaml +++ b/Documentation/devicetree/bindings/serial/renesas,scif.yaml @@ -120,6 +120,8 @@ properties: maxItems: 1 dmas: + minItems: 2 + maxItems: 4 description: Must contain a list of pairs of references to DMA specifiers, one for transmission, and one for reception. diff --git a/Documentation/devicetree/bindings/serial/renesas,scifa.yaml b/Documentation/devicetree/bindings/serial/renesas,scifa.yaml index dbffb9534835..3c67d3202e1b 100644 --- a/Documentation/devicetree/bindings/serial/renesas,scifa.yaml +++ b/Documentation/devicetree/bindings/serial/renesas,scifa.yaml @@ -55,6 +55,8 @@ properties: maxItems: 1 dmas: + minItems: 2 + maxItems: 4 description: Must contain a list of pairs of references to DMA specifiers, one for transmission, and one for reception. diff --git a/Documentation/devicetree/bindings/serial/renesas,scifb.yaml b/Documentation/devicetree/bindings/serial/renesas,scifb.yaml index 147f8a37e02a..d5571c7a4424 100644 --- a/Documentation/devicetree/bindings/serial/renesas,scifb.yaml +++ b/Documentation/devicetree/bindings/serial/renesas,scifb.yaml @@ -55,6 +55,8 @@ properties: maxItems: 1 dmas: + minItems: 2 + maxItems: 4 description: Must contain a list of pairs of references to DMA specifiers, one for transmission, and one for reception. diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml index dd47fef9854d..559aff13ae23 100644 --- a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml @@ -88,6 +88,7 @@ properties: description: Phandle to the codec analog controls in the PRCM allwinner,pa-gpios: + maxItems: 1 description: GPIO to enable the external amplifier required: diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml index 5095b780e2c7..837e3faa63a9 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -55,6 +55,7 @@ patternProperties: maxItems: 1 reg: + maxItems: 1 description: dai link address. cpu: diff --git a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml index 1c6947294825..5fff586dc802 100644 --- a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml @@ -62,12 +62,15 @@ properties: description: Supply for the micbias on the headset mic earpath-sel-gpios: + maxItems: 1 description: GPIO for switching between tv-out and mic paths headset-detect-gpios: + maxItems: 1 description: GPIO for detection of headset insertion headset-key-gpios: + maxItems: 1 description: GPIO for detection of headset key press io-channels: diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml index 578928e67e5c..095775c598fa 100644 --- a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -53,9 +53,11 @@ properties: description: Supply for the micbias on the Sub microphone fm-sel-gpios: + maxItems: 1 description: GPIO pin for FM selection lineout-sel-gpios: + maxItems: 1 description: GPIO pin for line out selection required: diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml index 27f7132ba2ef..acd4bbe69731 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.yaml +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -36,10 +36,12 @@ properties: I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f shut-down-gpios: + maxItems: 1 description: GPIO used to control the state of the device. deprecated: true shutdown-gpios: + maxItems: 1 description: GPIO used to control the state of the device. interrupts: diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml index 07e7f9951d2e..027bebf4e8cf 100644 --- a/Documentation/devicetree/bindings/sound/tas2770.yaml +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -27,9 +27,11 @@ properties: I2C address of the device can be between 0x41 to 0x48. reset-gpio: + maxItems: 1 description: GPIO used to reset the device. shutdown-gpios: + maxItems: 1 description: GPIO used to control the state of the device. interrupts: diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index df18be9d7b15..54d64785aad2 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -35,6 +35,7 @@ properties: I2C addresss of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f reset-gpios: + maxItems: 1 description: | GPIO used for hardware reset. diff --git a/Documentation/devicetree/bindings/spi/renesas,rspi.yaml b/Documentation/devicetree/bindings/spi/renesas,rspi.yaml index 10e83cb17e8d..8397f60d80a2 100644 --- a/Documentation/devicetree/bindings/spi/renesas,rspi.yaml +++ b/Documentation/devicetree/bindings/spi/renesas,rspi.yaml @@ -68,6 +68,8 @@ properties: maxItems: 1 dmas: + minItems: 2 + maxItems: 4 description: Must contain a list of pairs of references to DMA specifiers, one for transmission, and one for reception. diff --git a/Documentation/devicetree/bindings/sram/sram.yaml b/Documentation/devicetree/bindings/sram/sram.yaml index 19d116ff9ddc..2a62bb204bbe 100644 --- a/Documentation/devicetree/bindings/sram/sram.yaml +++ b/Documentation/devicetree/bindings/sram/sram.yaml @@ -35,6 +35,7 @@ properties: maxItems: 1 clocks: + maxItems: 1 description: A list of phandle and clock specifier pair that controls the single SRAM clock. @@ -46,6 +47,7 @@ properties: const: 1 ranges: + maxItems: 1 description: Should translate from local addresses within the sram to bus addresses. diff --git a/Documentation/devicetree/bindings/timer/allwinner,sun4i-a10-timer.yaml b/Documentation/devicetree/bindings/timer/allwinner,sun4i-a10-timer.yaml index d918cee100ac..1c7cf32e7ac2 100644 --- a/Documentation/devicetree/bindings/timer/allwinner,sun4i-a10-timer.yaml +++ b/Documentation/devicetree/bindings/timer/allwinner,sun4i-a10-timer.yaml @@ -22,6 +22,8 @@ properties: maxItems: 1 interrupts: + minItems: 2 + maxItems: 6 description: List of timers interrupts diff --git a/Documentation/devicetree/bindings/timer/intel,ixp4xx-timer.yaml b/Documentation/devicetree/bindings/timer/intel,ixp4xx-timer.yaml index 1a721d8af67a..a8de99b0c0f9 100644 --- a/Documentation/devicetree/bindings/timer/intel,ixp4xx-timer.yaml +++ b/Documentation/devicetree/bindings/timer/intel,ixp4xx-timer.yaml @@ -18,7 +18,7 @@ properties: - const: intel,ixp4xx-timer reg: - description: Should contain registers location and length + maxItems: 1 interrupts: minItems: 1 diff --git a/Documentation/devicetree/bindings/usb/allwinner,sun4i-a10-musb.yaml b/Documentation/devicetree/bindings/usb/allwinner,sun4i-a10-musb.yaml index d9207bf9d894..0f520f17735e 100644 --- a/Documentation/devicetree/bindings/usb/allwinner,sun4i-a10-musb.yaml +++ b/Documentation/devicetree/bindings/usb/allwinner,sun4i-a10-musb.yaml @@ -39,7 +39,7 @@ properties: maxItems: 1 phys: - description: PHY specifier for the OTG PHY + maxItems: 1 phy-names: const: usb diff --git a/Documentation/devicetree/bindings/usb/brcm,usb-pinmap.yaml b/Documentation/devicetree/bindings/usb/brcm,usb-pinmap.yaml index ffa148b9eaa8..d4618d15ecc1 100644 --- a/Documentation/devicetree/bindings/usb/brcm,usb-pinmap.yaml +++ b/Documentation/devicetree/bindings/usb/brcm,usb-pinmap.yaml @@ -22,6 +22,8 @@ properties: description: Interrupt for signals mirrored to out-gpios. in-gpios: + minItems: 1 + maxItems: 2 description: Array of one or two GPIO pins used for input signals. brcm,in-functions: @@ -33,6 +35,7 @@ properties: description: Array of enable and mask pairs, one per gpio in-gpios. out-gpios: + maxItems: 1 description: Array of one GPIO pin used for output signals. brcm,out-functions: diff --git a/Documentation/devicetree/bindings/usb/generic-ehci.yaml b/Documentation/devicetree/bindings/usb/generic-ehci.yaml index 247ef00381ea..f76b25f7fc7a 100644 --- a/Documentation/devicetree/bindings/usb/generic-ehci.yaml +++ b/Documentation/devicetree/bindings/usb/generic-ehci.yaml @@ -83,6 +83,7 @@ properties: Phandle of a companion. phys: + maxItems: 1 description: PHY specifier for the USB PHY phy-names: diff --git a/Documentation/devicetree/bindings/usb/generic-ohci.yaml b/Documentation/devicetree/bindings/usb/generic-ohci.yaml index 2178bcc401bc..8e2bd61f2075 100644 --- a/Documentation/devicetree/bindings/usb/generic-ohci.yaml +++ b/Documentation/devicetree/bindings/usb/generic-ohci.yaml @@ -71,6 +71,7 @@ properties: Overrides the detected port count phys: + maxItems: 1 description: PHY specifier for the USB PHY phy-names: diff --git a/Documentation/devicetree/bindings/usb/ingenic,musb.yaml b/Documentation/devicetree/bindings/usb/ingenic,musb.yaml index 678396eeeb78..f506225a4d57 100644 --- a/Documentation/devicetree/bindings/usb/ingenic,musb.yaml +++ b/Documentation/devicetree/bindings/usb/ingenic,musb.yaml @@ -40,7 +40,7 @@ properties: - const: mc phys: - description: PHY specifier for the USB PHY + maxItems: 1 usb-role-switch: type: boolean diff --git a/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml b/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml index 54c361d4a7af..e67223d90bb7 100644 --- a/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml +++ b/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml @@ -68,6 +68,7 @@ properties: Integer to use BUSWAIT register. renesas,enable-gpio: + maxItems: 1 description: | gpio specifier to check GPIO determining if USB function should be enabled. diff --git a/Documentation/devicetree/bindings/usb/ti,j721e-usb.yaml b/Documentation/devicetree/bindings/usb/ti,j721e-usb.yaml index 388245b91a55..adce36e48bc9 100644 --- a/Documentation/devicetree/bindings/usb/ti,j721e-usb.yaml +++ b/Documentation/devicetree/bindings/usb/ti,j721e-usb.yaml @@ -15,13 +15,14 @@ properties: - const: ti,j721e-usb reg: - description: module registers + maxItems: 1 power-domains: description: PM domain provider node and an args specifier containing the USB device id value. See, Documentation/devicetree/bindings/soc/ti/sci-pm-domain.txt + maxItems: 1 clocks: description: Clock phandles to usb2_refclk and lpm_clk diff --git a/Documentation/devicetree/bindings/usb/ti,keystone-dwc3.yaml b/Documentation/devicetree/bindings/usb/ti,keystone-dwc3.yaml index c1b19fc5d0a2..91ef374faba8 100644 --- a/Documentation/devicetree/bindings/usb/ti,keystone-dwc3.yaml +++ b/Documentation/devicetree/bindings/usb/ti,keystone-dwc3.yaml @@ -43,12 +43,14 @@ properties: maxItems: 2 power-domains: + maxItems: 1 description: Should contain a phandle to a PM domain provider node and an args specifier containing the USB device id value. This property is as per the binding, Documentation/devicetree/bindings/soc/ti/sci-pm-domain.txt phys: + maxItems: 1 description: PHY specifier for the USB3.0 PHY. Some SoCs need the USB3.0 PHY to be turned on before the controller. -- 2.27.0

2 1

Re: [PATCH 00/10] Remove support for TX49xx
by Geert Uytterhoeven 07 Jan '21

07 Jan '21

Hi Nemoto-san, On Thu, Jan 7, 2021 at 2:18 AM Atsushi Nemoto <anemo(a)mba.ocn.ne.jp> wrote: > On Wed, 6 Jan 2021 21:41:24 +0100, Geert Uytterhoeven <geert(a)linux-m68k.org> wrote: > >> > Is that sufficient to keep it? > >> > >> for me it is. But now we probaly need some reverts then... > > > > Indeed. Fortunately not all of it, as some removals were TX4938-only. > > These patches should not break RBTX4927: > > net: tc35815: Drop support for TX49XX boards > spi: txx9: Remove driver > mtd: Remove drivers used by TX49xx > char: hw_random: Remove tx4939 driver > rtc: tx4939: Remove driver > ide: tx4938ide: Remove driver Indeed. > And these patches just break audio-support only. > > dma: tx49 removal > ASoC: txx9: Remove driver > > I think dma and ASoC drivers are hard to maintain now, and can be > dropped for basic support for RBTX4927. > (TX39 boards does not have audio-support, so dma txx9 driver can be > dropped too) Agreed, I don't test audio anyway, but I know it used to work (I had intended to use the board as an MPD media server, but never got beyond the prototyping phase). Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

1 0

build fails using -flto=auto
by GitHub issues - edited 07 Jan '21

07 Jan '21

alsa-project/alsa-lib issue #110 was edited from shoober420: When compiling alsa-lib with -flto=auto, the build will fail. Here are the logs. 64bit: [alsa-lib-git-1.2.4-1-x86_64-build.log](https://github.com/alsa-project/alsa… 32bit: [lib32-alsa-lib-minimal-git-1.2.4-1-x86_64-build.log](https://github.com/als… 64bit --with-versioned: [alsa-lib-git-1.2.4-1-x86_64-build.log](https://github.com/alsa-project/alsa… Issue URL : https://github.com/alsa-project/alsa-lib/issues/110 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

build fails using -flto=auto
by GitHub issues - opened 07 Jan '21

07 Jan '21

alsa-project/alsa-lib issue #110 was opened from shoober420: When compiling alsa-lib with -flto=auto, the build will fail. Here are the logs. 64bit: [alsa-lib-git-1.2.4-1-x86_64-build.log](https://github.com/alsa-project/alsa… 32bit: [lib32-alsa-lib-minimal-git-1.2.4-1-x86_64-build.log](https://github.com/als… Issue URL : https://github.com/alsa-project/alsa-lib/issues/110 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

dmix + dshare and changes of period_size buffer_size
by GitHub issues - opened 06 Jan '21

06 Jan '21

alsa-project/alsa-lib issue #109 was opened from pietercardoen: Dear I'm trying to create my audio configuration using dmix or dshare plugins but i'm facing some issues. Currently, if I don't use dmix or dshare, I'm able to specify the period_size using aplay and changes are populated to the hardware. However if I include a dmix/dshare plugin, I'm not able to configure the period_time anymore. To configure dmix, one needs to specify this: slave { # Slave definition pcm STR # slave PCM name # or pcm { } # slave PCM definition format STR # format definition rate INT # rate definition channels INT period_time INT # in usec # or period_size INT # in frames buffer_time INT # in usec # or buffer_size INT # in frames periods INT # when buffer_size or buffer_time is not specified } Is it possible in any way to avoid the need to specify buffer_size/periods/... and to avoid the system to fall back on defaults? Kind regards Pieter Cardoen Issue URL : https://github.com/alsa-project/alsa-lib/issues/109 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: cros_ec_codec: Reset I2S RX when probing
by Yu-Hsuan Hsu 06 Jan '21

06 Jan '21

It is not guaranteed that I2S RX is disabled when the kernel booting. For example, if the kernel crashes while it is enabled, it will keep enabled until the next time EC reboots. Reset I2S RX when probing to fix this issue. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- include/linux/platform_data/cros_ec_commands.h | 1 + sound/soc/codecs/cros_ec_codec.c | 7 +++++++ 2 files changed, 8 insertions(+) diff --git a/include/linux/platform_data/cros_ec_commands.h b/include/linux/platform_data/cros_ec_commands.h index 86376779ab31..95889ada83a3 100644 --- a/include/linux/platform_data/cros_ec_commands.h +++ b/include/linux/platform_data/cros_ec_commands.h @@ -4600,6 +4600,7 @@ enum ec_codec_i2s_rx_subcmd { EC_CODEC_I2S_RX_SET_SAMPLE_DEPTH = 0x2, EC_CODEC_I2S_RX_SET_DAIFMT = 0x3, EC_CODEC_I2S_RX_SET_BCLK = 0x4, + EC_CODEC_I2S_RX_RESET = 0x5, EC_CODEC_I2S_RX_SUBCMD_COUNT, }; diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index f33a2a9654e7..28b3e2c48c86 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -1011,6 +1011,13 @@ static int cros_ec_codec_platform_probe(struct platform_device *pdev) } priv->ec_capabilities = r.capabilities; + /* Reset EC codec i2s rx. */ + p.cmd = EC_CODEC_I2S_RX_RESET; + ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX, + (uint8_t *)&p, sizeof(p), NULL, 0); + if (ret) + dev_warn(dev, "failed to EC_CODEC_I2S_RESET: %d\n", ret); + platform_set_drvdata(pdev, priv); ret = devm_snd_soc_register_component(dev, &i2s_rx_component_driver, -- 2.29.2.729.g45daf8777d-goog

2 1

[PATCH] ASoC: rt715: add main capture switch and main capture volume control
by jack.yu＠realtek.com 06 Jan '21

06 Jan '21

From: Jack Yu <jack.yu(a)realtek.com> Add main capture switch and main capture volume for callback to be in single operation. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715.c | 174 +++++++++++++++++++++++---------------- 1 file changed, 105 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 9a7d393d424a..cdcba70146da 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -80,85 +80,106 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; unsigned int addr_h, addr_l, val_h, val_ll, val_lr; - unsigned int read_ll, read_rl; - int i; - - /* Can't use update bit function, so read the original value first */ - addr_h = mc->reg; - addr_l = mc->rreg; - if (mc->shift == RT715_DIR_OUT_SFT) /* output */ - val_h = 0x80; - else /* input */ - val_h = 0x0; - - rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + unsigned int read_ll, read_rl, i, j, loop_cnt; - /* L Channel */ - if (mc->invert) { - /* for mute */ - val_ll = (mc->max - ucontrol->value.integer.value[0]) << 7; - /* keep gain */ - read_ll = read_ll & 0x7f; - val_ll |= read_ll; - } else { - /* for gain */ - val_ll = ((ucontrol->value.integer.value[0]) & 0x7f); - if (val_ll > mc->max) - val_ll = mc->max; - /* keep mute status */ - read_ll = read_ll & 0x80; - val_ll |= read_ll; - } - - /* R Channel */ - if (mc->invert) { - regmap_write(rt715->regmap, - RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); - /* for mute */ - val_lr = (mc->max - ucontrol->value.integer.value[1]) << 7; - /* keep gain */ - read_rl = read_rl & 0x7f; - val_lr |= read_rl; - } else { - /* for gain */ - val_lr = ((ucontrol->value.integer.value[1]) & 0x7f); - if (val_lr > mc->max) - val_lr = mc->max; - /* keep mute status */ - read_rl = read_rl & 0x80; - val_lr |= read_rl; - } - - for (i = 0; i < 3; i++) { /* retry 3 times at most */ + if (strstr(ucontrol->id.name, "Main Capture Switch") || + strstr(ucontrol->id.name, "Main Capture Volume")) + loop_cnt = 4; + else + loop_cnt = 1; - if (val_ll == val_lr) { - /* Set both L/R channels at the same time */ - val_h = (1 << mc->shift) | (3 << 4); - regmap_write(rt715->regmap, addr_h, - (val_h << 8 | val_ll)); - regmap_write(rt715->regmap, addr_l, - (val_h << 8 | val_ll)); + for (j = 0; j < loop_cnt; j++) { + /* Can't use update bit function, so read the original value first */ + if (loop_cnt == 1) { + addr_h = mc->reg; + addr_l = mc->rreg; } else { - /* Lch*/ - val_h = (1 << mc->shift) | (1 << 5); - regmap_write(rt715->regmap, addr_h, - (val_h << 8 | val_ll)); - /* Rch */ - val_h = (1 << mc->shift) | (1 << 4); - regmap_write(rt715->regmap, addr_l, - (val_h << 8 | val_lr)); + addr_h = capture_reg_H[j]; + addr_l = capture_reg_L[j]; } - /* check result */ + if (mc->shift == RT715_DIR_OUT_SFT) /* output */ val_h = 0x80; else /* input */ val_h = 0x0; - rt715_get_gain(rt715, addr_h, addr_l, val_h, - &read_rl, &read_ll); - if (read_rl == val_lr && read_ll == val_ll) - break; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* L Channel */ + if (mc->invert) { + /* for mute */ + val_ll = (mc->max - ucontrol->value.integer.value[0]) << 7; + /* keep gain */ + read_ll = read_ll & 0x7f; + val_ll |= read_ll; + } else { + /* for gain */ + val_ll = ((ucontrol->value.integer.value[0]) & 0x7f); + if (val_ll > mc->max) + val_ll = mc->max; + /* keep mute status */ + read_ll = read_ll & 0x80; + val_ll |= read_ll; + } + + /* R Channel */ + if (mc->invert) { + /* for mute */ + val_lr = (mc->max - ucontrol->value.integer.value[1]) << 7; + /* keep gain */ + read_rl = read_rl & 0x7f; + val_lr |= read_rl; + } else { + /* for gain */ + val_lr = ((ucontrol->value.integer.value[1]) & 0x7f); + if (val_lr > mc->max) + val_lr = mc->max; + /* keep mute status */ + read_rl = read_rl & 0x80; + val_lr |= read_rl; + } + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << mc->shift) | (3 << 4); + regmap_write(rt715->regmap, addr_h, + (val_h << 8 | val_ll)); + regmap_write(rt715->regmap, addr_l, + (val_h << 8 | val_ll)); + } else { + /* Lch*/ + val_h = (1 << mc->shift) | (1 << 5); + regmap_write(rt715->regmap, addr_h, + (val_h << 8 | val_ll)); + /* Rch */ + val_h = (1 << mc->shift) | (1 << 4); + regmap_write(rt715->regmap, addr_l, + (val_h << 8 | val_lr)); + } + /* check result */ + if (mc->shift == RT715_DIR_OUT_SFT) /* output */ + val_h = 0x80; + else /* input */ + val_h = 0x0; + + rt715_get_gain(rt715, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } } /* D0:power on state, D3: power saving mode */ if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) @@ -226,6 +247,13 @@ static const struct snd_kcontrol_new rt715_snd_controls[] = { SOC_DOUBLE_R_EXT("ADC 27 Capture Switch", RT715_SET_GAIN_MIX_ADC2_H, RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 1, 1, rt715_set_amp_gain_get, rt715_set_amp_gain_put), + /* + * "Main Capture Switch" looks the same as "ADC 07 Capture Switch", + * but the callback has different action internally. + */ + SOC_DOUBLE_R_EXT("Main Capture Switch", RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_MIC_ADC_L, RT715_DIR_IN_SFT, 1, 1, + rt715_set_amp_gain_get, rt715_set_amp_gain_put), /* Volume Control */ SOC_DOUBLE_R_EXT_TLV("ADC 07 Capture Volume", RT715_SET_GAIN_MIC_ADC_H, RT715_SET_GAIN_MIC_ADC_L, RT715_DIR_IN_SFT, 0x3f, 0, @@ -243,6 +271,14 @@ static const struct snd_kcontrol_new rt715_snd_controls[] = { RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 0x3f, 0, rt715_set_amp_gain_get, rt715_set_amp_gain_put, in_vol_tlv), + /* + * "Main Capture Volume" looks the same as "ADC 07 Capture Volume", + * but the callback has different action internally. + */ + SOC_DOUBLE_R_EXT_TLV("Main Capture Volume", RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_MIC_ADC_L, RT715_DIR_IN_SFT, 0x3f, 0, + rt715_set_amp_gain_get, rt715_set_amp_gain_put, + in_vol_tlv), /* MIC Boost Control */ SOC_DOUBLE_R_EXT_TLV("DMIC1 Boost", RT715_SET_GAIN_DMIC1_H, RT715_SET_GAIN_DMIC1_L, RT715_DIR_IN_SFT, 3, 0, -- 2.29.0

6 12

[PATCH v2 0/9] soundwire/regmap: use _no_pm routines
by Bard Liao 06 Jan '21

06 Jan '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

2 10

[RFC PATCH] ASoC: rt1015: remove bclk_ratio
by Tzung-Bi Shih 05 Jan '21

05 Jan '21

bclk_ratio is unused. Removes bclk_ratio and .set_bclk_ratio callback. Removes snd_soc_dai_set_bclk_ratio() in a few machine drivers which are obviously using rt1015. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- The patch applies after the series: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-December/178608.… and https://mailman.alsa-project.org/pipermail/alsa-devel/2020-December/178614.… sound/soc/amd/acp3x-rt5682-max9836.c | 4 +-- sound/soc/codecs/rt1015.c | 28 ------------------- sound/soc/codecs/rt1015.h | 1 - sound/soc/intel/boards/sof_rt5682.c | 7 ----- .../mediatek/mt8183/mt8183-da7219-max98357.c | 6 ---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 6 ---- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 6 ---- 7 files changed, 1 insertion(+), 57 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 1a4e8ca0f99c..cea320ad0e1c 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -140,9 +140,7 @@ static int acp3x_1015_hw_params(struct snd_pcm_substream *substream, for_each_rtd_codec_dais(rtd, i, codec_dai) { if (strcmp(codec_dai->name, "rt1015-aif")) continue; - ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64); - if (ret < 0) - return ret; + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, 64 * srate, 256 * srate); if (ret < 0) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 2dcb7b0fba60..dcb1261bbffa 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -884,14 +884,6 @@ static int rt1015_set_component_pll(struct snd_soc_component *component, freq_out == rt1015->pll_out) return 0; - if (source == RT1015_PLL_S_BCLK) { - if (rt1015->bclk_ratio == 0) { - dev_err(component->dev, - "Can not support bclk ratio as 0.\n"); - return -EINVAL; - } - } - switch (source) { case RT1015_PLL_S_MCLK: snd_soc_component_update_bits(component, RT1015_CLK2, @@ -931,23 +923,6 @@ static int rt1015_set_component_pll(struct snd_soc_component *component, return 0; } -static int rt1015_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) -{ - struct snd_soc_component *component = dai->component; - struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); - - dev_dbg(component->dev, "%s ratio=%d\n", __func__, ratio); - - rt1015->bclk_ratio = ratio; - - if (ratio == 50) { - dev_dbg(component->dev, "Unsupport bclk ratio\n"); - return -EINVAL; - } - - return 0; -} - static int rt1015_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { @@ -1054,8 +1029,6 @@ static int rt1015_probe(struct snd_soc_component *component) snd_soc_component_get_drvdata(component); rt1015->component = component; - rt1015->bclk_ratio = 0; - INIT_DELAYED_WORK(&rt1015->flush_work, rt1015_flush_work); return 0; @@ -1076,7 +1049,6 @@ static void rt1015_remove(struct snd_soc_component *component) static struct snd_soc_dai_ops rt1015_aif_dai_ops = { .hw_params = rt1015_hw_params, .set_fmt = rt1015_set_dai_fmt, - .set_bclk_ratio = rt1015_set_bclk_ratio, .set_tdm_slot = rt1015_set_tdm_slot, }; diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index e9b498a754e0..2aeaf65ba793 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -427,7 +427,6 @@ struct rt1015_priv { struct regmap *regmap; int sysclk; int sysclk_src; - int bclk_ratio; int pll_src; int pll_in; int pll_out; diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 8b1ca2da9bb9..55505e207bc0 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -323,13 +323,6 @@ static int sof_rt1015_hw_params(struct snd_pcm_substream *substream, fs = 64; for_each_rtd_codec_dais(rtd, i, codec_dai) { - /* Set tdm/i2s1 master bclk ratio */ - ret = snd_soc_dai_set_bclk_ratio(codec_dai, fs); - if (ret < 0) { - dev_err(card->dev, "failed to set bclk ratio\n"); - return ret; - } - ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, params_rate(params) * fs, params_rate(params) * 256); diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 078e58f1ad0b..870cdfd17fff 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -125,12 +125,6 @@ mt8183_da7219_rt1015_i2s_hw_params(struct snd_pcm_substream *substream, for_each_rtd_codec_dais(rtd, i, codec_dai) { if (!strcmp(codec_dai->component->name, RT1015_DEV0_NAME) || !strcmp(codec_dai->component->name, RT1015_DEV1_NAME)) { - ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64); - if (ret) { - dev_err(rtd->dev, "failed to set bclk ratio\n"); - return ret; - } - ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, rate * 64, rate * 256); diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 8c8340854859..0e4892245035 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -68,12 +68,6 @@ mt8183_mt6358_rt1015_i2s_hw_params(struct snd_pcm_substream *substream, int ret, i; for_each_rtd_codec_dais(rtd, i, codec_dai) { - ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64); - if (ret < 0) { - dev_err(card->dev, "failed to set bclk ratio\n"); - return ret; - } - ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, rate * 64, rate * 256); if (ret < 0) { diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index 716fbb4126b5..a988ce1e58de 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -46,12 +46,6 @@ static int mt8192_rt1015_i2s_hw_params(struct snd_pcm_substream *substream, int ret, i; for_each_rtd_codec_dais(rtd, i, codec_dai) { - ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64); - if (ret) { - dev_err(card->dev, "failed to set bclk ratio\n"); - return ret; - } - ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, params_rate(params) * 64, -- 2.29.2.729.g45daf8777d-goog

2 1

[PATCH] ASoC: rt1015: re-calibrate again when resuming
by Tzung-Bi Shih 05 Jan '21

05 Jan '21

Assuming the calibration state gets lost after system suspend. Re-calibrates again when resuming. The rt1015_priv is alloced by kzalloc. No need to initialize cali_done to 0 in component probe callback. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- The idea is inspired from previous discussion: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-December/178229.… The patch applies after the series: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-December/178608.… v1 attempt: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-December/178614.… sound/soc/codecs/rt1015.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 4a9e2eaebe30..3b06eaddccaf 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -1055,7 +1055,6 @@ static int rt1015_probe(struct snd_soc_component *component) rt1015->component = component; rt1015->bclk_ratio = 0; - rt1015->cali_done = 0; INIT_DELAYED_WORK(&rt1015->flush_work, rt1015_flush_work); @@ -1113,6 +1112,10 @@ static int rt1015_resume(struct snd_soc_component *component) regcache_cache_only(rt1015->regmap, false); regcache_sync(rt1015->regmap); + + if (rt1015->cali_done) + rt1015_calibrate(rt1015); + return 0; } #else -- 2.29.2.729.g45daf8777d-goog

2 1

[resend/standalone PATCH v4] Add auxiliary bus support
by Dan Williams 05 Jan '21

05 Jan '21

From: Dave Ertman <david.m.ertman(a)intel.com> Add support for the Auxiliary Bus, auxiliary_device and auxiliary_driver. It enables drivers to create an auxiliary_device and bind an auxiliary_driver to it. The bus supports probe/remove shutdown and suspend/resume callbacks. Each auxiliary_device has a unique string based id; driver binds to an auxiliary_device based on this id through the bus. Co-developed-by: Kiran Patil <kiran.patil(a)intel.com> Co-developed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Co-developed-by: Fred Oh <fred.oh(a)linux.intel.com> Co-developed-by: Leon Romanovsky <leonro(a)nvidia.com> Signed-off-by: Kiran Patil <kiran.patil(a)intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Fred Oh <fred.oh(a)linux.intel.com> Signed-off-by: Leon Romanovsky <leonro(a)nvidia.com> Signed-off-by: Dave Ertman <david.m.ertman(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Shiraz Saleem <shiraz.saleem(a)intel.com> Reviewed-by: Parav Pandit <parav(a)mellanox.com> Reviewed-by: Dan Williams <dan.j.williams(a)intel.com> Reviewed-by: Martin Habets <mhabets(a)solarflare.com> Link: https://lore.kernel.org/r/20201113161859.1775473-2-david.m.ertman@intel.com Signed-off-by: Dan Williams <dan.j.williams(a)intel.com> --- This patch is "To:" the maintainers that have a pending backlog of driver updates dependent on this facility, and "Cc:" Greg. Greg, I understand you have asked for more time to fully review this and apply it to driver-core.git, likely for v5.12, but please consider Acking it for v5.11 instead. It looks good to me and several other stakeholders. Namely, stakeholders that have pressure building up behind this facility in particular Mellanox RDMA, but also SOF, Intel Ethernet, and later on Compute Express Link. I will take the blame for the 2 months of silence that made this awkward to take through driver-core.git, but at the same time I do not want to see that communication mistake inconvenience other parties that reasonably thought this was shaping up to land in v5.11. I am willing to host this version at: git://git.kernel.org/pub/scm/linux/kernel/git/djbw/linux tags/auxiliary-bus-for-5.11 ...for all the independent drivers to have a common commit baseline. It is not there yet pending Greg's Ack. For example implementations incorporating this patch, see Dave Ertman's SOF series: https://lore.kernel.org/r/20201113161859.1775473-2-david.m.ertman@intel.com ...and Leon's mlx5 series: http://lore.kernel.org/r/20201026111849.1035786-1-leon@kernel.org PS: Greg I know I promised some review on newcomer patches to help with your queue, unfortunately Intel-internal review is keeping my plate full. Again, I do not want other stakeholder to be waiting on me to resolve that backlog. Documentation/driver-api/auxiliary_bus.rst | 234 ++++++++++++++++++++++++ Documentation/driver-api/index.rst | 1 drivers/base/Kconfig | 3 drivers/base/Makefile | 1 drivers/base/auxiliary.c | 268 ++++++++++++++++++++++++++++ include/linux/auxiliary_bus.h | 78 ++++++++ include/linux/mod_devicetable.h | 8 + scripts/mod/devicetable-offsets.c | 3 scripts/mod/file2alias.c | 8 + 9 files changed, 604 insertions(+) create mode 100644 Documentation/driver-api/auxiliary_bus.rst create mode 100644 drivers/base/auxiliary.c create mode 100644 include/linux/auxiliary_bus.h diff --git a/Documentation/driver-api/auxiliary_bus.rst b/Documentation/driver-api/auxiliary_bus.rst new file mode 100644 index 000000000000..5dd7804631ef --- /dev/null +++ b/Documentation/driver-api/auxiliary_bus.rst @@ -0,0 +1,234 @@ +.. SPDX-License-Identifier: GPL-2.0-only + +============= +Auxiliary Bus +============= + +In some subsystems, the functionality of the core device (PCI/ACPI/other) is +too complex for a single device to be managed by a monolithic driver +(e.g. Sound Open Firmware), multiple devices might implement a common +intersection of functionality (e.g. NICs + RDMA), or a driver may want to +export an interface for another subsystem to drive (e.g. SIOV Physical Function +export Virtual Function management). A split of the functinoality into child- +devices representing sub-domains of functionality makes it possible to +compartmentalize, layer, and distribute domain-specific concerns via a Linux +device-driver model. + +An example for this kind of requirement is the audio subsystem where a single +IP is handling multiple entities such as HDMI, Soundwire, local devices such as +mics/speakers etc. The split for the core's functionality can be arbitrary or +be defined by the DSP firmware topology and include hooks for test/debug. This +allows for the audio core device to be minimal and focused on hardware-specific +control and communication. + +Each auxiliary_device represents a part of its parent functionality. The +generic behavior can be extended and specialized as needed by encapsulating an +auxiliary_device within other domain-specific structures and the use of .ops +callbacks. Devices on the auxiliary bus do not share any structures and the use +of a communication channel with the parent is domain-specific. + +Note that ops are intended as a way to augment instance behavior within a class +of auxiliary devices, it is not the mechanism for exporting common +infrastructure from the parent. Consider EXPORT_SYMBOL_NS() to convey +infrastructure from the parent module to the auxiliary module(s). + + +When Should the Auxiliary Bus Be Used +===================================== + +The auxiliary bus is to be used when a driver and one or more kernel modules, +who share a common header file with the driver, need a mechanism to connect and +provide access to a shared object allocated by the auxiliary_device's +registering driver. The registering driver for the auxiliary_device(s) and the +kernel module(s) registering auxiliary_drivers can be from the same subsystem, +or from multiple subsystems. + +The emphasis here is on a common generic interface that keeps subsystem +customization out of the bus infrastructure. + +One example is a PCI network device that is RDMA-capable and exports a child +device to be driven by an auxiliary_driver in the RDMA subsystem. The PCI +driver allocates and registers an auxiliary_device for each physical +function on the NIC. The RDMA driver registers an auxiliary_driver that claims +each of these auxiliary_devices. This conveys data/ops published by the parent +PCI device/driver to the RDMA auxiliary_driver. + +Another use case is for the PCI device to be split out into multiple sub +functions. For each sub function an auxiliary_device is created. A PCI sub +function driver binds to such devices that creates its own one or more class +devices. A PCI sub function auxiliary device is likely to be contained in a +struct with additional attributes such as user defined sub function number and +optional attributes such as resources and a link to the parent device. These +attributes could be used by systemd/udev; and hence should be initialized +before a driver binds to an auxiliary_device. + +A key requirement for utilizing the auxiliary bus is that there is no +dependency on a physical bus, device, register accesses or regmap support. +These individual devices split from the core cannot live on the platform bus as +they are not physical devices that are controlled by DT/ACPI. The same +argument applies for not using MFD in this scenario as MFD relies on individual +function devices being physical devices. + +Auxiliary Device +================ + +An auxiliary_device represents a part of its parent device's functionality. It +is given a name that, combined with the registering drivers KBUILD_MODNAME, +creates a match_name that is used for driver binding, and an id that combined +with the match_name provide a unique name to register with the bus subsystem. + +Registering an auxiliary_device is a two-step process. First call +auxiliary_device_init(), which checks several aspects of the auxiliary_device +struct and performs a device_initialize(). After this step completes, any +error state must have a call to auxiliary_device_uninit() in its resolution path. +The second step in registering an auxiliary_device is to perform a call to +auxiliary_device_add(), which sets the name of the device and add the device to +the bus. + +Unregistering an auxiliary_device is also a two-step process to mirror the +register process. First call auxiliary_device_delete(), then call +auxiliary_device_uninit(). + +.. code-block:: c + + struct auxiliary_device { + struct device dev; + const char *name; + u32 id; + }; + +If two auxiliary_devices both with a match_name "mod.foo" are registered onto +the bus, they must have unique id values (e.g. "x" and "y") so that the +registered devices names are "mod.foo.x" and "mod.foo.y". If match_name + id +are not unique, then the device_add fails and generates an error message. + +The auxiliary_device.dev.type.release or auxiliary_device.dev.release must be +populated with a non-NULL pointer to successfully register the auxiliary_device. + +The auxiliary_device.dev.parent must also be populated. + +Auxiliary Device Memory Model and Lifespan +------------------------------------------ + +The registering driver is the entity that allocates memory for the +auxiliary_device and register it on the auxiliary bus. It is important to note +that, as opposed to the platform bus, the registering driver is wholly +responsible for the management for the memory used for the driver object. + +A parent object, defined in the shared header file, contains the +auxiliary_device. It also contains a pointer to the shared object(s), which +also is defined in the shared header. Both the parent object and the shared +object(s) are allocated by the registering driver. This layout allows the +auxiliary_driver's registering module to perform a container_of() call to go +from the pointer to the auxiliary_device, that is passed during the call to the +auxiliary_driver's probe function, up to the parent object, and then have +access to the shared object(s). + +The memory for the auxiliary_device is freed only in its release() callback +flow as defined by its registering driver. + +The memory for the shared object(s) must have a lifespan equal to, or greater +than, the lifespan of the memory for the auxiliary_device. The auxiliary_driver +should only consider that this shared object is valid as long as the +auxiliary_device is still registered on the auxiliary bus. It is up to the +registering driver to manage (e.g. free or keep available) the memory for the +shared object beyond the life of the auxiliary_device. + +The registering driver must unregister all auxiliary devices before its own +driver.remove() is completed. + +Auxiliary Drivers +================= + +Auxiliary drivers follow the standard driver model convention, where +discovery/enumeration is handled by the core, and drivers +provide probe() and remove() methods. They support power management +and shutdown notifications using the standard conventions. + +.. code-block:: c + + struct auxiliary_driver { + int (*probe)(struct auxiliary_device *, + const struct auxiliary_device_id *id); + int (*remove)(struct auxiliary_device *); + void (*shutdown)(struct auxiliary_device *); + int (*suspend)(struct auxiliary_device *, pm_message_t); + int (*resume)(struct auxiliary_device *); + struct device_driver driver; + const struct auxiliary_device_id *id_table; + }; + +Auxiliary drivers register themselves with the bus by calling +auxiliary_driver_register(). The id_table contains the match_names of auxiliary +devices that a driver can bind with. + +Example Usage +============= + +Auxiliary devices are created and registered by a subsystem-level core device +that needs to break up its functionality into smaller fragments. One way to +extend the scope of an auxiliary_device is to encapsulate it within a domain- +pecific structure defined by the parent device. This structure contains the +auxiliary_device and any associated shared data/callbacks needed to establish +the connection with the parent. + +An example is: + +.. code-block:: c + + struct foo { + struct auxiliary_device auxdev; + void (*connect)(struct auxiliary_device *auxdev); + void (*disconnect)(struct auxiliary_device *auxdev); + void *data; + }; + +The parent device then registers the auxiliary_device by calling +auxiliary_device_init(), and then auxiliary_device_add(), with the pointer to +the auxdev member of the above structure. The parent provides a name for the +auxiliary_device that, combined with the parent's KBUILD_MODNAME, creates a +match_name that is be used for matching and binding with a driver. + +Whenever an auxiliary_driver is registered, based on the match_name, the +auxiliary_driver's probe() is invoked for the matching devices. The +auxiliary_driver can also be encapsulated inside custom drivers that make the +core device's functionality extensible by adding additional domain-specific ops +as follows: + +.. code-block:: c + + struct my_ops { + void (*send)(struct auxiliary_device *auxdev); + void (*receive)(struct auxiliary_device *auxdev); + }; + + + struct my_driver { + struct auxiliary_driver auxiliary_drv; + const struct my_ops ops; + }; + +An example of this type of usage is: + +.. code-block:: c + + const struct auxiliary_device_id my_auxiliary_id_table[] = { + { .name = "foo_mod.foo_dev" }, + { }, + }; + + const struct my_ops my_custom_ops = { + .send = my_tx, + .receive = my_rx, + }; + + const struct my_driver my_drv = { + .auxiliary_drv = { + .name = "myauxiliarydrv", + .id_table = my_auxiliary_id_table, + .probe = my_probe, + .remove = my_remove, + .shutdown = my_shutdown, + }, + .ops = my_custom_ops, + }; diff --git a/Documentation/driver-api/index.rst b/Documentation/driver-api/index.rst index f357f3eb400c..86759a74b7f1 100644 --- a/Documentation/driver-api/index.rst +++ b/Documentation/driver-api/index.rst @@ -72,6 +72,7 @@ available subsections can be seen below. thermal/index fpga/index acpi/index + auxiliary_bus backlight/lp855x-driver.rst connector console diff --git a/drivers/base/Kconfig b/drivers/base/Kconfig index 8d7001712062..040be48ce046 100644 --- a/drivers/base/Kconfig +++ b/drivers/base/Kconfig @@ -1,6 +1,9 @@ # SPDX-License-Identifier: GPL-2.0 menu "Generic Driver Options" +config AUXILIARY_BUS + bool + config UEVENT_HELPER bool "Support for uevent helper" help diff --git a/drivers/base/Makefile b/drivers/base/Makefile index 41369fc7004f..5e7bf9669a81 100644 --- a/drivers/base/Makefile +++ b/drivers/base/Makefile @@ -7,6 +7,7 @@ obj-y := component.o core.o bus.o dd.o syscore.o \ attribute_container.o transport_class.o \ topology.o container.o property.o cacheinfo.o \ swnode.o +obj-$(CONFIG_AUXILIARY_BUS) += auxiliary.o obj-$(CONFIG_DEVTMPFS) += devtmpfs.o obj-y += power/ obj-$(CONFIG_ISA_BUS_API) += isa.o diff --git a/drivers/base/auxiliary.c b/drivers/base/auxiliary.c new file mode 100644 index 000000000000..ef2af417438b --- /dev/null +++ b/drivers/base/auxiliary.c @@ -0,0 +1,268 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Copyright (c) 2019-2020 Intel Corporation + * + * Please see Documentation/driver-api/auxiliary_bus.rst for more information. + */ + +#define pr_fmt(fmt) "%s:%s: " fmt, KBUILD_MODNAME, __func__ + +#include <linux/device.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pm_domain.h> +#include <linux/pm_runtime.h> +#include <linux/string.h> +#include <linux/auxiliary_bus.h> + +static const struct auxiliary_device_id *auxiliary_match_id(const struct auxiliary_device_id *id, + const struct auxiliary_device *auxdev) +{ + for (; id->name[0]; id++) { + const char *p = strrchr(dev_name(&auxdev->dev), '.'); + int match_size; + + if (!p) + continue; + match_size = p - dev_name(&auxdev->dev); + + /* use dev_name(&auxdev->dev) prefix before last '.' char to match to */ + if (strlen(id->name) == match_size && + !strncmp(dev_name(&auxdev->dev), id->name, match_size)) + return id; + } + return NULL; +} + +static int auxiliary_match(struct device *dev, struct device_driver *drv) +{ + struct auxiliary_device *auxdev = to_auxiliary_dev(dev); + struct auxiliary_driver *auxdrv = to_auxiliary_drv(drv); + + return !!auxiliary_match_id(auxdrv->id_table, auxdev); +} + +static int auxiliary_uevent(struct device *dev, struct kobj_uevent_env *env) +{ + const char *name, *p; + + name = dev_name(dev); + p = strrchr(name, '.'); + + return add_uevent_var(env, "MODALIAS=%s%.*s", AUXILIARY_MODULE_PREFIX, (int)(p - name), + name); +} + +static const struct dev_pm_ops auxiliary_dev_pm_ops = { + SET_RUNTIME_PM_OPS(pm_generic_runtime_suspend, pm_generic_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_generic_suspend, pm_generic_resume) +}; + +static int auxiliary_bus_probe(struct device *dev) +{ + struct auxiliary_driver *auxdrv = to_auxiliary_drv(dev->driver); + struct auxiliary_device *auxdev = to_auxiliary_dev(dev); + int ret; + + ret = dev_pm_domain_attach(dev, true); + if (ret) { + dev_warn(dev, "Failed to attach to PM Domain : %d\n", ret); + return ret; + } + + ret = auxdrv->probe(auxdev, auxiliary_match_id(auxdrv->id_table, auxdev)); + if (ret) + dev_pm_domain_detach(dev, true); + + return ret; +} + +static int auxiliary_bus_remove(struct device *dev) +{ + struct auxiliary_driver *auxdrv = to_auxiliary_drv(dev->driver); + struct auxiliary_device *auxdev = to_auxiliary_dev(dev); + int ret = 0; + + if (auxdrv->remove) + ret = auxdrv->remove(auxdev); + dev_pm_domain_detach(dev, true); + + return ret; +} + +static void auxiliary_bus_shutdown(struct device *dev) +{ + struct auxiliary_driver *auxdrv = to_auxiliary_drv(dev->driver); + struct auxiliary_device *auxdev = to_auxiliary_dev(dev); + + if (auxdrv->shutdown) + auxdrv->shutdown(auxdev); +} + +static struct bus_type auxiliary_bus_type = { + .name = "auxiliary", + .probe = auxiliary_bus_probe, + .remove = auxiliary_bus_remove, + .shutdown = auxiliary_bus_shutdown, + .match = auxiliary_match, + .uevent = auxiliary_uevent, + .pm = &auxiliary_dev_pm_ops, +}; + +/** + * auxiliary_device_init - check auxiliary_device and initialize + * @auxdev: auxiliary device struct + * + * This is the first step in the two-step process to register an auxiliary_device. + * + * When this function returns an error code, then the device_initialize will *not* have + * been performed, and the caller will be responsible to free any memory allocated for the + * auxiliary_device in the error path directly. + * + * It returns 0 on success. On success, the device_initialize has been performed. After this + * point any error unwinding will need to include a call to auxiliary_device_uninit(). + * In this post-initialize error scenario, a call to the device's .release callback will be + * triggered, and all memory clean-up is expected to be handled there. + */ +int auxiliary_device_init(struct auxiliary_device *auxdev) +{ + struct device *dev = &auxdev->dev; + + if (!dev->parent) { + pr_err("auxiliary_device has a NULL dev->parent\n"); + return -EINVAL; + } + + if (!auxdev->name) { + pr_err("auxiliary_device has a NULL name\n"); + return -EINVAL; + } + + dev->bus = &auxiliary_bus_type; + device_initialize(&auxdev->dev); + return 0; +} +EXPORT_SYMBOL_GPL(auxiliary_device_init); + +/** + * __auxiliary_device_add - add an auxiliary bus device + * @auxdev: auxiliary bus device to add to the bus + * @modname: name of the parent device's driver module + * + * This is the second step in the two-step process to register an auxiliary_device. + * + * This function must be called after a successful call to auxiliary_device_init(), which + * will perform the device_initialize. This means that if this returns an error code, then a + * call to auxiliary_device_uninit() must be performed so that the .release callback will + * be triggered to free the memory associated with the auxiliary_device. + * + * The expectation is that users will call the "auxiliary_device_add" macro so that the caller's + * KBUILD_MODNAME is automatically inserted for the modname parameter. Only if a user requires + * a custom name would this version be called directly. + */ +int __auxiliary_device_add(struct auxiliary_device *auxdev, const char *modname) +{ + struct device *dev = &auxdev->dev; + int ret; + + if (!modname) { + pr_err("auxiliary device modname is NULL\n"); + return -EINVAL; + } + + ret = dev_set_name(dev, "%s.%s.%d", modname, auxdev->name, auxdev->id); + if (ret) { + pr_err("auxiliary device dev_set_name failed: %d\n", ret); + return ret; + } + + ret = device_add(dev); + if (ret) + dev_err(dev, "adding auxiliary device failed!: %d\n", ret); + + return ret; +} +EXPORT_SYMBOL_GPL(__auxiliary_device_add); + +/** + * auxiliary_find_device - auxiliary device iterator for locating a particular device. + * @start: Device to begin with + * @data: Data to pass to match function + * @match: Callback function to check device + * + * This function returns a reference to a device that is 'found' + * for later use, as determined by the @match callback. + * + * The callback should return 0 if the device doesn't match and non-zero + * if it does. If the callback returns non-zero, this function will + * return to the caller and not iterate over any more devices. + */ +struct auxiliary_device * +auxiliary_find_device(struct device *start, const void *data, + int (*match)(struct device *dev, const void *data)) +{ + struct device *dev; + + dev = bus_find_device(&auxiliary_bus_type, start, data, match); + if (!dev) + return NULL; + + return to_auxiliary_dev(dev); +} +EXPORT_SYMBOL_GPL(auxiliary_find_device); + +/** + * __auxiliary_driver_register - register a driver for auxiliary bus devices + * @auxdrv: auxiliary_driver structure + * @owner: owning module/driver + * @modname: KBUILD_MODNAME for parent driver + */ +int __auxiliary_driver_register(struct auxiliary_driver *auxdrv, struct module *owner, + const char *modname) +{ + if (WARN_ON(!auxdrv->probe) || WARN_ON(!auxdrv->id_table)) + return -EINVAL; + + if (auxdrv->name) + auxdrv->driver.name = kasprintf(GFP_KERNEL, "%s.%s", modname, auxdrv->name); + else + auxdrv->driver.name = kasprintf(GFP_KERNEL, "%s", modname); + if (!auxdrv->driver.name) + return -ENOMEM; + + auxdrv->driver.owner = owner; + auxdrv->driver.bus = &auxiliary_bus_type; + auxdrv->driver.mod_name = modname; + + return driver_register(&auxdrv->driver); +} +EXPORT_SYMBOL_GPL(__auxiliary_driver_register); + +/** + * auxiliary_driver_unregister - unregister a driver + * @auxdrv: auxiliary_driver structure + */ +void auxiliary_driver_unregister(struct auxiliary_driver *auxdrv) +{ + driver_unregister(&auxdrv->driver); + kfree(auxdrv->driver.name); +} +EXPORT_SYMBOL_GPL(auxiliary_driver_unregister); + +static int __init auxiliary_bus_init(void) +{ + return bus_register(&auxiliary_bus_type); +} + +static void __exit auxiliary_bus_exit(void) +{ + bus_unregister(&auxiliary_bus_type); +} + +module_init(auxiliary_bus_init); +module_exit(auxiliary_bus_exit); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Auxiliary Bus"); +MODULE_AUTHOR("David Ertman <david.m.ertman(a)intel.com>"); +MODULE_AUTHOR("Kiran Patil <kiran.patil(a)intel.com>"); diff --git a/include/linux/auxiliary_bus.h b/include/linux/auxiliary_bus.h new file mode 100644 index 000000000000..282fbf7bf9af --- /dev/null +++ b/include/linux/auxiliary_bus.h @@ -0,0 +1,78 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Copyright (c) 2019-2020 Intel Corporation + * + * Please see Documentation/driver-api/auxiliary_bus.rst for more information. + */ + +#ifndef _AUXILIARY_BUS_H_ +#define _AUXILIARY_BUS_H_ + +#include <linux/device.h> +#include <linux/mod_devicetable.h> +#include <linux/slab.h> + +struct auxiliary_device { + struct device dev; + const char *name; + u32 id; +}; + +struct auxiliary_driver { + int (*probe)(struct auxiliary_device *auxdev, const struct auxiliary_device_id *id); + int (*remove)(struct auxiliary_device *auxdev); + void (*shutdown)(struct auxiliary_device *auxdev); + int (*suspend)(struct auxiliary_device *auxdev, pm_message_t state); + int (*resume)(struct auxiliary_device *auxdev); + const char *name; + struct device_driver driver; + const struct auxiliary_device_id *id_table; +}; + +static inline struct auxiliary_device *to_auxiliary_dev(struct device *dev) +{ + return container_of(dev, struct auxiliary_device, dev); +} + +static inline struct auxiliary_driver *to_auxiliary_drv(struct device_driver *drv) +{ + return container_of(drv, struct auxiliary_driver, driver); +} + +int auxiliary_device_init(struct auxiliary_device *auxdev); +int __auxiliary_device_add(struct auxiliary_device *auxdev, const char *modname); +#define auxiliary_device_add(auxdev) __auxiliary_device_add(auxdev, KBUILD_MODNAME) + +static inline void auxiliary_device_uninit(struct auxiliary_device *auxdev) +{ + put_device(&auxdev->dev); +} + +static inline void auxiliary_device_delete(struct auxiliary_device *auxdev) +{ + device_del(&auxdev->dev); +} + +int __auxiliary_driver_register(struct auxiliary_driver *auxdrv, struct module *owner, + const char *modname); +#define auxiliary_driver_register(auxdrv) \ + __auxiliary_driver_register(auxdrv, THIS_MODULE, KBUILD_MODNAME) + +void auxiliary_driver_unregister(struct auxiliary_driver *auxdrv); + +/** + * module_auxiliary_driver() - Helper macro for registering an auxiliary driver + * @__auxiliary_driver: auxiliary driver struct + * + * Helper macro for auxiliary drivers which do not do anything special in + * module init/exit. This eliminates a lot of boilerplate. Each module may only + * use this macro once, and calling it replaces module_init() and module_exit() + */ +#define module_auxiliary_driver(__auxiliary_driver) \ + module_driver(__auxiliary_driver, auxiliary_driver_register, auxiliary_driver_unregister) + +struct auxiliary_device * +auxiliary_find_device(struct device *start, const void *data, + int (*match)(struct device *dev, const void *data)); + +#endif /* _AUXILIARY_BUS_H_ */ diff --git a/include/linux/mod_devicetable.h b/include/linux/mod_devicetable.h index 5b08a473cdba..c425290b21e2 100644 --- a/include/linux/mod_devicetable.h +++ b/include/linux/mod_devicetable.h @@ -838,4 +838,12 @@ struct mhi_device_id { kernel_ulong_t driver_data; }; +#define AUXILIARY_NAME_SIZE 32 +#define AUXILIARY_MODULE_PREFIX "auxiliary:" + +struct auxiliary_device_id { + char name[AUXILIARY_NAME_SIZE]; + kernel_ulong_t driver_data; +}; + #endif /* LINUX_MOD_DEVICETABLE_H */ diff --git a/scripts/mod/devicetable-offsets.c b/scripts/mod/devicetable-offsets.c index 27007c18e754..e377f52dbfa3 100644 --- a/scripts/mod/devicetable-offsets.c +++ b/scripts/mod/devicetable-offsets.c @@ -243,5 +243,8 @@ int main(void) DEVID(mhi_device_id); DEVID_FIELD(mhi_device_id, chan); + DEVID(auxiliary_device_id); + DEVID_FIELD(auxiliary_device_id, name); + return 0; } diff --git a/scripts/mod/file2alias.c b/scripts/mod/file2alias.c index 2417dd1dee33..fb4827027536 100644 --- a/scripts/mod/file2alias.c +++ b/scripts/mod/file2alias.c @@ -1364,6 +1364,13 @@ static int do_mhi_entry(const char *filename, void *symval, char *alias) { DEF_FIELD_ADDR(symval, mhi_device_id, chan); sprintf(alias, MHI_DEVICE_MODALIAS_FMT, *chan); + return 1; +} + +static int do_auxiliary_entry(const char *filename, void *symval, char *alias) +{ + DEF_FIELD_ADDR(symval, auxiliary_device_id, name); + sprintf(alias, AUXILIARY_MODULE_PREFIX "%s", *name); return 1; } @@ -1442,6 +1449,7 @@ static const struct devtable devtable[] = { {"tee", SIZE_tee_client_device_id, do_tee_entry}, {"wmi", SIZE_wmi_device_id, do_wmi_entry}, {"mhi", SIZE_mhi_device_id, do_mhi_entry}, + {"auxiliary", SIZE_auxiliary_device_id, do_auxiliary_entry}, }; /* Create MODULE_ALIAS() statements.

13 56

[PATCH v3 1/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 05 Jan '21

05 Jan '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 dcab9527ba3d..7c5d950a8628 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2073,6 +2073,9 @@ 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); + static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, struct snd_pcm_substream *substream, int cmd) { @@ -2092,8 +2095,10 @@ 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; @@ -2113,7 +2118,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2123,7 +2128,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2133,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2142,12 +2147,12 @@ 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 = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2155,12 +2160,12 @@ 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 = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2168,17 +2173,20 @@ 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 = dpcm_do_trigger(dpcm, 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; } @@ -2916,10 +2924,9 @@ 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) @@ -2933,17 +2940,12 @@ 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[] = { @@ -2954,6 +2956,23 @@ 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); /* @@ -2963,6 +2982,9 @@ 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, @@ -2970,6 +2992,10 @@ 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

Re: [PATCH 01/10] MIPS: TX49xx: Drop support
by Guenter Roeck 05 Jan '21

05 Jan '21

On 1/5/21 6:02 AM, Thomas Bogendoerfer wrote: > Looks like there are no boards with TX49xx CPUS other than reference > boards available. So it's time to drop Linux support for it. > > Signed-off-by: Thomas Bogendoerfer <tsbogend(a)alpha.franken.de> > --- > drivers/watchdog/Kconfig | 2 +- Acked-by: Guenter Roeck <linux(a)roeck-us.net> Guenter

1 0

[RFC][PATCH v3 0/4] design a way to change audio Jack state by software
by Hui Wang 05 Jan '21

05 Jan '21

the changes in the v3 (for easy to review, divide change into 4 patches): - address the comment about the snd_jack_report() and _snd_jack_report(), the v2 design is hard to understand and is hard to review, in the v3, separate the jack_report to snd_jack_report() and snd_jack_inject_report(), hw jack events call snd_jack_report() as before, if a jack contains multi jack_kctl and the jack_kctl's sw_inject is enabled, the status and the related input-dev's events will not be reproted. The injection events call snd_jack_inject_report(). This change is squashed to 0001-xxx.patch, the rest part of 0001-xxx.patch is same as v2. - address the comment about folders'name in the 0002-xxx.patch, so far, drop the '/', ',', '=' and ' ' from the folders' name. - address the comment about adding more debugfs nodes in the 0003-xxx.patch, it adds kctl_id, mask_bits, status and type. - address the comment about save-n-restore jack's hw status in the 0004-xxx.patch, adding a hw_status_cache and save the last reported jack hw event, once the sw_inject is disabled, will restore all jack_kctl's state under the same snd_jack with hw_status_cache. Now the layout of folders looks like below: /sys/kernel/debug# tree sound-core/ sound-core/ └── card0 ├── HDMIDPpcm3Jack │ ├── jackin_inject │ ├── kctl_id │ ├── mask_bits │ ├── status │ ├── sw_inject_enable │ └── type ├── HDMIDPpcm4Jack │ ├── jackin_inject │ ├── kctl_id │ ├── mask_bits │ ├── status │ ├── sw_inject_enable │ └── type ├── HDMIDPpcm5Jack │ ├── jackin_inject │ ├── kctl_id │ ├── mask_bits │ ├── status │ ├── sw_inject_enable │ └── type ├── HeadphoneJack │ ├── jackin_inject │ ├── kctl_id │ ├── mask_bits │ ├── status │ ├── sw_inject_enable │ └── type └── MicJack ├── jackin_inject ├── kctl_id ├── mask_bits ├── status ├── sw_inject_enable └── type /sys/kernel/debug# cat sound-core/card0/HeadphoneJack/kctl_id Headphone Jack /sys/kernel/debug# cat sound-core/card0/HeadphoneJack/mask_bits 0x0001 HEADPHONE(0x0001) /sys/kernel/debug# cat sound-core/card0/HeadphoneJack/status 0x0000 /sys/kernel/debug# cat sound-core/card0/HeadphoneJack/sw_inject_enable Jack: Headphone Jack Inject Enabled: 0 /sys/kernel/debug# cat sound-core/card0/HeadphoneJack/type 0x0001 HEADPHONE(0x0001) the changes in the V2: - using debugfs instead of sysfs - using jack_ctrl to create a folder instead of snd_jack, since ASoC drivers could create multi jack_ctrls within a snd_jack - create a folder for each jack_ctrl instead for all jack_ctrls [ snip ] Hui Wang (4): alsa: jack: implement software jack injection via debugfs alsa: jack: adjust jack_kctl debugfs folder's name alsa: jack: add more jack_kctl debugfs nodes alsa: jack: implement save-and-restore for jack's hw status include/sound/core.h | 2 + include/sound/jack.h | 1 + sound/core/init.c | 7 + sound/core/jack.c | 351 ++++++++++++++++++++++++++++++++++++++++++- sound/core/sound.c | 8 + 5 files changed, 365 insertions(+), 4 deletions(-) -- 2.25.1

2 8

[PATCH] ASoC: wm8962: Add optional mclk device tree binding
by Adam Ford 04 Jan '21

04 Jan '21

The driver can request an optional clock for mclk. Update the txt file to reflect this. Suggested-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Signed-off-by: Adam Ford <aford173(a)gmail.com> diff --git a/Documentation/devicetree/bindings/sound/wm8962.txt b/Documentation/devicetree/bindings/sound/wm8962.txt index dcfa9a3369fd..c36c649ddfd0 100644 --- a/Documentation/devicetree/bindings/sound/wm8962.txt +++ b/Documentation/devicetree/bindings/sound/wm8962.txt @@ -9,6 +9,9 @@ Required properties: - reg : the I2C address of the device. Optional properties: + + - clocks : The clock source of the mclk + - spk-mono: This is a boolean property. If present, the SPK_MONO bit of R51 (Class D Control 2) gets set, indicating that the speaker is in mono mode. @@ -27,6 +30,7 @@ Example: wm8962: codec@1a { compatible = "wlf,wm8962"; reg = <0x1a>; + clocks = <&clks IMX6QDL_CLK_CKO>; gpio-cfg = < 0x0000 /* 0:Default */ -- 2.25.1

5 4

[PATCH] ASoC: fsl_aud2htx: select SND_SOC_IMX_PCM_DMA
by Arnd Bergmann 04 Jan '21

04 Jan '21

From: Arnd Bergmann <arnd(a)arndb.de> The newly added driver requires DMA support and fails to build when that is disabled: aarch64-linux-ld: sound/soc/fsl/fsl_aud2htx.o: in function `fsl_aud2htx_probe': fsl_aud2htx.c:(.text+0x3e0): undefined reference to `imx_pcm_dma_init' Fixes: 8a24c834c053 ("ASoC: fsl_aud2htx: Add aud2htx module driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/fsl/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 84db0b7b9d59..d7f30036d434 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -108,6 +108,7 @@ config SND_SOC_FSL_XCVR config SND_SOC_FSL_AUD2HTX tristate "AUDIO TO HDMI TX module support" depends on ARCH_MXC || COMPILE_TEST + select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n help Say Y if you want to add AUDIO TO HDMI TX support for NXP. -- 2.29.2

3 2

[PATCH] ALSA: hda/via: Fix runtime PM for Clevo W35xSS
by Takashi Iwai 04 Jan '21

04 Jan '21

Clevo W35xSS_370SS with VIA codec has had the runtime PM problem that looses the power state of some nodes after the runtime resume. This was worked around by disabling the default runtime PM via a denylist entry. Since 5.10.x made the runtime PM applied (casually) even though it's disabled in the denylist, this problem was revisited. The result was that disabling power_save_node feature suffices for the runtime PM problem. This patch implements the disablement of power_save_node feature in VIA codec for the device. It also drops the former denylist entry, too, as the runtime PM should work in the codec side properly now. Fixes: b529ef2464ad ("ALSA: hda: Add Clevo W35xSS_370SS to the power_save blacklist") Reported-by: Christian Labisch <clnetbox(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 2 -- sound/pci/hda/patch_via.c | 13 +++++++++++++ 2 files changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6852668f1bcb..770ad25f1907 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2220,8 +2220,6 @@ static const struct snd_pci_quirk power_save_denylist[] = { SND_PCI_QUIRK(0x1849, 0x7662, "Asrock H81M-HDS", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), - /* https://bugzilla.redhat.com/show_bug.cgi?id=1581607 */ - SND_PCI_QUIRK(0x1558, 0x3501, "Clevo W35xSS_370SS", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1558, 0x6504, "Clevo W65_67SB", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 7ef8f3105cdb..0ab40a8a68fb 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1002,6 +1002,7 @@ static const struct hda_verb vt1802_init_verbs[] = { enum { VIA_FIXUP_INTMIC_BOOST, VIA_FIXUP_ASUS_G75, + VIA_FIXUP_POWER_SAVE, }; static void via_fixup_intmic_boost(struct hda_codec *codec, @@ -1011,6 +1012,13 @@ static void via_fixup_intmic_boost(struct hda_codec *codec, override_mic_boost(codec, 0x30, 0, 2, 40); } +static void via_fixup_power_save(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + codec->power_save_node = 0; +} + static const struct hda_fixup via_fixups[] = { [VIA_FIXUP_INTMIC_BOOST] = { .type = HDA_FIXUP_FUNC, @@ -1025,11 +1033,16 @@ static const struct hda_fixup via_fixups[] = { { } } }, + [VIA_FIXUP_POWER_SAVE] = { + .type = HDA_FIXUP_FUNC, + .v.func = via_fixup_power_save, + }, }; static const struct snd_pci_quirk vt2002p_fixups[] = { SND_PCI_QUIRK(0x1043, 0x1487, "Asus G75", VIA_FIXUP_ASUS_G75), SND_PCI_QUIRK(0x1043, 0x8532, "Asus X202E", VIA_FIXUP_INTMIC_BOOST), + SND_PCI_QUIRK(0x1558, 0x3501, "Clevo W35xSS_370SS", VIA_FIXUP_POWER_SAVE), {} }; -- 2.26.2

1 0

[PATCH] ASoC: rt1015: modify calibration sequence for better performance
by Jack Yu 04 Jan '21

04 Jan '21

New calibration sequence to fix power issue in idle state. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 40 ++++++++++----------------------------- 1 file changed, 10 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 32e6bcf763d1..5fdf7bbfd375 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -497,40 +497,20 @@ static void rt1015_calibrate(struct rt1015_priv *rt1015) snd_soc_dapm_mutex_lock(&component->dapm); regcache_cache_bypass(regmap, true); - regmap_write(regmap, RT1015_PWR9, 0xAA60); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0089); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008A); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008C); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008D); - regmap_write(regmap, RT1015_PWR4, 0x80B2); - regmap_write(regmap, RT1015_CLASSD_SEQ, 0x5797); - regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2100); - regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0100); - regmap_write(regmap, RT1015_PWR5, 0x2175); - regmap_write(regmap, RT1015_MIXER1, 0x005D); - regmap_write(regmap, RT1015_CLSD_INTERNAL1, 0x00A1); - regmap_write(regmap, RT1015_CLSD_INTERNAL2, 0x12F7); - regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x1205); - msleep(200); - regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2000); - regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0180); - regmap_write(regmap, RT1015_CLSD_INTERNAL1, 0x00A1); - regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x0A05); - msleep(200); + regmap_write(regmap, RT1015_CLK_DET, 0x0000); regmap_write(regmap, RT1015_PWR4, 0x00B2); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0009); + msleep(100); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000A); + msleep(100); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000C); + msleep(100); regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2028); regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0140); - regmap_write(regmap, RT1015_PWR5, 0x0175); - regmap_write(regmap, RT1015_CLSD_INTERNAL1, 0x1721); - regmap_write(regmap, RT1015_CLASSD_SEQ, 0x570E); - regmap_write(regmap, RT1015_MIXER1, 0x203D); - regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5A01); - regmap_write(regmap, RT1015_CLSD_INTERNAL2, 0x12FF); - regmap_write(regmap, RT1015_GAT_BOOST, 0x0eFE); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008E); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0088); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000D); + msleep(300); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0008); regmap_write(regmap, RT1015_SYS_RST1, 0x05F5); - regmap_write(regmap, RT1015_SYS_RST2, 0x0b9a); regcache_cache_bypass(regmap, false); regcache_mark_dirty(regmap); -- 2.29.0

2 2

[PATCH 1/2] ASoC: SOF: Intel: hda: Modify existing helper to disable WAKEEN
by Kai-Heng Feng 04 Jan '21

04 Jan '21

Modify hda_codec_jack_wake_enable() to also support disable WAKEEN. This is a preparation for next patch. No functional change intended. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/soc/sof/intel/hda-codec.c | 16 +++++++--------- sound/soc/sof/intel/hda-dsp.c | 6 ++++-- sound/soc/sof/intel/hda.h | 2 +- 3 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 6875fa570c2c..bc9ac4abdab5 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -63,16 +63,18 @@ static int hda_codec_load_module(struct hda_codec *codec) } /* enable controller wake up event for all codecs with jack connectors */ -void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) +void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev, bool enable) { struct hda_bus *hbus = sof_to_hbus(sdev); struct hdac_bus *bus = sof_to_bus(sdev); struct hda_codec *codec; unsigned int mask = 0; - list_for_each_codec(codec, hbus) - if (codec->jacktbl.used) - mask |= BIT(codec->core.addr); + if (enable) { + list_for_each_codec(codec, hbus) + if (codec->jacktbl.used) + mask |= BIT(codec->core.addr); + } snd_hdac_chip_updatew(bus, WAKEEN, STATESTS_INT_MASK, mask); } @@ -81,12 +83,8 @@ void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) void hda_codec_jack_check(struct snd_sof_dev *sdev) { struct hda_bus *hbus = sof_to_hbus(sdev); - struct hdac_bus *bus = sof_to_bus(sdev); struct hda_codec *codec; - /* disable controller Wake Up event*/ - snd_hdac_chip_updatew(bus, WAKEEN, STATESTS_INT_MASK, 0); - list_for_each_codec(codec, hbus) /* * Wake up all jack-detecting codecs regardless whether an event @@ -97,7 +95,7 @@ void hda_codec_jack_check(struct snd_sof_dev *sdev) codec->jackpoll_interval); } #else -void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) {} +void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev, bool enable) {} void hda_codec_jack_check(struct snd_sof_dev *sdev) {} #endif /* CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC */ EXPORT_SYMBOL_NS(hda_codec_jack_wake_enable, SND_SOC_SOF_HDA_AUDIO_CODEC); diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 2b001151fe37..7d00107cf3b2 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -617,7 +617,7 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) if (runtime_suspend) - hda_codec_jack_wake_enable(sdev); + hda_codec_jack_wake_enable(sdev, true); /* power down all hda link */ snd_hdac_ext_bus_link_power_down_all(bus); @@ -683,8 +683,10 @@ static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume) #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) /* check jack status */ - if (runtime_resume) + if (runtime_resume) { + hda_codec_jack_wake_enable(sdev, false); hda_codec_jack_check(sdev); + } /* turn off the links that were off before suspend */ list_for_each_entry(hlink, &bus->hlink_list, list) { diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 9ec8ae0fd649..a3b6f3e9121c 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -650,7 +650,7 @@ void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev); */ void hda_codec_probe_bus(struct snd_sof_dev *sdev, bool hda_codec_use_common_hdmi); -void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev); +void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev, bool enable); void hda_codec_jack_check(struct snd_sof_dev *sdev); #endif /* CONFIG_SND_SOC_SOF_HDA */ -- 2.29.2

2 7

[PATCH] ASoC: rt1015: modify calibration sequence for better performance
by jack.yu＠realtek.com 04 Jan '21

04 Jan '21

From: Jack Yu <jack.yu(a)realtek.com> New calibration sequence to fix power issue in idle state. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 40 ++++++++++----------------------------- 1 file changed, 10 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 32e6bcf763d1..5fdf7bbfd375 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -497,40 +497,20 @@ static void rt1015_calibrate(struct rt1015_priv *rt1015) snd_soc_dapm_mutex_lock(&component->dapm); regcache_cache_bypass(regmap, true); - regmap_write(regmap, RT1015_PWR9, 0xAA60); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0089); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008A); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008C); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008D); - regmap_write(regmap, RT1015_PWR4, 0x80B2); - regmap_write(regmap, RT1015_CLASSD_SEQ, 0x5797); - regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2100); - regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0100); - regmap_write(regmap, RT1015_PWR5, 0x2175); - regmap_write(regmap, RT1015_MIXER1, 0x005D); - regmap_write(regmap, RT1015_CLSD_INTERNAL1, 0x00A1); - regmap_write(regmap, RT1015_CLSD_INTERNAL2, 0x12F7); - regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x1205); - msleep(200); - regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2000); - regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0180); - regmap_write(regmap, RT1015_CLSD_INTERNAL1, 0x00A1); - regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x0A05); - msleep(200); + regmap_write(regmap, RT1015_CLK_DET, 0x0000); regmap_write(regmap, RT1015_PWR4, 0x00B2); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0009); + msleep(100); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000A); + msleep(100); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000C); + msleep(100); regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2028); regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0140); - regmap_write(regmap, RT1015_PWR5, 0x0175); - regmap_write(regmap, RT1015_CLSD_INTERNAL1, 0x1721); - regmap_write(regmap, RT1015_CLASSD_SEQ, 0x570E); - regmap_write(regmap, RT1015_MIXER1, 0x203D); - regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5A01); - regmap_write(regmap, RT1015_CLSD_INTERNAL2, 0x12FF); - regmap_write(regmap, RT1015_GAT_BOOST, 0x0eFE); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008E); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0088); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000D); + msleep(300); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0008); regmap_write(regmap, RT1015_SYS_RST1, 0x05F5); - regmap_write(regmap, RT1015_SYS_RST2, 0x0b9a); regcache_cache_bypass(regmap, false); regcache_mark_dirty(regmap); -- 2.29.0

1 0

Re: [PATCH v2] ALSA: usb-audio: Add quirk for RC-505
by Takashi Iwai 03 Jan '21

03 Jan '21

On Sat, 02 Jan 2021 22:08:35 +0100, Timon Reinold wrote: > > BOSS RC-505 (shown by lsusb as "Roland Corp. RC-505") does require the > same quirk as these other BOSS devices. > > Without this quirk it is neither possible to capture audio from nor to > write audio to the RC-505. Both just result in an empty audio > stream. With these changes both capture and playback seem to work > quite fine. MIDI funtionality was not tested. > > Tested-by: Harry Reinold <harry.reinold(a)posteo.de> > Signed-off-by: Timon Reinold <tirei(a)agon.one> > --- > > I just noticed that these entries should be sorted by ID. Sorry for that. Thanks, applied now. Takashi

1 0

[PATCH] ALSA: usb-audio: Add quirk for RC-505
by Timon Reinold 02 Jan '21

02 Jan '21

BOSS RC-505 (shown by lsusb as "Roland Corp. RC-505") does require the same quirk as these other BOSS devices. Without this quirk it is neither possible to capture audio from nor to write audio to the RC-505. Both just result in an empty audio stream. With these changes both capture and playback seem to work quite fine. MIDI funtionality was not tested. Tested-by: Harry Reinold <harry.reinold(a)posteo.de> Signed-off-by: Timon Reinold <tirei(a)agon.one> --- We found the hint to enable the same quirk as reqired by the BOSS Katana for the RC-505 in this post: https://old.reddit.com/r/linuxaudio/comments/fom7z4/roland_rc505_usb_audio_… But none of the code posted there was used (since that predates implicit.c), so I think that shouldn't create any copyright problems. We're currently running the older version of this patch (without implicit.c) on Linux Mint and it seems quite stable so far. We did also test this version of the patch and that seems to work fine, too. Yours sincerely, Timon Reinold sound/usb/implicit.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 70b9777b2e63..2d7f747c0237 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -79,6 +79,7 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d8), /* BOSS Katana */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01e5), /* BOSS GT-001 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x0203), /* BOSS AD-10 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0171), /* BOSS RC-505 */ {} /* terminator */ }; @@ -91,6 +92,7 @@ static const struct snd_usb_implicit_fb_match capture_implicit_fb_quirks[] = { IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d8, 0x0d, 0x01), /* BOSS Katana */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01e5, 0x0d, 0x01), /* BOSS GT-001 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0203, 0x0d, 0x01), /* BOSS AD-10 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0171, 0x0d, 0x01), /* BOSS RC-505 */ {} /* terminator */ }; -- 2.29.2

1 0

Re: Haswell audio no longer working with new Catpt driver
by Lars-Peter Clausen 02 Jan '21

02 Jan '21

Hi, I'm taking this conversation back to the mailinglist. Hope you don't mind. On 12/31/20 2:15 PM, Christian Labisch wrote: > Hi Amadeusz, Greg, and Lars-Peter, > > Do you have a suggestion for a workaround ? > A special boot parameter that I could add ? > Will the drivers work in the next version ? I had a quick look at the changes to the HDA driver between v5.9 and v5.10. And there isn't that much, expect some changes to power management. Curiously enough pretty much your only difference between your v5.9 and v5.10 alsa-info.sh is that a few of the nodes are in D3 state (meaning powered down) in v5.10. One of them is the internal speaker. But this could also be a coincidence, would be interesting to see what the output of alsa-info.sh looks like when the you are trying to play some audio. If it is still powered down then there is a bug. - Lars > > Regards, > Christian > > >> On Thu, 2020-12-31 at 12:20 +0100, Amadeusz Sławiński wrote: >>> On 12/31/2020 11:50 AM, Christian Labisch wrote: >>>> Hi Lars-Peter, >>>> >>>> Thank you, please find attached the requested information from both kernels. >>>> I freshly installed the fedora kernel 5.10.4 to give you the latest results. >>>> >>>> Regards, >>>> Christian >>>> >>>> Christian Labisch >>>> Red Hat Accelerator >>>> clnetbox.blogspot.com >>>> access.redhat.com/community >>>> access.redhat.com/accelerators >>>> >>>> >>>> On Thu, 2020-12-31 at 11:04 +0100, Lars-Peter Clausen wrote: >>>>> On 12/31/20 9:33 AM, Greg Kroah-Hartman wrote: >>>>>> On Wed, Dec 30, 2020 at 07:10:16PM +0100, Christian Labisch wrote: >>>>>>> Update : >>>>>>> >>>>>>> I've just tested the kernel 5.10.4 from ELRepo. >>>>>>> Unfortunately nothing changed - still no sound. >>>>>> Ah, sad. Can you run 'git bisect' between 5.9 and 5.10 to determine the >>>>>> commit that caused the problem? >>>>> The problem is that one driver was replaced with another driver. git >>>>> bisect wont really help to narrow down why the new driver does not work. >>>>> >>>>> Christian can you run the alsa-info.sh[1] script on your system and send >>>>> back the result? >>>>> >>>>> You say sound is not working, can you clarify that a bit? Is no sound >>>>> card registered? Is it registered but output stays silent? >>>>> >>>>> - Lars >>>>> >>>>> [1] https://www.alsa-project.org/wiki/AlsaInfo >>>>> <https://www.alsa-project.org/wiki/AlsaInfo> >>>>> >>>>> >>> Hi, >>> >>> from reading provided files it seems that you use snd_intel_hda driver, >>> it should be possible to git bisect it between 5.9 and 5.10 as it wasn't >>> replaced. >>> >>> Catpt driver is used on machines using DSP. >>> >>> Amadeusz

4 8

[PATCH v2] ALSA: hda/hdmi: fix silent stream for first playback to DP
by Kai Vehmanen 02 Jan '21

02 Jan '21

A problem exists in enabling silent stream when connection type is DisplayPort. Silent stream programming is completed when a new DP receiver is connected, but infoframe transmission does not actually start until PCM is opened for the first time. This can result in audible gap of multiple seconds. This only affects the first PCM open. Fix the issue by properly assigning a converter to the silent stream, and modifying the required stream ID programming sequence. This change only affects Intel display audio codecs. BugLink: https://github.com/thesofproject/linux/issues/2468 Fixes: 951894cf30f4 ("ALSA: hda/hdmi: Add Intel silent stream support") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 98 +++++++++++++++++++++++++++++++++----- 1 file changed, 86 insertions(+), 12 deletions(-) v2 changes: - locking missing in silent_stream_disable() diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b0068f8ca46d..2ddc27db8c01 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -78,6 +78,7 @@ struct hdmi_spec_per_pin { int pcm_idx; /* which pcm is attached. -1 means no pcm is attached */ int repoll_count; bool setup; /* the stream has been set up by prepare callback */ + bool silent_stream; int channels; /* current number of channels */ bool non_pcm; bool chmap_set; /* channel-map override by ALSA API? */ @@ -979,6 +980,13 @@ static int hdmi_choose_cvt(struct hda_codec *codec, else per_pin = get_pin(spec, pin_idx); + if (per_pin && per_pin->silent_stream) { + cvt_idx = cvt_nid_to_cvt_index(codec, per_pin->cvt_nid); + if (cvt_id) + *cvt_id = cvt_idx; + return 0; + } + /* Dynamically assign converter to stream */ for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { per_cvt = get_cvt(spec, cvt_idx); @@ -1642,30 +1650,95 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, snd_hda_power_down_pm(codec); } +#define I915_SILENT_RATE 48000 +#define I915_SILENT_CHANNELS 2 +#define I915_SILENT_FORMAT SNDRV_PCM_FORMAT_S16_LE +#define I915_SILENT_FORMAT_BITS 16 +#define I915_SILENT_FMT_MASK 0xf + static void silent_stream_enable(struct hda_codec *codec, - struct hdmi_spec_per_pin *per_pin) + struct hdmi_spec_per_pin *per_pin) { - unsigned int newval, oldval; - - codec_dbg(codec, "hdmi: enabling silent stream for NID %d\n", - per_pin->pin_nid); + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_cvt *per_cvt; + int cvt_idx, pin_idx, err; + unsigned int format; mutex_lock(&per_pin->lock); - if (!per_pin->channels) - per_pin->channels = 2; + if (per_pin->setup) { + codec_dbg(codec, "hdmi: PCM already open, no silent stream\n"); + goto unlock_out; + } - oldval = snd_hda_codec_read(codec, per_pin->pin_nid, 0, - AC_VERB_GET_CONV, 0); - newval = (oldval & 0xF0) | 0xF; - snd_hda_codec_write(codec, per_pin->pin_nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, newval); + pin_idx = pin_id_to_pin_index(codec, per_pin->pin_nid, per_pin->dev_id); + err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx); + if (err) { + codec_err(codec, "hdmi: no free converter to enable silent mode\n"); + goto unlock_out; + } + + per_cvt = get_cvt(spec, cvt_idx); + per_cvt->assigned = 1; + per_pin->cvt_nid = per_cvt->cvt_nid; + per_pin->silent_stream = true; + codec_dbg(codec, "hdmi: enabling silent stream pin-NID=0x%x cvt-NID=0x%x\n", + per_pin->pin_nid, per_cvt->cvt_nid); + + snd_hda_set_dev_select(codec, per_pin->pin_nid, per_pin->dev_id); + snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, + AC_VERB_SET_CONNECT_SEL, + per_pin->mux_idx); + + /* configure unused pins to choose other converters */ + pin_cvt_fixup(codec, per_pin, 0); + + snd_hdac_sync_audio_rate(&codec->core, per_pin->pin_nid, + per_pin->dev_id, I915_SILENT_RATE); + + /* trigger silent stream generation in hw */ + format = snd_hdac_calc_stream_format(I915_SILENT_RATE, I915_SILENT_CHANNELS, + I915_SILENT_FORMAT, I915_SILENT_FORMAT_BITS, 0); + snd_hda_codec_setup_stream(codec, per_pin->cvt_nid, + I915_SILENT_FMT_MASK, I915_SILENT_FMT_MASK, format); + usleep_range(100, 200); + snd_hda_codec_setup_stream(codec, per_pin->cvt_nid, I915_SILENT_FMT_MASK, 0, format); + + per_pin->channels = I915_SILENT_CHANNELS; hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); + unlock_out: mutex_unlock(&per_pin->lock); } +static void silent_stream_disable(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin) +{ + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_cvt *per_cvt; + int cvt_idx; + + mutex_lock(&per_pin->lock); + if (!per_pin->silent_stream) + goto unlock_out; + + codec_dbg(codec, "HDMI: disable silent stream on pin-NID=0x%x cvt-NID=0x%x\n", + per_pin->pin_nid, per_pin->cvt_nid); + + cvt_idx = cvt_nid_to_cvt_index(codec, per_pin->cvt_nid); + if (cvt_idx >= 0 && cvt_idx < spec->num_cvts) { + per_cvt = get_cvt(spec, cvt_idx); + per_cvt->assigned = 0; + } + + per_pin->cvt_nid = 0; + per_pin->silent_stream = false; + + unlock_out: + mutex_unlock(&spec->pcm_lock); +} + /* update ELD and jack state via audio component */ static void sync_eld_via_acomp(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin) @@ -1701,6 +1774,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec, pm_ret); silent_stream_enable(codec, per_pin); } else if (monitor_prev && !monitor_next) { + silent_stream_disable(codec, per_pin); pm_ret = snd_hda_power_down_pm(codec); if (pm_ret < 0) codec_err(codec, -- 2.29.2

4 4

[PATCH] ALSA: hda/hdmi: Fix incorrect mutex unlock in silent_stream_disable()
by Takashi Iwai 01 Jan '21

01 Jan '21

The silent_stream_disable() function introduced by the commit b1a5039759cb ("ALSA: hda/hdmi: fix silent stream for first playback to DP") takes the per_pin->lock mutex, but it unlocks the wrong one, spec->pcm_lock, which causes a deadlock. This patch corrects it. Fixes: b1a5039759cb ("ALSA: hda/hdmi: fix silent stream for first playback to DP") Reported-by: Jan Alexander Steffens (heftig) <heftig(a)archlinux.org> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1e4a4b83fbf6..74d246a0dc6d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1733,7 +1733,7 @@ static void silent_stream_disable(struct hda_codec *codec, per_pin->silent_stream = false; unlock_out: - mutex_unlock(&spec->pcm_lock); + mutex_unlock(&per_pin->lock); } /* update ELD and jack state via audio component */ -- 2.26.2

2 1

[PATCH v2] docs: Fix reST markup when linking to sections
by Nícolas F. R. A. Prado 31 Dec '20

31 Dec '20

During the process of converting the documentation to reST, some links were converted using the following wrong syntax (and sometimes using %20 instead of spaces): `Display text <#section-name-in-html>`__ This syntax isn't valid according to the docutils' spec [1], but more importantly, it is specific to HTML, since it uses '#' to link to an HTML anchor. The right syntax would instead use a docutils hyperlink reference as the embedded URI to point to the section [2], that is: `Display text <Section Name_>`__ This syntax works in both HTML and PDF. The LaTeX toolchain doesn't mind the HTML anchor syntax when generating the pdf documentation (make pdfdocs), that is, the build succeeds but the links don't work, but that syntax causes errors when trying to build using the not-yet-merged rst2pdf: ValueError: format not resolved, probably missing URL scheme or undefined destination target for 'Forcing%20Quiescent%20States' So, use the correct syntax in order to have it work in all different output formats. [1]: https://docutils.sourceforge.io/docs/ref/rst/restructuredtext.html#referenc… [2]: https://docutils.sourceforge.io/docs/ref/rst/restructuredtext.html#embedded… Fixes: ccc9971e2147 ("docs: rcu: convert some articles from html to ReST") Fixes: c8cce10a62aa ("docs: Fix the reference labels in Locking.rst") Fixes: e548cdeffcd8 ("docs-rst: convert kernel-locking to ReST") Fixes: 7ddedebb03b7 ("ALSA: doc: ReSTize writing-an-alsa-driver document") Signed-off-by: Nícolas F. R. A. Prado <nfraprado(a)protonmail.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> --- Changes in v2: - Thanks to Mauro: - Simplify the syntax of some links by taking advantage of docutils' case-insensitivity when dealing with references. .../Tree-RCU-Memory-Ordering.rst | 8 ++++---- .../RCU/Design/Requirements/Requirements.rst | 20 +++++++++---------- Documentation/kernel-hacking/locking.rst | 8 ++++---- .../kernel-api/writing-an-alsa-driver.rst | 16 +++++++-------- 4 files changed, 26 insertions(+), 26 deletions(-) diff --git a/Documentation/RCU/Design/Memory-Ordering/Tree-RCU-Memory-Ordering.rst b/Documentation/RCU/Design/Memory-Ordering/Tree-RCU-Memory-Ordering.rst index e44cfcb7adcc..e57927427786 100644 --- a/Documentation/RCU/Design/Memory-Ordering/Tree-RCU-Memory-Ordering.rst +++ b/Documentation/RCU/Design/Memory-Ordering/Tree-RCU-Memory-Ordering.rst @@ -473,7 +473,7 @@ read-side critical sections that follow the idle period (the oval near the bottom of the diagram above). Plumbing this into the full grace-period execution is described -`below <#Forcing%20Quiescent%20States>`__. +`below <Forcing Quiescent States_>`__. CPU-Hotplug Interface ^^^^^^^^^^^^^^^^^^^^^ @@ -494,7 +494,7 @@ mask to detect CPUs having gone offline since the beginning of this grace period. Plumbing this into the full grace-period execution is described -`below <#Forcing%20Quiescent%20States>`__. +`below <Forcing Quiescent States_>`__. Forcing Quiescent States ^^^^^^^^^^^^^^^^^^^^^^^^ @@ -532,7 +532,7 @@ from other CPUs. | RCU. But this diagram is complex enough as it is, so simplicity | | overrode accuracy. You can think of it as poetic license, or you can | | think of it as misdirection that is resolved in the | -| `stitched-together diagram <#Putting%20It%20All%20Together>`__. | +| `stitched-together diagram <Putting It All Together_>`__. | +-----------------------------------------------------------------------+ Grace-Period Cleanup @@ -596,7 +596,7 @@ maintain ordering. For example, if the callback function wakes up a task that runs on some other CPU, proper ordering must in place in both the callback function and the task being awakened. To see why this is important, consider the top half of the `grace-period -cleanup <#Grace-Period%20Cleanup>`__ diagram. The callback might be +cleanup`_ diagram. The callback might be running on a CPU corresponding to the leftmost leaf ``rcu_node`` structure, and awaken a task that is to run on a CPU corresponding to the rightmost leaf ``rcu_node`` structure, and the grace-period kernel diff --git a/Documentation/RCU/Design/Requirements/Requirements.rst b/Documentation/RCU/Design/Requirements/Requirements.rst index 1ae79a10a8de..ce1075f040be 100644 --- a/Documentation/RCU/Design/Requirements/Requirements.rst +++ b/Documentation/RCU/Design/Requirements/Requirements.rst @@ -45,7 +45,7 @@ requirements: #. `Other RCU Flavors`_ #. `Possible Future Changes`_ -This is followed by a `summary <#Summary>`__, however, the answers to +This is followed by a summary_, however, the answers to each quick quiz immediately follows the quiz. Select the big white space with your mouse to see the answer. @@ -1096,7 +1096,7 @@ memory barriers. | case, voluntary context switch) within an RCU read-side critical | | section. However, sleeping locks may be used within userspace RCU | | read-side critical sections, and also within Linux-kernel sleepable | -| RCU `(SRCU) <#Sleepable%20RCU>`__ read-side critical sections. In | +| RCU `(SRCU) <Sleepable RCU_>`__ read-side critical sections. In | | addition, the -rt patchset turns spinlocks into a sleeping locks so | | that the corresponding critical sections can be preempted, which also | | means that these sleeplockified spinlocks (but not other sleeping | @@ -1186,7 +1186,7 @@ non-preemptible (``CONFIG_PREEMPT=n``) kernels, and thus `tiny RCU <https://lkml.kernel.org/g/20090113221724.GA15307@linux.vnet.ibm.com>`__ was born. Josh Triplett has since taken over the small-memory banner with his `Linux kernel tinification <https://tiny.wiki.kernel.org/>`__ -project, which resulted in `SRCU <#Sleepable%20RCU>`__ becoming optional +project, which resulted in `SRCU <Sleepable RCU_>`__ becoming optional for those kernels not needing it. The remaining performance requirements are, for the most part, @@ -1457,8 +1457,8 @@ will vary as the value of ``HZ`` varies, and can also be changed using the relevant Kconfig options and kernel boot parameters. RCU currently does not do much sanity checking of these parameters, so please use caution when changing them. Note that these forward-progress measures -are provided only for RCU, not for `SRCU <#Sleepable%20RCU>`__ or `Tasks -RCU <#Tasks%20RCU>`__. +are provided only for RCU, not for `SRCU <Sleepable RCU_>`__ or `Tasks +RCU`_. RCU takes the following steps in ``call_rcu()`` to encourage timely invocation of callbacks when any given non-\ ``rcu_nocbs`` CPU has @@ -1477,8 +1477,8 @@ encouragement was provided: Again, these are default values when running at ``HZ=1000``, and can be overridden. Again, these forward-progress measures are provided only for -RCU, not for `SRCU <#Sleepable%20RCU>`__ or `Tasks -RCU <#Tasks%20RCU>`__. Even for RCU, callback-invocation forward +RCU, not for `SRCU <Sleepable RCU_>`__ or `Tasks +RCU`_. Even for RCU, callback-invocation forward progress for ``rcu_nocbs`` CPUs is much less well-developed, in part because workloads benefiting from ``rcu_nocbs`` CPUs tend to invoke ``call_rcu()`` relatively infrequently. If workloads emerge that need @@ -1920,7 +1920,7 @@ Hotplug CPU The Linux kernel supports CPU hotplug, which means that CPUs can come and go. It is of course illegal to use any RCU API member from an -offline CPU, with the exception of `SRCU <#Sleepable%20RCU>`__ read-side +offline CPU, with the exception of `SRCU <Sleepable RCU_>`__ read-side critical sections. This requirement was present from day one in DYNIX/ptx, but on the other hand, the Linux kernel's CPU-hotplug implementation is “interesting.” @@ -2147,7 +2147,7 @@ handles these states differently: However, RCU must be reliably informed as to whether any given CPU is currently in the idle loop, and, for ``NO_HZ_FULL``, also whether that CPU is executing in usermode, as discussed -`earlier <#Energy%20Efficiency>`__. It also requires that the +`earlier <Energy Efficiency_>`__. It also requires that the scheduling-clock interrupt be enabled when RCU needs it to be: #. If a CPU is either idle or executing in usermode, and RCU believes it @@ -2264,7 +2264,7 @@ Performance, Scalability, Response Time, and Reliability ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Expanding on the `earlier -discussion <#Performance%20and%20Scalability>`__, RCU is used heavily by +discussion <Performance and Scalability_>`__, RCU is used heavily by hot code paths in performance-critical portions of the Linux kernel's networking, security, virtualization, and scheduling code paths. RCU must therefore use efficient implementations, especially in its diff --git a/Documentation/kernel-hacking/locking.rst b/Documentation/kernel-hacking/locking.rst index 6ed806e6061b..c3448929a824 100644 --- a/Documentation/kernel-hacking/locking.rst +++ b/Documentation/kernel-hacking/locking.rst @@ -118,11 +118,11 @@ spinlock, but you may block holding a mutex. If you can't lock a mutex, your task will suspend itself, and be woken up when the mutex is released. This means the CPU can do something else while you are waiting. There are many cases when you simply can't sleep (see -`What Functions Are Safe To Call From Interrupts? <#sleeping-things>`__), +`What Functions Are Safe To Call From Interrupts?`_), and so have to use a spinlock instead. Neither type of lock is recursive: see -`Deadlock: Simple and Advanced <#deadlock>`__. +`Deadlock: Simple and Advanced`_. Locks and Uniprocessor Kernels ------------------------------ @@ -179,7 +179,7 @@ perfect world). Note that you can also use spin_lock_irq() or spin_lock_irqsave() here, which stop hardware interrupts -as well: see `Hard IRQ Context <#hard-irq-context>`__. +as well: see `Hard IRQ Context`_. This works perfectly for UP as well: the spin lock vanishes, and this macro simply becomes local_bh_disable() @@ -230,7 +230,7 @@ The Same Softirq ~~~~~~~~~~~~~~~~ The same softirq can run on the other CPUs: you can use a per-CPU array -(see `Per-CPU Data <#per-cpu-data>`__) for better performance. If you're +(see `Per-CPU Data`_) for better performance. If you're going so far as to use a softirq, you probably care about scalable performance enough to justify the extra complexity. diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 73bbd59afc33..e6365836fa8b 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -71,7 +71,7 @@ core/oss The codes for PCM and mixer OSS emulation modules are stored in this directory. The rawmidi OSS emulation is included in the ALSA rawmidi code since it's quite small. The sequencer code is stored in -``core/seq/oss`` directory (see `below <#core-seq-oss>`__). +``core/seq/oss`` directory (see `below <core/seq/oss_>`__). core/seq ~~~~~~~~ @@ -382,7 +382,7 @@ where ``enable[dev]`` is the module option. Each time the ``probe`` callback is called, check the availability of the device. If not available, simply increment the device index and returns. dev will be incremented also later (`step 7 -<#set-the-pci-driver-data-and-return-zero>`__). +<7) Set the PCI driver data and return zero._>`__). 2) Create a card instance ~~~~~~~~~~~~~~~~~~~~~~~~~ @@ -450,10 +450,10 @@ field contains the information shown in ``/proc/asound/cards``. 5) Create other components, such as mixer, MIDI, etc. ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ -Here you define the basic components such as `PCM <#PCM-Interface>`__, -mixer (e.g. `AC97 <#API-for-AC97-Codec>`__), MIDI (e.g. -`MPU-401 <#MIDI-MPU401-UART-Interface>`__), and other interfaces. -Also, if you want a `proc file <#Proc-Interface>`__, define it here, +Here you define the basic components such as `PCM <PCM Interface_>`__, +mixer (e.g. `AC97 <API for AC97 Codec_>`__), MIDI (e.g. +`MPU-401 <MIDI (MPU401-UART) Interface_>`__), and other interfaces. +Also, if you want a `proc file <Proc Interface_>`__, define it here, too. 6) Register the card instance. @@ -941,7 +941,7 @@ The allocation of an interrupt source is done like this: chip->irq = pci->irq; where :c:func:`snd_mychip_interrupt()` is the interrupt handler -defined `later <#pcm-interface-interrupt-handler>`__. Note that +defined `later <PCM Interrupt Handler_>`__. Note that ``chip->irq`` should be defined only when :c:func:`request_irq()` succeeded. @@ -3104,7 +3104,7 @@ processing the output stream in the irq handler. If the MPU-401 interface shares its interrupt with the other logical devices on the card, set ``MPU401_INFO_IRQ_HOOK`` (see -`below <#MIDI-Interrupt-Handler>`__). +`below <MIDI Interrupt Handler_>`__). Usually, the port address corresponds to the command port and port + 1 corresponds to the data port. If not, you may change the ``cport`` -- 2.29.2

3 2

[PATCH] ASoC: fsl: fix -Wmaybe-uninitialized warning
by Arnd Bergmann 31 Dec '20

31 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> Clang points out a code path that returns an undefined value in an error case: sound/soc/fsl/imx-hdmi.c:165:6: error: variable 'ret' is used uninitialized whenever 'if' condition is true [-Werror,-Wsom etimes-uninitialized] if ((hdmi_out && hdmi_in) || (!hdmi_out && !hdmi_in)) { ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/fsl/imx-hdmi.c:212:9: note: uninitialized use occurs here return ret; The driver returns -EINVAL for other broken DT properties, so do it the same way here. Fixes: 6a5f850aa83a ("ASoC: fsl: Add imx-hdmi machine driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/fsl/imx-hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index 2c2a76a71940..ede4a9ad1054 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -164,6 +164,7 @@ static int imx_hdmi_probe(struct platform_device *pdev) if ((hdmi_out && hdmi_in) || (!hdmi_out && !hdmi_in)) { dev_err(&pdev->dev, "Invalid HDMI DAI link\n"); + ret = -EINVAL; goto fail; } -- 2.29.2

3 2

[PATCH] ASoC: rt5645: Remove the redundant delay time
by Oder Chiou 31 Dec '20

31 Dec '20

This patch removes the redundant delay time to speed up from the cold start of headphone playback. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 33 ++++++++------------------------- 1 file changed, 8 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 420003d062c7..1c9e61abda30 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -435,7 +435,6 @@ struct rt5645_priv { int jack_type; bool en_button_func; - bool hp_on; int v_id; }; @@ -1645,6 +1644,7 @@ static void hp_amp_power(struct snd_soc_component *component, int on) { static int hp_amp_power_count; struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + int i, val; if (on) { if (hp_amp_power_count <= 0) { @@ -1655,7 +1655,13 @@ static void hp_amp_power(struct snd_soc_component *component, int on) snd_soc_component_write(component, RT5645_DEPOP_M1, 0x000d); regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_HP_DCC_INT1, 0x9f01); - msleep(20); + for (i = 0; i < 20; i++) { + usleep_range(1000, 1500); + regmap_read(rt5645->regmap, RT5645_PR_BASE + + RT5645_HP_DCC_INT1, &val); + if (!(val & 0x8000)) + break; + } snd_soc_component_update_bits(component, RT5645_DEPOP_M1, RT5645_HP_CO_MASK, RT5645_HP_CO_EN); regmap_write(rt5645->regmap, RT5645_PR_BASE + @@ -1665,7 +1671,6 @@ static void hp_amp_power(struct snd_soc_component *component, int on) RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_component_write(component, RT5645_DEPOP_M2, 0x1140); msleep(90); - rt5645->hp_on = true; } else { /* depop parameters */ snd_soc_component_update_bits(component, RT5645_DEPOP_M2, @@ -1885,27 +1890,6 @@ static int rt5645_bst2_event(struct snd_soc_dapm_widget *w, return 0; } -static int rt5650_hp_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - if (rt5645->hp_on) { - msleep(100); - rt5645->hp_on = false; - } - break; - - default: - return 0; - } - - return 0; -} - static int rt5645_set_micbias1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { @@ -2242,7 +2226,6 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("PDM1R"), SND_SOC_DAPM_OUTPUT("SPOL"), SND_SOC_DAPM_OUTPUT("SPOR"), - SND_SOC_DAPM_POST("DAPM_POST", rt5650_hp_event), }; static const struct snd_soc_dapm_widget rt5645_specific_dapm_widgets[] = { -- 2.29.0

2 1

[PATCH] ASoC: mediatek: add MTK_PMIC_WRAP dependency
by Arnd Bergmann 31 Dec '20

31 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> Randconfig builds often show harmless warnings like WARNING: unmet direct dependencies detected for SND_SOC_MT6359 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && MTK_PMIC_WRAP [=n] Selected by [y]: - SND_SOC_MT8192_MT6359_RT1015_RT5682 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=y] && SND_SOC_MT8192 [=y] Add a dependency to avoid that. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/mediatek/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 8d3dcfb6a580..effdb76369e4 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -172,7 +172,7 @@ config SND_SOC_MT8192 config SND_SOC_MT8192_MT6359_RT1015_RT5682 tristate "ASoC Audio driver for MT8192 with MT6359 RT1015 RT5682 codec" depends on I2C - depends on SND_SOC_MT8192 + depends on SND_SOC_MT8192 && MTK_PMIC_WRAP select SND_SOC_MT6359 select SND_SOC_RT1015 select SND_SOC_RT1015P -- 2.29.2

3 2

[PATCH 1/2] ASoC: wm_adsp: Only use __be32 for big-endian data
by Richard Fitzgerald 31 Dec '20

31 Dec '20

This fixes some minor cases where u32 or unsigned int types were used to store big-endian data, and __be32 types used to store both big-endian and cpu-endian data. This was producing sparse warnings. Most cases resulted from using the same variable to hold the big-endian value and its converted cpu-endian value. These can be simply fixed by introducing another local variable, or avoiding storing the intermediate value back into the original variable. One special case is the raw_buf used in the compressed streams to transfer data from DSP to user-side. The endian conversion happens in-place (as there's no point introducing another buffer) so a cast to (__be32 *) is added when passing it to wm_adsp_read_raw_data_block(). Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 33 ++++++++++++++++----------------- 1 file changed, 16 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5b7d81a91df3..8cfa8ac1b8c4 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -980,7 +980,7 @@ static int wm_coeff_write_acked_control(struct wm_coeff_ctl *ctl, unsigned int event_id) { struct wm_adsp *dsp = ctl->dsp; - u32 val = cpu_to_be32(event_id); + __be32 val = cpu_to_be32(event_id); unsigned int reg; int i, ret; @@ -3704,6 +3704,7 @@ static int wm_adsp_write_data_word(struct wm_adsp *dsp, int mem_type, unsigned int mem_addr, u32 data) { struct wm_adsp_region const *mem = wm_adsp_find_region(dsp, mem_type); + __be32 val = cpu_to_be32(data & 0x00ffffffu); unsigned int reg; if (!mem) @@ -3711,9 +3712,7 @@ static int wm_adsp_write_data_word(struct wm_adsp *dsp, int mem_type, reg = dsp->ops->region_to_reg(mem, mem_addr); - data = cpu_to_be32(data & 0x00ffffffu); - - return regmap_raw_write(dsp->regmap, reg, &data, sizeof(data)); + return regmap_raw_write(dsp->regmap, reg, &val, sizeof(val)); } static inline int wm_adsp_buffer_read(struct wm_adsp_compr_buf *buf, @@ -3870,7 +3869,8 @@ static int wm_adsp_buffer_parse_coeff(struct wm_coeff_ctl *ctl) { struct wm_adsp_host_buf_coeff_v1 coeff_v1; struct wm_adsp_compr_buf *buf; - unsigned int val, reg; + unsigned int reg, version; + __be32 bufp; int ret, i; ret = wm_coeff_base_reg(ctl, &reg); @@ -3878,17 +3878,17 @@ static int wm_adsp_buffer_parse_coeff(struct wm_coeff_ctl *ctl) return ret; for (i = 0; i < 5; ++i) { - ret = regmap_raw_read(ctl->dsp->regmap, reg, &val, sizeof(val)); + ret = regmap_raw_read(ctl->dsp->regmap, reg, &bufp, sizeof(bufp)); if (ret < 0) return ret; - if (val) + if (bufp) break; usleep_range(1000, 2000); } - if (!val) { + if (!bufp) { adsp_err(ctl->dsp, "Failed to acquire host buffer\n"); return -EIO; } @@ -3898,7 +3898,7 @@ static int wm_adsp_buffer_parse_coeff(struct wm_coeff_ctl *ctl) return -ENOMEM; buf->host_buf_mem_type = ctl->alg_region.type; - buf->host_buf_ptr = be32_to_cpu(val); + buf->host_buf_ptr = be32_to_cpu(bufp); ret = wm_adsp_buffer_populate(buf); if (ret < 0) @@ -3918,14 +3918,13 @@ static int wm_adsp_buffer_parse_coeff(struct wm_coeff_ctl *ctl) if (ret < 0) return ret; - coeff_v1.versions = be32_to_cpu(coeff_v1.versions); - val = coeff_v1.versions & HOST_BUF_COEFF_COMPAT_VER_MASK; - val >>= HOST_BUF_COEFF_COMPAT_VER_SHIFT; + version = be32_to_cpu(coeff_v1.versions) & HOST_BUF_COEFF_COMPAT_VER_MASK; + version >>= HOST_BUF_COEFF_COMPAT_VER_SHIFT; - if (val > HOST_BUF_COEFF_SUPPORTED_COMPAT_VER) { + if (version > HOST_BUF_COEFF_SUPPORTED_COMPAT_VER) { adsp_err(ctl->dsp, "Host buffer coeff ver %u > supported version %u\n", - val, HOST_BUF_COEFF_SUPPORTED_COMPAT_VER); + version, HOST_BUF_COEFF_SUPPORTED_COMPAT_VER); return -EINVAL; } @@ -3935,9 +3934,9 @@ static int wm_adsp_buffer_parse_coeff(struct wm_coeff_ctl *ctl) (char *)&coeff_v1.name); compr_dbg(buf, "host_buf_ptr=%x coeff version %u\n", - buf->host_buf_ptr, val); + buf->host_buf_ptr, version); - return val; + return version; } static int wm_adsp_buffer_init(struct wm_adsp *dsp) @@ -4269,7 +4268,7 @@ static int wm_adsp_buffer_capture_block(struct wm_adsp_compr *compr, int target) /* Read data from DSP */ ret = wm_adsp_read_raw_data_block(buf->dsp, mem_type, adsp_addr, - nwords, compr->raw_buf); + nwords, (__be32 *)compr->raw_buf); if (ret < 0) return ret; -- 2.20.1

2 2

Re: Haswell audio no longer working with new Catpt driver (was: sound)
by Lars-Peter Clausen 31 Dec '20

31 Dec '20

On 12/31/20 9:33 AM, Greg Kroah-Hartman wrote: > On Wed, Dec 30, 2020 at 07:10:16PM +0100, Christian Labisch wrote: >> Update : >> >> I've just tested the kernel 5.10.4 from ELRepo. >> Unfortunately nothing changed - still no sound. > Ah, sad. Can you run 'git bisect' between 5.9 and 5.10 to determine the > commit that caused the problem? The problem is that one driver was replaced with another driver. git bisect wont really help to narrow down why the new driver does not work. Christian can you run the alsa-info.sh[1] script on your system and send back the result? You say sound is not working, can you clarify that a bit? Is no sound card registered? Is it registered but output stays silent? - Lars [1] https://www.alsa-project.org/wiki/AlsaInfo <https://www.alsa-project.org/wiki/AlsaInfo>

3 4

[PATCH] ALSA: hda/realtek: Enable mute and micmute LED on HP EliteBook 850 G7
by Kai-Heng Feng 31 Dec '20

31 Dec '20

HP EliteBook 850 G7 uses the same GPIO pins as ALC285_FIXUP_HP_GPIO_LED to enable mute and micmute LED. So apply the quirk to enable the LEDs. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- 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 dde5ba209541..b12e1f083029 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7959,6 +7959,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED), -- 2.29.2

2 1

add two intel reference SSID in the ALC256
by Pshou 31 Dec '20

31 Dec '20

Hi Alsa-devel ML: Can you help me add some source code to patch_realtek.c? Add two intel reference board SSID in the alc256. Best regards Pshou

2 1

[PATCH] ALSA: hda/realtek: Add mute LED quirk for more HP laptops
by Manuel Jiménez 31 Dec '20

31 Dec '20

HP Pavilion 13-bb0000 (SSID 103c:87c8) needs the same quirk as other models with ALC287. Signed-off-by: Manuel Jiménez <mjbfm99(a)me.com> --- 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 dde5ba209541..b77cef72c2d5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7964,6 +7964,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED), + 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(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), -- 2.29.2

2 1

[PATCH] sound: pci: hda: add a new hda codec
by bo.liu＠senarytech.com 31 Dec '20

31 Dec '20

From: bo liu <bo.liu(a)senarytech.com> The current kernel does not support the cx11970 codec chip. Add a codec configuration item to kernel. Signed-off-by: bo liu <bo.liu(a)senarytech.com> diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index be5000dd1585..e80fe8ba537c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1070,6 +1070,7 @@ static int patch_conexant_auto(struct hda_codec *codec) static const struct hda_device_id snd_hda_id_conexant[] = { HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f120D0, "CX11970", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto), --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index be5000dd1585..e80fe8ba537c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1070,6 +1070,7 @@ static int patch_conexant_auto(struct hda_codec *codec) static const struct hda_device_id snd_hda_id_conexant[] = { HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f120D0, "CX11970", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto), -- 2.25.1

2 1

[PATCH AUTOSEL 4.14 7/8] ALSA: pcm: Clear the full allocated memory at hw_params
by Sasha Levin 30 Dec '20

30 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 618de0f4ef11acd8cf26902e65493d46cc20cc89 ] The PCM hw_params core function tries to clear up the PCM buffer before actually using for avoiding the information leak from the previous usages or the usage before a new allocation. It performs the memset() with runtime->dma_bytes, but this might still leave some remaining bytes untouched; namely, the PCM buffer size is aligned in page size for mmap, hence runtime->dma_bytes doesn't necessarily cover all PCM buffer pages, and the remaining bytes are exposed via mmap. This patch changes the memory clearance to cover the all buffer pages if the stream is supposed to be mmap-ready (that guarantees that the buffer size is aligned in page size). Reviewed-by: Lars-Peter Clausen <lars(a)metafoo.de> Link: https://lore.kernel.org/r/20201218145625.2045-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/pcm_native.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 071e09c3d8557..c78db361cbbaa 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -721,8 +721,13 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->boundary *= 2; /* clear the buffer for avoiding possible kernel info leaks */ - if (runtime->dma_area && !substream->ops->copy_user) - memset(runtime->dma_area, 0, runtime->dma_bytes); + if (runtime->dma_area && !substream->ops->copy_user) { + size_t size = runtime->dma_bytes; + + if (runtime->info & SNDRV_PCM_INFO_MMAP) + size = PAGE_ALIGN(size); + memset(runtime->dma_area, 0, size); + } snd_pcm_timer_resolution_change(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP); -- 2.27.0

1 0

[PATCH AUTOSEL 4.19 09/10] ALSA: pcm: Clear the full allocated memory at hw_params
by Sasha Levin 30 Dec '20

30 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 618de0f4ef11acd8cf26902e65493d46cc20cc89 ] The PCM hw_params core function tries to clear up the PCM buffer before actually using for avoiding the information leak from the previous usages or the usage before a new allocation. It performs the memset() with runtime->dma_bytes, but this might still leave some remaining bytes untouched; namely, the PCM buffer size is aligned in page size for mmap, hence runtime->dma_bytes doesn't necessarily cover all PCM buffer pages, and the remaining bytes are exposed via mmap. This patch changes the memory clearance to cover the all buffer pages if the stream is supposed to be mmap-ready (that guarantees that the buffer size is aligned in page size). Reviewed-by: Lars-Peter Clausen <lars(a)metafoo.de> Link: https://lore.kernel.org/r/20201218145625.2045-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/pcm_native.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 7c12b0deb4eb5..db62dbe7eaa8a 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -753,8 +753,13 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->boundary *= 2; /* clear the buffer for avoiding possible kernel info leaks */ - if (runtime->dma_area && !substream->ops->copy_user) - memset(runtime->dma_area, 0, runtime->dma_bytes); + if (runtime->dma_area && !substream->ops->copy_user) { + size_t size = runtime->dma_bytes; + + if (runtime->info & SNDRV_PCM_INFO_MMAP) + size = PAGE_ALIGN(size); + memset(runtime->dma_area, 0, size); + } snd_pcm_timer_resolution_change(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP); -- 2.27.0

1 0

[PATCH AUTOSEL 5.4 16/17] ALSA: pcm: Clear the full allocated memory at hw_params
by Sasha Levin 30 Dec '20

30 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 618de0f4ef11acd8cf26902e65493d46cc20cc89 ] The PCM hw_params core function tries to clear up the PCM buffer before actually using for avoiding the information leak from the previous usages or the usage before a new allocation. It performs the memset() with runtime->dma_bytes, but this might still leave some remaining bytes untouched; namely, the PCM buffer size is aligned in page size for mmap, hence runtime->dma_bytes doesn't necessarily cover all PCM buffer pages, and the remaining bytes are exposed via mmap. This patch changes the memory clearance to cover the all buffer pages if the stream is supposed to be mmap-ready (that guarantees that the buffer size is aligned in page size). Reviewed-by: Lars-Peter Clausen <lars(a)metafoo.de> Link: https://lore.kernel.org/r/20201218145625.2045-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/pcm_native.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ec501fbaabe49..0c5b7a54ca81c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -717,8 +717,13 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->boundary *= 2; /* clear the buffer for avoiding possible kernel info leaks */ - if (runtime->dma_area && !substream->ops->copy_user) - memset(runtime->dma_area, 0, runtime->dma_bytes); + if (runtime->dma_area && !substream->ops->copy_user) { + size_t size = runtime->dma_bytes; + + if (runtime->info & SNDRV_PCM_INFO_MMAP) + size = PAGE_ALIGN(size); + memset(runtime->dma_area, 0, size); + } snd_pcm_timer_resolution_change(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP); -- 2.27.0

1 0

[PATCH AUTOSEL 5.10 28/31] ALSA: pcm: Clear the full allocated memory at hw_params
by Sasha Levin 30 Dec '20

30 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 618de0f4ef11acd8cf26902e65493d46cc20cc89 ] The PCM hw_params core function tries to clear up the PCM buffer before actually using for avoiding the information leak from the previous usages or the usage before a new allocation. It performs the memset() with runtime->dma_bytes, but this might still leave some remaining bytes untouched; namely, the PCM buffer size is aligned in page size for mmap, hence runtime->dma_bytes doesn't necessarily cover all PCM buffer pages, and the remaining bytes are exposed via mmap. This patch changes the memory clearance to cover the all buffer pages if the stream is supposed to be mmap-ready (that guarantees that the buffer size is aligned in page size). Reviewed-by: Lars-Peter Clausen <lars(a)metafoo.de> Link: https://lore.kernel.org/r/20201218145625.2045-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/pcm_native.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 47b155a49226f..9f3f8e953ff04 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -755,8 +755,13 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->boundary *= 2; /* clear the buffer for avoiding possible kernel info leaks */ - if (runtime->dma_area && !substream->ops->copy_user) - memset(runtime->dma_area, 0, runtime->dma_bytes); + if (runtime->dma_area && !substream->ops->copy_user) { + size_t size = runtime->dma_bytes; + + if (runtime->info & SNDRV_PCM_INFO_MMAP) + size = PAGE_ALIGN(size); + memset(runtime->dma_area, 0, size); + } snd_pcm_timer_resolution_change(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP); -- 2.27.0

1 0

[PATCH] ALSA: hda/realtek: Add mute LED quirk for more HP laptops
by Manuel Jiménez 30 Dec '20

30 Dec '20

HP Pavilion 13-bb0000 (SSID 103c:87c8) needs the same quirk as other models with ALC287. Signed-off-by: Manuel Jiménez <mjbfm99(a)me.com> --- 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 dde5ba209541..b77cef72c2d5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7964,6 +7964,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED), + 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(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), -- 2.29.2

1 0

configuring usb 2.0 audio gadget correctly via configfs?
by Bert Schiettecatte 30 Dec '20

30 Dec '20

Hi all I am using kernel 5.9 stable and I'm trying to configure the usb 2.0 audio gadget correctly so my device shows up as an audio interface when I plug it into my PC or Mac. I previously was on 4.11 and had a bunch of hacks I applied to the uac2 driver which worked fine, but a lot has changed since then and I'm wondering if by now the gadget works correctly out of the box, so I don't have to apply my hacks anymore. I'm configuring the device using configfs and that works fine, but when I plug it into my Mac I just see zero channels and the device name is capture or playback inactive. Anyway, just wanted to know what the current state of things is regarding the gadget driver and alsa. Thanks Bert

2 1

[PATCH 0/5] ASoC: rt1015: apply some refactors
by Tzung-Bi Shih 29 Dec '20

29 Dec '20

The series refactors rt1015.c. The 1st patch sorts header inclusions alphabetically. The 2nd and 3rd patch improve error handling for kcontrols. The 4th patch enhances readability. The 5th patch removes unused variables in rt1015_priv. Tzung-Bi Shih (5): 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 sound/soc/codecs/rt1015.c | 49 ++++++++++++++++++++------------------- sound/soc/codecs/rt1015.h | 4 ---- 2 files changed, 25 insertions(+), 28 deletions(-) -- 2.29.2.729.g45daf8777d-goog

2 6

[PATCH] ASoC: Intel: common: Fix some typos
by Christophe JAILLET 29 Dec '20

29 Dec '20

s/apci/acpi/ Turn an ICL into a TGL because it is likely a cut'n'paste error Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/intel/common/soc-acpi-intel-ehl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-ehl-match.c b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c index badafc1d54d2..5c7a1214403d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ehl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only /* - * soc-apci-intel-ehl-match.c - tables and support for EHL ACPI enumeration. + * soc-acpi-intel-ehl-match.c - tables and support for EHL ACPI enumeration. * * Copyright (c) 2019, Intel Corporation. * diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 98196e9fad62..4933184df5f5 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only /* - * soc-apci-intel-tgl-match.c - tables and support for ICL ACPI enumeration. + * soc-acpi-intel-tgl-match.c - tables and support for TGL ACPI enumeration. * * Copyright (c) 2019, Intel Corporation. * -- 2.27.0

2 1

[PATCH] ALSA: usb-audio: Add quirk for BOSS AD-10
by Takashi Iwai 29 Dec '20

29 Dec '20

BOSS AD-10 requires the very same quirk like other BOSS devices to enable the special implicit feedback mode. Reported-and-tested-by: Martin Passing <martin(a)passing.name> 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 eb3a4c433c3e..70b9777b2e63 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -78,6 +78,7 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d6), /* BOSS GT-1 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d8), /* BOSS Katana */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01e5), /* BOSS GT-001 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0203), /* BOSS AD-10 */ {} /* terminator */ }; @@ -89,6 +90,7 @@ static const struct snd_usb_implicit_fb_match capture_implicit_fb_quirks[] = { IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d6, 0x0d, 0x01), /* BOSS GT-1 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d8, 0x0d, 0x01), /* BOSS Katana */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01e5, 0x0d, 0x01), /* BOSS GT-001 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0203, 0x0d, 0x01), /* BOSS AD-10 */ {} /* terminator */ }; -- 2.26.2

1 0

[PATCH] ALSA:usb-audio:check urb before kill it
by Defang Bo 29 Dec '20

29 Dec '20

Similar to commit<124751d5e>, there should be a check for urb before kill it. Signed-off-by: Defang Bo <bodefang(a)126.com> --- sound/usb/mixer.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 81e987e..0223ef3 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3589,8 +3589,10 @@ void snd_usb_mixer_disconnect(struct usb_mixer_interface *mixer) /* stop any bus activity of a mixer */ static void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer) { - usb_kill_urb(mixer->urb); - usb_kill_urb(mixer->rc_urb); + if (mixer->urb) + usb_kill_urb(mixer->urb); + if (mixer->rc_urb) + usb_kill_urb(mixer->rc_urb); } static int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) -- 2.7.4

3 2

[PATCH v2] confmisc: fix memory leak in snd_func_concat
by Alex Henrie 29 Dec '20

29 Dec '20

Signed-off-by: Alex Henrie <alexhenrie24(a)gmail.com> --- src/confmisc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/src/confmisc.c b/src/confmisc.c index eb8218c1..3ce95c7a 100644 --- a/src/confmisc.c +++ b/src/confmisc.c @@ -419,7 +419,6 @@ int snd_func_concat(snd_config_t **dst, snd_config_t *root, snd_config_t *src, tmp = realloc(res, len + len1 + 1); if (tmp == NULL) { free(ptr); - free(res); err = -ENOMEM; goto __error; } @@ -440,8 +439,8 @@ int snd_func_concat(snd_config_t **dst, snd_config_t *root, snd_config_t *src, err = snd_config_get_id(src, &id); if (err >= 0) err = snd_config_imake_string(dst, id, res); - free(res); __error: + free(res); return err; } #ifndef DOC_HIDDEN -- 2.29.2

2 1

regression of 'make config.log'
by GitHub issues - opened 29 Dec '20

29 Dec '20

alsa-project/alsa-lib issue #108 was opened from libinyang: In the latest alsa-lib code, make config.log will fail. After reverting the commit "5275d170", make config.log successfully. Steps to reproduce: 1. update alsa-lib to the latest 2. ./gitcompile 3. cd test/lsb/ && make config.log The build will fail. And it shows: ../../test-driver: line 107: 78589 Segmentation fault (core dumped) "$@" > $log_file 2>&1 FAIL: config After reverting commit 5275d170 (conf: fix use after free in _snd_config_load_with_include), the build works. Issue URL : https://github.com/alsa-project/alsa-lib/issues/108 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[syzbot+7b8f7100327c574c016f@syzkaller.appspotmail.com: INFO: task hung in remove_proc_subtree]
by Alexey Dobriyan 28 Dec '20

28 Dec '20

----- Forwarded message from syzbot <syzbot+7b8f7100327c574c016f(a)syzkaller.appspotmail.com> ----- From: syzbot <syzbot+7b8f7100327c574c016f(a)syzkaller.appspotmail.com> Subject: INFO: task hung in remove_proc_subtree To: adobriyan(a)gmail.com, akpm(a)linux-foundation.org, ebiederm(a)xmission.com, gladkov.alexey(a)gmail.com, keescook(a)chromium.org, linux-fsdevel(a)vger.kernel.org, linux-kernel(a)vger.kernel.org, syzkaller-bugs(a)googlegroups.com Date: Wed, 16 Dec 2020 09:54:09 -0800 Hello, syzbot found the following issue on: HEAD commit: 7f376f19 Merge tag 'mtd/fixes-for-5.10-rc8' of git://git.k.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=102ac937500000 kernel config: https://syzkaller.appspot.com/x/.config?x=3416bb960d5c705d dashboard link: https://syzkaller.appspot.com/bug?extid=7b8f7100327c574c016f compiler: gcc (GCC) 10.1.0-syz 20200507 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=166e240f500000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=147eddef500000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+7b8f7100327c574c016f(a)syzkaller.appspotmail.com INFO: task kworker/1:2:8489 blocked for more than 143 seconds. Not tainted 5.10.0-rc7-syzkaller #0 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. task:kworker/1:2 state:D stack:24264 pid: 8489 ppid: 2 flags:0x00004000 Workqueue: usb_hub_wq hub_event Call Trace: context_switch kernel/sched/core.c:3779 [inline] __schedule+0x893/0x2130 kernel/sched/core.c:4528 schedule+0xcf/0x270 kernel/sched/core.c:4606 schedule_timeout+0x1d8/0x250 kernel/time/timer.c:1847 do_wait_for_common kernel/sched/completion.c:85 [inline] __wait_for_common kernel/sched/completion.c:106 [inline] wait_for_common kernel/sched/completion.c:117 [inline] wait_for_completion+0x163/0x260 kernel/sched/completion.c:138 proc_entry_rundown+0x1a1/0x1d0 fs/proc/inode.c:262 remove_proc_subtree+0x26f/0x520 fs/proc/generic.c:752 proc_remove fs/proc/generic.c:775 [inline] proc_remove+0x66/0x90 fs/proc/generic.c:772 snd_info_disconnect+0x946/0xc20 sound/core/info.c:757 snd_info_disconnect sound/core/info.c:756 [inline] snd_info_card_disconnect+0x134/0x230 sound/core/info.c:577 snd_card_disconnect+0x2be/0x510 sound/core/init.c:421 usb_audio_disconnect+0x2c4/0x7f0 sound/usb/card.c:877 usb_unbind_interface+0x1d8/0x8d0 drivers/usb/core/driver.c:458 __device_release_driver+0x3bd/0x6f0 drivers/base/dd.c:1154 device_release_driver_internal drivers/base/dd.c:1185 [inline] device_release_driver+0x26/0x40 drivers/base/dd.c:1208 bus_remove_device+0x2eb/0x5a0 drivers/base/bus.c:533 device_del+0x502/0xec0 drivers/base/core.c:3115 usb_disable_device+0x35b/0x7b0 drivers/usb/core/message.c:1410 usb_disconnect.cold+0x27d/0x780 drivers/usb/core/hub.c:2217 hub_port_connect drivers/usb/core/hub.c:5073 [inline] hub_port_connect_change drivers/usb/core/hub.c:5362 [inline] port_event drivers/usb/core/hub.c:5508 [inline] hub_event+0x1c8a/0x42d0 drivers/usb/core/hub.c:5590 process_one_work+0x933/0x15a0 kernel/workqueue.c:2272 worker_thread+0x64c/0x1120 kernel/workqueue.c:2418 kthread+0x3b1/0x4a0 kernel/kthread.c:292 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:296 INFO: task syz-executor488:14592 blocked for more than 143 seconds. Not tainted 5.10.0-rc7-syzkaller #0 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. task:syz-executor488 state:D stack:28144 pid:14592 ppid: 8497 flags:0x00000004 Call Trace: context_switch kernel/sched/core.c:3779 [inline] __schedule+0x893/0x2130 kernel/sched/core.c:4528 schedule+0xcf/0x270 kernel/sched/core.c:4606 schedule_preempt_disabled+0xf/0x20 kernel/sched/core.c:4665 __mutex_lock_common kernel/locking/mutex.c:1033 [inline] __mutex_lock+0x3e2/0x10e0 kernel/locking/mutex.c:1103 snd_info_text_entry_open+0x80/0x2e0 sound/core/info.c:373 proc_reg_open+0x25d/0x620 fs/proc/inode.c:538 do_dentry_open+0x4b9/0x11b0 fs/open.c:817 do_open fs/namei.c:3252 [inline] path_openat+0x1b9a/0x2730 fs/namei.c:3369 do_filp_open+0x17e/0x3c0 fs/namei.c:3396 do_sys_openat2+0x16d/0x420 fs/open.c:1168 do_sys_open fs/open.c:1184 [inline] __do_sys_openat fs/open.c:1200 [inline] __se_sys_openat fs/open.c:1195 [inline] __x64_sys_openat+0x13f/0x1f0 fs/open.c:1195 do_syscall_64+0x2d/0x70 arch/x86/entry/common.c:46 entry_SYSCALL_64_after_hwframe+0x44/0xa9 RIP: 0033:0x450739 RSP: 002b:00007f4a57aefce8 EFLAGS: 00000246 ORIG_RAX: 0000000000000101 RAX: ffffffffffffffda RBX: 0000000000700028 RCX: 0000000000450739 RDX: 0000000000000001 RSI: 00000000200001c0 RDI: ffffffffffffff9c RBP: 0000000000700020 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 000000000070002c R13: 000000000080fbcf R14: 00007f4a57af09c0 R15: 0000000000000000 Showing all locks held in the system: 1 lock held by khungtaskd/1654: #0: ffffffff8b3378e0 (rcu_read_lock){....}-{1:2}, at: debug_show_all_locks+0x53/0x260 kernel/locking/lockdep.c:6254 1 lock held by in:imklog/8177: #0: ffff888012ef1270 (&f->f_pos_lock){+.+.}-{3:3}, at: __fdget_pos+0xe9/0x100 fs/file.c:932 7 locks held by kworker/1:2/8489: #0: ffff888014c3fd38 ((wq_completion)usb_hub_wq){+.+.}-{0:0}, at: arch_atomic64_set arch/x86/include/asm/atomic64_64.h:34 [inline] #0: ffff888014c3fd38 ((wq_completion)usb_hub_wq){+.+.}-{0:0}, at: atomic64_set include/asm-generic/atomic-instrumented.h:856 [inline] #0: ffff888014c3fd38 ((wq_completion)usb_hub_wq){+.+.}-{0:0}, at: atomic_long_set include/asm-generic/atomic-long.h:41 [inline] #0: ffff888014c3fd38 ((wq_completion)usb_hub_wq){+.+.}-{0:0}, at: set_work_data kernel/workqueue.c:616 [inline] #0: ffff888014c3fd38 ((wq_completion)usb_hub_wq){+.+.}-{0:0}, at: set_work_pool_and_clear_pending kernel/workqueue.c:643 [inline] #0: ffff888014c3fd38 ((wq_completion)usb_hub_wq){+.+.}-{0:0}, at: process_one_work+0x821/0x15a0 kernel/workqueue.c:2243 #1: ffffc90000e9fda8 ((work_completion)(&hub->events)){+.+.}-{0:0}, at: process_one_work+0x854/0x15a0 kernel/workqueue.c:2247 #2: ffff88801c28a218 (&dev->mutex){....}-{3:3}, at: device_lock include/linux/device.h:731 [inline] #2: ffff88801c28a218 (&dev->mutex){....}-{3:3}, at: hub_event+0x1c5/0x42d0 drivers/usb/core/hub.c:5536 #3: ffff888022bc8218 (&dev->mutex){....}-{3:3}, at: device_lock include/linux/device.h:731 [inline] #3: ffff888022bc8218 (&dev->mutex){....}-{3:3}, at: usb_disconnect.cold+0x43/0x780 drivers/usb/core/hub.c:2208 #4: ffff888018ea21a8 (&dev->mutex){....}-{3:3}, at: device_lock include/linux/device.h:731 [inline] #4: ffff888018ea21a8 (&dev->mutex){....}-{3:3}, at: __device_driver_lock drivers/base/dd.c:975 [inline] #4: ffff888018ea21a8 (&dev->mutex){....}-{3:3}, at: device_release_driver_internal drivers/base/dd.c:1182 [inline] #4: ffff888018ea21a8 (&dev->mutex){....}-{3:3}, at: device_release_driver+0x1c/0x40 drivers/base/dd.c:1208 #5: ffffffff8c8ae5e8 (register_mutex#6){+.+.}-{3:3}, at: usb_audio_disconnect+0xe4/0x7f0 sound/usb/card.c:866 #6: ffffffff8c838028 (info_mutex){+.+.}-{3:3}, at: snd_info_card_disconnect+0x33/0x230 sound/core/info.c:573 1 lock held by syz-executor488/14592: #0: ffffffff8c838028 (info_mutex){+.+.}-{3:3}, at: snd_info_text_entry_open+0x80/0x2e0 sound/core/info.c:373 ============================================= NMI backtrace for cpu 1 CPU: 1 PID: 1654 Comm: khungtaskd Not tainted 5.10.0-rc7-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x107/0x163 lib/dump_stack.c:118 nmi_cpu_backtrace.cold+0x44/0xd7 lib/nmi_backtrace.c:105 nmi_trigger_cpumask_backtrace+0x1b3/0x230 lib/nmi_backtrace.c:62 trigger_all_cpu_backtrace include/linux/nmi.h:146 [inline] check_hung_uninterruptible_tasks kernel/hung_task.c:209 [inline] watchdog+0xd43/0xfa0 kernel/hung_task.c:294 kthread+0x3b1/0x4a0 kernel/kthread.c:292 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:296 Sending NMI from CPU 1 to CPUs 0: NMI backtrace for cpu 0 CPU: 0 PID: 27 Comm: kworker/u4:2 Not tainted 5.10.0-rc7-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: phy3 ieee80211_iface_work RIP: 0010:__lock_acquire+0x372/0x5500 kernel/locking/lockdep.c:4770 Code: ed 44 09 e8 41 81 e7 00 00 03 00 41 c1 e1 13 41 09 c7 89 d8 48 c1 ea 03 c1 e0 12 25 00 00 04 00 41 09 c7 8b 84 24 38 01 00 00 <45> 09 f9 c1 e0 14 41 09 c1 41 8b 44 24 20 25 ff 1f 00 00 41 09 c1 RSP: 0018:ffffc90000e1f890 EFLAGS: 00000006 RAX: 0000000000000000 RBX: 0000000000000000 RCX: 0000000000000000 RDX: 1ffff110021e47c6 RSI: 0000000000000004 RDI: ffff888010f23e34 RBP: ffff888010f23480 R08: 0000000000000001 R09: 0000000000080000 R10: 0000000000000078 R11: 0000000000000000 R12: ffff888010f23e10 R13: 0000000000000000 R14: ffffffff8b3378e0 R15: 0000000000020000 FS: 0000000000000000(0000) GS:ffff8880b9e00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007fa2708ab000 CR3: 0000000025718000 CR4: 00000000001506f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: lock_acquire kernel/locking/lockdep.c:5437 [inline] lock_acquire+0x29d/0x740 kernel/locking/lockdep.c:5402 rcu_lock_acquire include/linux/rcupdate.h:248 [inline] rcu_read_lock include/linux/rcupdate.h:641 [inline] cgroup_account_cputime include/linux/cgroup.h:781 [inline] update_curr+0x311/0x840 kernel/sched/fair.c:870 enqueue_entity+0x435/0x2080 kernel/sched/fair.c:4206 enqueue_task_fair+0x1d8/0x1a70 kernel/sched/fair.c:5502 enqueue_task kernel/sched/core.c:1572 [inline] activate_task kernel/sched/core.c:1591 [inline] ttwu_do_activate+0x17f/0x660 kernel/sched/core.c:2511 ttwu_queue kernel/sched/core.c:2701 [inline] try_to_wake_up+0x54c/0x1330 kernel/sched/core.c:2979 wake_up_worker kernel/workqueue.c:837 [inline] process_one_work+0x75a/0x15a0 kernel/workqueue.c:2235 worker_thread+0x64c/0x1120 kernel/workqueue.c:2418 kthread+0x3b1/0x4a0 kernel/kthread.c:292 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:296 --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches ----- End forwarded message -----

1 0

[PATCH] ALSA: hda: Resume codec for system suspend if LED is controlled by codec
by Kai-Heng Feng 28 Dec '20

28 Dec '20

Laptop with codec controlled LEDs takes a very long time to suspend after commit 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization"): [ 90.065964] PM: suspend entry (s2idle) [ 90.067337] Filesystems sync: 0.001 seconds [ 90.185758] Freezing user space processes ... (elapsed 0.002 seconds) done. [ 90.188713] OOM killer disabled. [ 90.188714] Freezing remaining freezable tasks ... (elapsed 0.001 seconds) done. [ 90.190024] printk: Suspending console(s) (use no_console_suspend to debug) [ 90.904912] intel_pch_thermal 0000:00:12.0: CPU-PCH is cool [49C], continue to suspend [ 321.262505] snd_hda_codec_realtek ehdaudio0D0: Unable to sync register 0x2b8000. -5 [ 328.426919] snd_hda_codec_realtek ehdaudio0D0: Unable to sync register 0x2b8000. -5 [ 329.490933] ACPI: EC: interrupt blocked That commit keeps codec suspended during the system suspend. This doesn't play well with codec controlled mute and micmute LEDs, because LED core will use codec registers to turn off those LEDs. Currently, all users of create_mute_led_cdev() use codec to control LED, so unconditionally runtime resume those codecs before system suspend to solve the problem. Fixes: 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization") Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- include/sound/hda_codec.h | 1 + sound/pci/hda/hda_codec.c | 7 +++++++ sound/pci/hda/hda_generic.c | 1 + 3 files changed, 9 insertions(+) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 2e8d51937acd..b01d76abe008 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -255,6 +255,7 @@ struct hda_codec { unsigned int relaxed_resume:1; /* don't resume forcibly for jack */ unsigned int forced_resume:1; /* forced resume for jack */ unsigned int mst_no_extra_pcms:1; /* no backup PCMs for DP-MST */ + unsigned int resume_for_sleep:1; /* runtime resume for system sleep */ #ifdef CONFIG_PM unsigned long power_on_acct; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 687216e74526..b890d9b4339e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2983,6 +2983,13 @@ static int hda_codec_runtime_resume(struct device *dev) #ifdef CONFIG_PM_SLEEP static int hda_codec_pm_prepare(struct device *dev) { + struct hda_codec *codec = dev_to_hda_codec(dev); + + if (codec->resume_for_sleep) { + pm_runtime_resume(dev); + return 0; + } + return pm_runtime_suspended(dev); } diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8060cc86dfea..6d259d5bb5bb 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3913,6 +3913,7 @@ static int create_mute_led_cdev(struct hda_codec *codec, cdev->brightness_set_blocking = callback; cdev->brightness = ledtrig_audio_get(micmute ? LED_AUDIO_MICMUTE : LED_AUDIO_MUTE); cdev->flags = LED_CORE_SUSPENDRESUME; + codec->resume_for_sleep = 1; return devm_led_classdev_register(&codec->core.dev, cdev); } -- 2.29.2

2 2

[PATCH] ASoC: amd:Replacing MSI with Legacy IRQ model
by Ravulapati Vishnu vardhan rao 28 Dec '20

28 Dec '20

When we try to play and capture simultaneously we see that interrupts are genrated but our handler is not being acknowledged, After investigating further more in detail on this issue we found that IRQ delivery via MSI from the ACP IP is unreliable and so sometimes interrupt generated will not be acknowledged so MSI model shouldn't be used and using legacy IRQs will resolve interrupt handling issue. This patch replaces MSI interrupt handling with legacy IRQ model. Issue can be reproduced easily by running below python script: import subprocess import time import threading def do2(): cmd = 'aplay -f dat -D hw:2,1 /dev/zero -d 1' subprocess.call(cmd, stdin=subprocess.PIPE, stderr=subprocess.PIPE, shell=True) print('Play Done') def run(): for i in range(1000): do2() def do(i): cmd = 'arecord -f dat -D hw:2,2 /dev/null -d 1' subprocess.call(cmd, stdout=subprocess.PIPE, stderr=subprocess.PIPE, shell=True) print(datetime.datetime.now(), i) t = threading.Thread(target=run) t.start() for i in range(1000): do(i) t.join() After applying this patch issue is resolved. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index 8c138e490f0c..d3536fd6a124 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -140,21 +140,14 @@ static int snd_acp3x_probe(struct pci_dev *pci, goto release_regions; } - /* check for msi interrupt support */ - ret = pci_enable_msi(pci); - if (ret) - /* msi is not enabled */ - irqflags = IRQF_SHARED; - else - /* msi is enabled */ - irqflags = 0; + irqflags = IRQF_SHARED; addr = pci_resource_start(pci, 0); adata->acp3x_base = devm_ioremap(&pci->dev, addr, pci_resource_len(pci, 0)); if (!adata->acp3x_base) { ret = -ENOMEM; - goto disable_msi; + goto release_regions; } pci_set_master(pci); pci_set_drvdata(pci, adata); @@ -162,7 +155,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, adata->pme_en = rv_readl(adata->acp3x_base + mmACP_PME_EN); ret = acp3x_init(adata); if (ret) - goto disable_msi; + goto release_regions; val = rv_readl(adata->acp3x_base + mmACP_I2S_PIN_CONFIG); switch (val) { @@ -251,8 +244,6 @@ static int snd_acp3x_probe(struct pci_dev *pci, de_init: if (acp3x_deinit(adata->acp3x_base)) dev_err(&pci->dev, "ACP de-init failed\n"); -disable_msi: - pci_disable_msi(pci); release_regions: pci_release_regions(pci); disable_pci: @@ -311,7 +302,6 @@ static void snd_acp3x_remove(struct pci_dev *pci) dev_err(&pci->dev, "ACP de-init failed\n"); pm_runtime_forbid(&pci->dev); pm_runtime_get_noresume(&pci->dev); - pci_disable_msi(pci); pci_release_regions(pci); pci_disable_device(pci); } -- 2.17.1

5 5

[PATCH v2] ASoC: AMD Renoir - add DMI entry for Lenovo ThinkPad E14 Gen 2
by Jaroslav Kysela 28 Dec '20

28 Dec '20

The ThinkPad E14 Gen 2 latop does not have the internal digital microphone connected to the AMD's ACP bridge, but it's advertised via BIOS. The internal microphone is connected to the HDA codec. Use DMI to block the microphone PCM device for this platform. Reported-by: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Cc: <stable(a)kernel.org> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index fa169bf09886..17ec35be73ac 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -171,6 +171,13 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "LNVNB161216"), } }, + { + /* Lenovo ThinkPad E14 Gen 2 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "20T6CTO1WW"), + } + }, {} }; -- 2.29.2

2 1

[PATCH v2] ASoC: AMD Renoir - add DMI entry for Lenovo ThinkPad X395
by Jaroslav Kysela 28 Dec '20

28 Dec '20

The ThinkPad X395 latop does not have the internal digital microphone connected to the AMD's ACP bridge, but it's advertised via BIOS. The internal microphone is connected to the HDA codec. Use DMI to block the microphone PCM device for this platform. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1892115 Cc: <stable(a)kernel.org> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 17ec35be73ac..deca8c7a0e87 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -178,6 +178,13 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20T6CTO1WW"), } }, + { + /* Lenovo ThinkPad X395 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), + } + }, {} }; -- 2.29.2

2 2

[PATCH] ASoC: wm_adsp: Improve handling of raw byte streams
by Richard Fitzgerald 28 Dec '20

28 Dec '20

As the register map is 16-bit or 32-bit big-endian, the 24-bit DSP words appear padded and with the bytes swapped. When reading a raw stream of bytes, the pad bytes must be removed and the data bytes swapped back to their original order. The previous implementation of this assumed that the be32_to_cpu() in wm_adsp_read_data_block() would swap back to little-endian. But this is obviously only true on a little-endian CPU. It also made two walks through the data, once to endian-swap and again to strip the pad bytes. This patch re-works the code so that the endian-swap and unpad are done together in a single walk, and it is not dependent on the endianness of the CPU. The data_word_size argument to wm_adsp_remove_padding() has been dropped because currently this is always 3. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 55 +++++++++++++++++++++----------------- 1 file changed, 30 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index bcf18bf15a02..7c52ada10af3 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3662,12 +3662,12 @@ int wm_adsp_compr_get_caps(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(wm_adsp_compr_get_caps); -static int wm_adsp_read_data_block(struct wm_adsp *dsp, int mem_type, - unsigned int mem_addr, - unsigned int num_words, u32 *data) +static int wm_adsp_read_raw_data_block(struct wm_adsp *dsp, int mem_type, + unsigned int mem_addr, + unsigned int num_words, __be32 *data) { struct wm_adsp_region const *mem = wm_adsp_find_region(dsp, mem_type); - unsigned int i, reg; + unsigned int reg; int ret; if (!mem) @@ -3680,16 +3680,22 @@ static int wm_adsp_read_data_block(struct wm_adsp *dsp, int mem_type, if (ret < 0) return ret; - for (i = 0; i < num_words; ++i) - data[i] = be32_to_cpu(data[i]) & 0x00ffffffu; - return 0; } static inline int wm_adsp_read_data_word(struct wm_adsp *dsp, int mem_type, unsigned int mem_addr, u32 *data) { - return wm_adsp_read_data_block(dsp, mem_type, mem_addr, 1, data); + __be32 raw; + int ret; + + ret = wm_adsp_read_raw_data_block(dsp, mem_type, mem_addr, 1, &raw); + if (ret) + return ret; + + *data = be32_to_cpu(raw) & 0x00ffffffu; + + return 0; } static int wm_adsp_write_data_word(struct wm_adsp *dsp, int mem_type, @@ -3722,18 +3728,22 @@ static inline int wm_adsp_buffer_write(struct wm_adsp_compr_buf *buf, buf->host_buf_ptr + field_offset, data); } -static void wm_adsp_remove_padding(u32 *buf, int nwords, int data_word_size) +static void wm_adsp_remove_padding(u32 *buf, int nwords) { - u8 *pack_in = (u8 *)buf; + const __be32 *pack_in = (__be32 *)buf; u8 *pack_out = (u8 *)buf; - int i, j; + int i; - /* Remove the padding bytes from the data read from the DSP */ + /* + * DSP words from the register map have pad bytes and the data bytes + * are in swapped order. This swaps back to the original little-endian + * order and strips the pad bytes. + */ for (i = 0; i < nwords; i++) { - for (j = 0; j < data_word_size; j++) - *pack_out++ = *pack_in++; - - pack_in += sizeof(*buf) - data_word_size; + u32 word = be32_to_cpu(*pack_in++); + *pack_out++ = (u8)word; + *pack_out++ = (u8)(word >> 8); + *pack_out++ = (u8)(word >> 16); } } @@ -3917,12 +3927,7 @@ static int wm_adsp_buffer_parse_coeff(struct wm_coeff_ctl *ctl) return -EINVAL; } - for (i = 0; i < ARRAY_SIZE(coeff_v1.name); i++) - coeff_v1.name[i] = be32_to_cpu(coeff_v1.name[i]); - - wm_adsp_remove_padding((u32 *)&coeff_v1.name, - ARRAY_SIZE(coeff_v1.name), - WM_ADSP_DATA_WORD_SIZE); + wm_adsp_remove_padding((u32 *)&coeff_v1.name, ARRAY_SIZE(coeff_v1.name)); buf->name = kasprintf(GFP_KERNEL, "%s-dsp-%s", ctl->dsp->part, (char *)&coeff_v1.name); @@ -4261,12 +4266,12 @@ static int wm_adsp_buffer_capture_block(struct wm_adsp_compr *compr, int target) return 0; /* Read data from DSP */ - ret = wm_adsp_read_data_block(buf->dsp, mem_type, adsp_addr, - nwords, compr->raw_buf); + ret = wm_adsp_read_raw_data_block(buf->dsp, mem_type, adsp_addr, + nwords, compr->raw_buf); if (ret < 0) return ret; - wm_adsp_remove_padding(compr->raw_buf, nwords, WM_ADSP_DATA_WORD_SIZE); + wm_adsp_remove_padding(compr->raw_buf, nwords); /* update read index to account for words read */ buf->read_index += nwords; -- 2.20.1

3 2

[PATCH] ASoC: meson: Use managed DMA buffer allocation
by Lars-Peter Clausen 28 Dec '20

28 Dec '20

Using a managed buffer will pre-allocate the buffer using snd_pcm_lib_preallocate_pages() and automatically free it when the PCM is destroyed. In addition it will call snd_pcm_lib_malloc_pages() before the driver's hw_params() callback and snd_pcm_lib_free_pages() after the driver's hw_free() callback. This slightly reduces the boilerplate code of the driver. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/meson/aiu-fifo-i2s.c | 1 - sound/soc/meson/aiu-fifo-spdif.c | 1 - sound/soc/meson/aiu-fifo.c | 18 ++---------------- 3 files changed, 2 insertions(+), 18 deletions(-) diff --git a/sound/soc/meson/aiu-fifo-i2s.c b/sound/soc/meson/aiu-fifo-i2s.c index d91b0d874342..2388a2d0b3a6 100644 --- a/sound/soc/meson/aiu-fifo-i2s.c +++ b/sound/soc/meson/aiu-fifo-i2s.c @@ -124,7 +124,6 @@ const struct snd_soc_dai_ops aiu_fifo_i2s_dai_ops = { .trigger = aiu_fifo_i2s_trigger, .prepare = aiu_fifo_i2s_prepare, .hw_params = aiu_fifo_i2s_hw_params, - .hw_free = aiu_fifo_hw_free, .startup = aiu_fifo_startup, .shutdown = aiu_fifo_shutdown, }; diff --git a/sound/soc/meson/aiu-fifo-spdif.c b/sound/soc/meson/aiu-fifo-spdif.c index 44eb6faacf44..2fb30f89bf7a 100644 --- a/sound/soc/meson/aiu-fifo-spdif.c +++ b/sound/soc/meson/aiu-fifo-spdif.c @@ -158,7 +158,6 @@ const struct snd_soc_dai_ops aiu_fifo_spdif_dai_ops = { .trigger = fifo_spdif_trigger, .prepare = fifo_spdif_prepare, .hw_params = fifo_spdif_hw_params, - .hw_free = aiu_fifo_hw_free, .startup = aiu_fifo_startup, .shutdown = aiu_fifo_shutdown, }; diff --git a/sound/soc/meson/aiu-fifo.c b/sound/soc/meson/aiu-fifo.c index aa88aae8e517..4ad23267cace 100644 --- a/sound/soc/meson/aiu-fifo.c +++ b/sound/soc/meson/aiu-fifo.c @@ -99,11 +99,6 @@ int aiu_fifo_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct aiu_fifo *fifo = dai->playback_dma_data; dma_addr_t end; - int ret; - - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - if (ret < 0) - return ret; /* Setup the fifo boundaries */ end = runtime->dma_addr + runtime->dma_bytes - fifo->fifo_block; @@ -124,12 +119,6 @@ int aiu_fifo_hw_params(struct snd_pcm_substream *substream, return 0; } -int aiu_fifo_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - return snd_pcm_lib_free_pages(substream); -} - static irqreturn_t aiu_fifo_isr(int irq, void *dev_id) { struct snd_pcm_substream *playback = dev_id; @@ -187,15 +176,12 @@ void aiu_fifo_shutdown(struct snd_pcm_substream *substream, int aiu_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { - struct snd_pcm_substream *substream = - rtd->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; struct snd_card *card = rtd->card->snd_card; struct aiu_fifo *fifo = dai->playback_dma_data; size_t size = fifo->pcm->buffer_bytes_max; - snd_pcm_lib_preallocate_pages(substream, - SNDRV_DMA_TYPE_DEV, - card->dev, size, size); + snd_pcm_set_managed_buffer_all(rtd->pcm, SNDRV_DMA_TYPE_DEV, + card->dev, size, size); return 0; } -- 2.20.1

3 4

[PATCH v1] catpt: Switch to use list_entry_is_head() helper
by Andy Shevchenko 28 Dec '20

28 Dec '20

Since we got list_entry_is_head() helper in the generic header, we may switch catpt to use it. It removes the need in additional variable. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/catpt/pcm.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index e5d54bb1c42a..1c7260db3238 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -331,10 +331,10 @@ static int catpt_dai_apply_usettings(struct snd_soc_dai *dai, { struct catpt_dev *cdev = dev_get_drvdata(dai->dev); struct snd_soc_component *component = dai->component; - struct snd_kcontrol *pos, *kctl = NULL; const char *name; int ret; u32 id = stream->info.stream_hw_id; + struct snd_kcontrol *pos; /* only selected streams have individual controls */ switch (id) { @@ -356,17 +356,15 @@ static int catpt_dai_apply_usettings(struct snd_soc_dai *dai, list_for_each_entry(pos, &component->card->snd_card->controls, list) { if (pos->private_data == component && - !strncmp(name, pos->id.name, sizeof(pos->id.name))) { - kctl = pos; + !strncmp(name, pos->id.name, sizeof(pos->id.name))) break; - } } - if (!kctl) + if (list_entry_is_head(pos, &component->card->snd_card->controls, list)) return -ENOENT; if (stream->template->type != CATPT_STRM_TYPE_LOOPBACK) - return catpt_set_dspvol(cdev, id, (long *)kctl->private_value); - ret = catpt_ipc_mute_loopback(cdev, id, *(bool *)kctl->private_value); + return catpt_set_dspvol(cdev, id, (long *)pos->private_value); + ret = catpt_ipc_mute_loopback(cdev, id, *(bool *)pos->private_value); if (ret) return CATPT_IPC_ERROR(ret); return 0; -- 2.29.2

3 4

[PATCH v1] ASoC: sprd: Switch to use list_entry_is_head() helper
by Andy Shevchenko 28 Dec '20

28 Dec '20

Since we got list_entry_is_head() helper in the generic header, we may switch driver to use it. This eliminates the need in additional variable. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/sprd/sprd-mcdt.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index f439e5503a3c..34b2ce733b54 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -866,23 +866,23 @@ EXPORT_SYMBOL_GPL(sprd_mcdt_chan_dma_disable); struct sprd_mcdt_chan *sprd_mcdt_request_chan(u8 channel, enum sprd_mcdt_channel_type type) { - struct sprd_mcdt_chan *temp, *chan = NULL; + struct sprd_mcdt_chan *temp; mutex_lock(&sprd_mcdt_list_mutex); list_for_each_entry(temp, &sprd_mcdt_chan_list, list) { if (temp->type == type && temp->id == channel) { - chan = temp; + list_del_init(&temp->list); break; } } - if (chan) - list_del(&chan->list); + if (list_entry_is_head(temp, &sprd_mcdt_chan_list, list)) + temp = NULL; mutex_unlock(&sprd_mcdt_list_mutex); - return chan; + return temp; } EXPORT_SYMBOL_GPL(sprd_mcdt_request_chan); -- 2.29.2

2 1

[PATCH v2] catpt: Switch to use list_entry_is_head() helper
by Andy Shevchenko 28 Dec '20

28 Dec '20

Since we got list_entry_is_head() helper in the generic header, we may switch catpt to use it. It removes the need in additional variable. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- v2: left the structure definition at its initial location (Cezary) sound/soc/intel/catpt/pcm.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index e5d54bb1c42a..9d304b3a817f 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -331,7 +331,7 @@ static int catpt_dai_apply_usettings(struct snd_soc_dai *dai, { struct catpt_dev *cdev = dev_get_drvdata(dai->dev); struct snd_soc_component *component = dai->component; - struct snd_kcontrol *pos, *kctl = NULL; + struct snd_kcontrol *pos; const char *name; int ret; u32 id = stream->info.stream_hw_id; @@ -356,17 +356,15 @@ static int catpt_dai_apply_usettings(struct snd_soc_dai *dai, list_for_each_entry(pos, &component->card->snd_card->controls, list) { if (pos->private_data == component && - !strncmp(name, pos->id.name, sizeof(pos->id.name))) { - kctl = pos; + !strncmp(name, pos->id.name, sizeof(pos->id.name))) break; - } } - if (!kctl) + if (list_entry_is_head(pos, &component->card->snd_card->controls, list)) return -ENOENT; if (stream->template->type != CATPT_STRM_TYPE_LOOPBACK) - return catpt_set_dspvol(cdev, id, (long *)kctl->private_value); - ret = catpt_ipc_mute_loopback(cdev, id, *(bool *)kctl->private_value); + return catpt_set_dspvol(cdev, id, (long *)pos->private_value); + ret = catpt_ipc_mute_loopback(cdev, id, *(bool *)pos->private_value); if (ret) return CATPT_IPC_ERROR(ret); return 0; -- 2.29.2

2 2

cross-compilation broken since cf09b6742a507555bb712101aff9e919009f0683
by GitHub issues - opened 28 Dec '20

28 Dec '20

alsa-project/alsa-firmware issue #3 was opened from flokli: I was excited to see c9e5b369919a956cf56d50a6f0be076e364f7820 getting merged, but realized this broke immediately afterwards in cf09b6742a507555bb712101aff9e919009f0683: ``` git bisect start # good: [c9e5b369919a956cf56d50a6f0be076e364f7820] Get cross compilation to work git bisect good c9e5b369919a956cf56d50a6f0be076e364f7820 # bad: [3ab9711ae106851b4185da42d5873702a3008744] Release v1.2.4 git bisect bad 3ab9711ae106851b4185da42d5873702a3008744 # bad: [41008541de9bd2b184058ec2204f172e3c9f64fe] introduce Makefile.hotplug git bisect bad 41008541de9bd2b184058ec2204f172e3c9f64fe # bad: [546aa5b30694889b00b424a6e9de2c5f34ce5da2] introduce Makefile.fw_writer git bisect bad 546aa5b30694889b00b424a6e9de2c5f34ce5da2 # bad: [cf09b6742a507555bb712101aff9e919009f0683] upgrade ax_prog_cc_for_build.m4 to serial 18 git bisect bad cf09b6742a507555bb712101aff9e919009f0683 # first bad commit: [cf09b6742a507555bb712101aff9e919009f0683] upgrade ax_prog_cc_for_build.m4 to serial 18 ``` cc @perexg Issue URL : https://github.com/alsa-project/alsa-firmware/issues/3 Repository URL: https://github.com/alsa-project/alsa-firmware

1 0

[PATCH] dt-bindings: Drop redundant maxItems/items
by Rob Herring 28 Dec '20

28 Dec '20

'maxItems' equal to the 'items' list length is redundant. 'maxItems' is preferred for a single entry while greater than 1 should have an 'items' list. A meta-schema check for this is pending once these existing cases are fixed. Cc: Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Jassi Brar <jaswinder.singh(a)linaro.org> Cc: dri-devel(a)lists.freedesktop.org Cc: dmaengine(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-usb(a)vger.kernel.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../devicetree/bindings/display/xlnx/xlnx,zynqmp-dpsub.yaml | 1 - Documentation/devicetree/bindings/dma/renesas,rcar-dmac.yaml | 1 - Documentation/devicetree/bindings/mailbox/arm,mhu.yaml | 1 - .../devicetree/bindings/sound/nvidia,tegra30-hda.yaml | 2 -- Documentation/devicetree/bindings/usb/renesas,usb-xhci.yaml | 1 - Documentation/devicetree/bindings/usb/renesas,usbhs.yaml | 3 --- 6 files changed, 9 deletions(-) diff --git a/Documentation/devicetree/bindings/display/xlnx/xlnx,zynqmp-dpsub.yaml b/Documentation/devicetree/bindings/display/xlnx/xlnx,zynqmp-dpsub.yaml index 7b9d468c3e52..403d57977ee7 100644 --- a/Documentation/devicetree/bindings/display/xlnx/xlnx,zynqmp-dpsub.yaml +++ b/Documentation/devicetree/bindings/display/xlnx/xlnx,zynqmp-dpsub.yaml @@ -98,7 +98,6 @@ properties: maxItems: 1 dmas: - maxItems: 4 items: - description: Video layer, plane 0 (RGB or luma) - description: Video layer, plane 1 (U/V or U) diff --git a/Documentation/devicetree/bindings/dma/renesas,rcar-dmac.yaml b/Documentation/devicetree/bindings/dma/renesas,rcar-dmac.yaml index b548e4723936..c07eb6f2fc8d 100644 --- a/Documentation/devicetree/bindings/dma/renesas,rcar-dmac.yaml +++ b/Documentation/devicetree/bindings/dma/renesas,rcar-dmac.yaml @@ -73,7 +73,6 @@ properties: maxItems: 1 clock-names: - maxItems: 1 items: - const: fck diff --git a/Documentation/devicetree/bindings/mailbox/arm,mhu.yaml b/Documentation/devicetree/bindings/mailbox/arm,mhu.yaml index d43791a2dde7..d07eb00b97c8 100644 --- a/Documentation/devicetree/bindings/mailbox/arm,mhu.yaml +++ b/Documentation/devicetree/bindings/mailbox/arm,mhu.yaml @@ -61,7 +61,6 @@ properties: - description: low-priority non-secure - description: high-priority non-secure - description: Secure - maxItems: 3 clocks: maxItems: 1 diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml index e543a6123792..b55775e21de6 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml @@ -44,7 +44,6 @@ properties: maxItems: 3 clock-names: - maxItems: 3 items: - const: hda - const: hda2hdmi @@ -54,7 +53,6 @@ properties: maxItems: 3 reset-names: - maxItems: 3 items: - const: hda - const: hda2hdmi diff --git a/Documentation/devicetree/bindings/usb/renesas,usb-xhci.yaml b/Documentation/devicetree/bindings/usb/renesas,usb-xhci.yaml index 0f078bd0a3e5..22603256ddf8 100644 --- a/Documentation/devicetree/bindings/usb/renesas,usb-xhci.yaml +++ b/Documentation/devicetree/bindings/usb/renesas,usb-xhci.yaml @@ -51,7 +51,6 @@ properties: maxItems: 1 phy-names: - maxItems: 1 items: - const: usb diff --git a/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml b/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml index 737c1f47b7de..54c361d4a7af 100644 --- a/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml +++ b/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml @@ -74,11 +74,8 @@ properties: phys: maxItems: 1 - items: - - description: phandle + phy specifier pair. phy-names: - maxItems: 1 items: - const: usb -- 2.27.0

7 9

pulse plugin crash when stream has a hole
by GitHub issues - edited 28 Dec '20

28 Dec '20

alsa-project/alsa-plugins issue #16 was edited from i-garrison: Following up on this debian bug report https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=824850 Looks like this crash is caused by missing stream hole handling in pulse plugin. Pulseaudio change was done just before `pulseaudio-v3.0` release and this issue should be reproducible with all current versions: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/dfd44036b54d6… quoting from there: ``` ... let the caller decide how the holes should be handled, so in case of holes, pa_stream_peek() will return NULL data pointer and the length of the hole in the nbytes argument ``` Here in pulse plugin, if there is a hole in stream `pa_stream_peek` would return NULL buffer pointer but non-zero `frag_length` and later memcpy is happily trying to copy from NULL source: https://github.com/alsa-project/alsa-plugins/blob/7028580da1db9a2362dd9b063… Instead, pulse plugin should detect there is a hole by checking if `src_buf` is still NULL while `frag_length` is not zero, and adjust transfer result accordingly. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/16 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v3 0/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 28 Dec '20

28 Dec '20

>Hi Gyeongtaek > >Thank you for your help > >> I tried to find out some change from the history. >> However, I saw that dpcm_lock never be held in dpcm_be_dai_trigger() from the beginning. >> So, I just fixed some miss from the commit comment and resend. >> If you have another suggestion or request, just let me know. > >OK, thanks. >I just thought it was related to recent (especially mine :) patch. > >BYW, your posted v2 still doesn't have alignment on log. >Is it mailer issue ? Sorry. I didn't fully understand what alignment means. I fixed the log by looking at some previous commits. Thank you so much. Gyeongtaek Lee > >> And I have a question. >> This patch can?t be applied to stable branches directly by conflict. >> I found out following commits are needed to apply this patch into the 5.4 stable. >> ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() >> ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code >> In this case, what should I do when stable maintainer requests resolving conflict to me? >> Should I send those patches to the maintainer? >> Or rewrite the patch on the latest of the each stable? > >Mark, please help him > >Thank you for your help !! > >Best regards >--- >Kuninori Morimoto > 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: fa02fcd94b0c8dff6cc65714510cf25ad194b90d -- 2.21.0

1 0

pulse: handle reading pulse stream hole
by GitHub pull_request - opened 28 Dec '20

28 Dec '20

alsa-project/alsa-plugins pull request #17 was opened from i-garrison: If there is a hole in pulse stream, handle it by generating silence. This should fix #16 Request URL : https://github.com/alsa-project/alsa-plugins/pull/17 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/17.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

pulse plugin crash when stream has a hole
by GitHub issues - opened 28 Dec '20

28 Dec '20

alsa-project/alsa-plugins issue #16 was opened from i-garrison: Following up on this debian bug report https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=824850 Looks like this crash is caused by missing stream hole handling in pulse plugin. Pulseaudio change was done just before `pulseaudio-v3.0` release and this issue should be reproducible with all current versions: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/dfd44036b54d6… quoting from there: ``` ... let the caller decide how the holes should be handled, so in case of holes, pa_stream_peek() will return NULL data pointer and the length of the hole in the nbytes argument ``` Here in pulse plugin, if there is a hole in stream `pa_stream_peek` would return NULL buffer pointer but non-zero `frag_length` and later memcpy is happily trying to copy from NULL source: https://github.com/alsa-project/alsa-plugins/blob/7028580da1db9a2362dd9b063… Instead, pulse plugin should detect there is a hole by checking if `src_buf` is still NULL while `frag_lenght` is not zero, and adjust transfer result accordingly. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/16 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH 0/9] scan-build fixes
by Alex Henrie 28 Dec '20

28 Dec '20

These defects were identified by scan-build <https://clang-analyzer.llvm.org/scan-build.html>. I hope that drawing attention to the problems helps even if you decide to fix some of them in different ways than I am proposing. Happy holidays! -Alex Alex Henrie (9): conf: fix use after free in _snd_config_load_with_include ucm: fix bad frees in get_list0 and get_list20 rawmidi: fix memory leak in snd_rawmidi_virtual_open confmisc: fix memory leak in snd_func_concat timer: fix sizeof operator mismatch in snd_timer_query_hw_open pcm: remove dead assignments from snd_pcm_rate_(commit_area|grab_next_period) pcm_multi: remove dead assignment from _snd_pcm_multi_open pcm: fix undefined bit shift in bad_pcm_state conf: remove dead code from get_hexachar include/pcm.h | 4 +++- src/conf.c | 13 ++++--------- src/confmisc.c | 2 +- src/pcm/pcm.c | 2 ++ src/pcm/pcm_local.h | 2 +- src/pcm/pcm_multi.c | 1 - src/pcm/pcm_rate.c | 2 -- src/rawmidi/rawmidi_virt.c | 3 ++- src/timer/timer_query_hw.c | 2 +- src/ucm/main.c | 4 ++-- 10 files changed, 16 insertions(+), 19 deletions(-) -- 2.29.2.368.ge46b91665e.dirty

3 23

[PATCH] ASoC: AMD Renoir - add DMI entry for Lenovo ThinkPad E14 Gen 2
by Jaroslav Kysela 27 Dec '20

27 Dec '20

The ThinkPad E14 Gen 2 latop does not have the internal digital microphone connected to the AMD's ACP bridge, but it's advertised via BIOS. The internal microphone is connected to the HDA codec. Use DMI to block the microphone PCM device for this platform. Reported-by: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Cc: <stable(a)kernel.org> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index fa169bf09886..749e520eaca9 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -169,6 +169,11 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_EXACT_MATCH(DMI_BOARD_NAME, "LNVNB161216"), + }, + /* Lenovo ThinkPad E14 Gen 2 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "20T6CTO1WW"), } }, {} -- 2.29.2

2 1

Re: [PATCH] docs: Fix reST markup when linking to sections
by Mauro Carvalho Chehab 27 Dec '20

27 Dec '20

Em Sat, 26 Dec 2020 13:18:58 +0000 Nícolas F. R. A. Prado <nfraprado(a)protonmail.com> escreveu: > During the process of converting the documentation to reST, some links > were converted using the following wrong syntax (and sometimes using %20 > instead of spaces): > The patch itself looks ok, although the description has an issue (IMHO). Also, some references can be cleaned. See below. > `Display text <#section-name-in-html>`__ > > This syntax can work in html, but isn't the one described in docutils, Well, docutils define two types of references at: https://docutils.sourceforge.io/docs/ref/rst/restructuredtext.html#referenc… The first one are "simple reference names", defined as: ``Simple reference names are single words consisting of alphanumerics plus isolated (no two adjacent) internal hyphens, underscores, periods, colons and plus signs; no whitespace or other characters are allowed.`` On this type, "-_.,+" characters are allowed. "%" and "#" aren't. The second one is "phrase-references", defined as: ``Reference names using punctuation or whose names are phrases (two or more space-separated words) are called "phrase-references".`` Here, the spec doesn't specify the charset associated with "punctuation". As this kind of reference is auto-generated from the chapter titles, I would expect it to allow all ASCII punctuation characters (e. g. all non-alphanumeric symbols between 0x20-0x7f). So, a reference like those: #Summary #Forcing%20Quiescent%20States Violates the spec, as it would be a simple reference with invalid chars, but: #Forcing Quiescent States Should be valid, according with the spec (still, while doing such cleanup, I would remove "#"). I would add something like the above at the patch description. > and it also doesn't work on pdf. The following syntax should be used > instead: > > `Display text <Section Name_>`__ > > The usual toolchain doesn't mind this unusual syntax, but it causes > errors when trying to build using the not-yet-merged rst2pdf: > > ValueError: format not resolved, probably missing URL scheme or undefined destination target for 'Forcing%20Quiescent%20States' > > Fixes: ccc9971e2147 ("docs: rcu: convert some articles from html to ReST") > Fixes: c8cce10a62aa ("docs: Fix the reference labels in Locking.rst") > Fixes: e548cdeffcd8 ("docs-rst: convert kernel-locking to ReST") > Fixes: 7ddedebb03b7 ("ALSA: doc: ReSTize writing-an-alsa-driver document") > Signed-off-by: Nícolas F. R. A. Prado <nfraprado(a)protonmail.com> ... > @@ -596,7 +596,7 @@ maintain ordering. For example, if the callback function wakes up a task > that runs on some other CPU, proper ordering must in place in both the > callback function and the task being awakened. To see why this is > important, consider the top half of the `grace-period > -cleanup <#Grace-Period%20Cleanup>`__ diagram. The callback might be > +cleanup <Grace-Period Cleanup_>`__ diagram. The callback might be > running on a CPU corresponding to the leftmost leaf ``rcu_node`` > structure, and awaken a task that is to run on a CPU corresponding to > the rightmost leaf ``rcu_node`` structure, and the grace-period kernel > diff --git a/Documentation/RCU/Design/Requirements/Requirements.rst b/Documentation/RCU/Design/Requirements/Requirements.rst > index 1ae79a10a8de..d4125caf394e 100644 > --- a/Documentation/RCU/Design/Requirements/Requirements.rst > +++ b/Documentation/RCU/Design/Requirements/Requirements.rst > @@ -45,7 +45,7 @@ requirements: > #. `Other RCU Flavors`_ > #. `Possible Future Changes`_ > > -This is followed by a `summary <#Summary>`__, however, the answers to > +This is followed by a `summary <Summary_>`__, however, the answers to Hmm... why are you ending "Summary" with a "_"? This should be equivalent to: `summary <summary>`__ In this specific case, however, you could use, instead[1]: summary_ as there's no need to use an indirect hyperlink target here. (the same applies to a few other similar cases on your patch) [1] https://docutils.sourceforge.io/docs/user/rst/quickref.html#hyperlink-targe… Thanks, Mauro

2 1

[PATCH] ASoC: AMD Renoir - add DMI entry for Lenovo ThinkPad X395
by Jaroslav Kysela 27 Dec '20

27 Dec '20

The ThinkPad X395 latop does not have the internal digital microphone connected to the AMD's ACP bridge, but it's advertised via BIOS. The internal microphone is connected to the HDA codec. Use DMI to block the microphone PCM device for this platform. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1892115 Cc: <stable(a)kernel.org> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 749e520eaca9..a2993696e600 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -174,6 +174,11 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_EXACT_MATCH(DMI_BOARD_NAME, "20T6CTO1WW"), + }, + /* Lenovo ThinkPad X395 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, {} -- 2.29.2

1 0

Re: [PATCH] docs: Fix reST markup when linking to sections
by Takashi Iwai 27 Dec '20

27 Dec '20

On Sat, 26 Dec 2020 14:18:58 +0100, Nícolas F. R. A. Prado wrote: > > During the process of converting the documentation to reST, some links > were converted using the following wrong syntax (and sometimes using %20 > instead of spaces): > > `Display text <#section-name-in-html>`__ > > This syntax can work in html, but isn't the one described in docutils, > and it also doesn't work on pdf. The following syntax should be used > instead: > > `Display text <Section Name_>`__ > > The usual toolchain doesn't mind this unusual syntax, but it causes > errors when trying to build using the not-yet-merged rst2pdf: > > ValueError: format not resolved, probably missing URL scheme or undefined destination target for 'Forcing%20Quiescent%20States' > > Fixes: ccc9971e2147 ("docs: rcu: convert some articles from html to ReST") > Fixes: c8cce10a62aa ("docs: Fix the reference labels in Locking.rst") > Fixes: e548cdeffcd8 ("docs-rst: convert kernel-locking to ReST") > Fixes: 7ddedebb03b7 ("ALSA: doc: ReSTize writing-an-alsa-driver document") > Signed-off-by: Nícolas F. R. A. Prado <nfraprado(a)protonmail.com> For the sound part: Reviewed-by: Takashi Iwai <tiwai(a)suse.de> thanks, Takashi

1 0

No sound from speakers, only headphones working
by Слава Банщиков 25 Dec '20

25 Dec '20

Here is the output from alsa-info.sh http://alsa-project.org/db/?f=ba0e450b747066e59c76642aa3c9f15a9a9f8a0b

1 0

[PATCH 01/18] ALSA: core: Use DIV_ROUND_UP() instead of open-coding it
by Lars-Peter Clausen 25 Dec '20

25 Dec '20

Use DIV_ROUND_UP() instead of open-coding it. This documents intent and makes it more clear what is going on for the casual reviewer. Generated using the following the Coccinelle semantic patch. // <smpl> @@ expression x, y; @@ -(((x) + (y) - 1) / (y)) +DIV_ROUND_UP(x, y) // </smpl> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/core/control.c | 4 ++-- sound/core/seq/seq_memory.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 3b44378b9dec..1571c7f7c43b 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -836,7 +836,7 @@ static void fill_remaining_elem_value(struct snd_ctl_elem_value *control, { size_t offset = value_sizes[info->type] * info->count; - offset = (offset + sizeof(u32) - 1) / sizeof(u32); + offset = DIV_ROUND_UP(offset, sizeof(u32)); memset32((u32 *)control->value.bytes.data + offset, pattern, sizeof(control->value) / sizeof(u32) - offset); } @@ -928,7 +928,7 @@ static int sanity_check_elem_value(struct snd_card *card, /* check whether the remaining area kept untouched */ offset = value_sizes[info->type] * info->count; - offset = (offset + sizeof(u32) - 1) / sizeof(u32); + offset = DIV_ROUND_UP(offset, sizeof(u32)); p = (u32 *)control->value.bytes.data + offset; for (; offset < sizeof(control->value) / sizeof(u32); offset++, p++) { if (*p != pattern) { diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index 65db1a7c77b7..e245bb6ba533 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -290,7 +290,7 @@ int snd_seq_event_dup(struct snd_seq_pool *pool, struct snd_seq_event *event, extlen = 0; if (snd_seq_ev_is_variable(event)) { extlen = event->data.ext.len & ~SNDRV_SEQ_EXT_MASK; - ncells = (extlen + sizeof(struct snd_seq_event) - 1) / sizeof(struct snd_seq_event); + ncells = DIV_ROUND_UP(extlen, sizeof(struct snd_seq_event)); } if (ncells >= pool->total_elements) return -ENOMEM; -- 2.20.1

2 18

[PATCH] ALSA: pcm: use bit field on snd_pcm
by Kuninori Morimoto 25 Dec '20

25 Dec '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_pcm's internal, nonatomic, no_device_suspend are defined as bool. But we can reduce struct size if we use bit field for these. This patch converts these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/pcm.h | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2336bf9243e1..e836741d2708 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -520,12 +520,13 @@ struct snd_pcm { wait_queue_head_t open_wait; void *private_data; void (*private_free) (struct snd_pcm *pcm); - bool internal; /* pcm is for internal use only */ - bool nonatomic; /* whole PCM operations are in non-atomic context */ - bool no_device_suspend; /* don't invoke device PM suspend */ #if IS_ENABLED(CONFIG_SND_PCM_OSS) struct snd_pcm_oss oss; #endif + /* bit field */ + unsigned int internal:1; /* pcm is for internal use only */ + unsigned int nonatomic:1; /* whole PCM operations are in non-atomic context */ + unsigned int no_device_suspend:1; /* don't invoke device PM suspend */ }; /* -- 2.25.1

2 1

Fwd: question about alsa tracepoints and alsa debugging
by Bert Schiettecatte 25 Dec '20

25 Dec '20

Hi all, I have a rk3288 board I am working with, it has two ADCs and one DAC. The ADCs are connected via an FPGA to the rk3288. The DAC is directly connected to the I2S port of the rk3288. The rk3288 generates the master clock. I am using buildroot and busybox and kernel 5.9.12 (stable). I have a driver which receives the ADC data from the FPGA via SDIO. When I run speaker-test in the background and use arecord to record simultaneously in the foreground, both arecord and speaker-test seem to lock up. When I run speaker-test alone I do not have a problem. The playback device uses the rockchip_i2s.c driver while the capture device uses my SDIO driver. My driver based on dw_mmc.c calls snd_pcm_period_elapsed() for each buffer of frames that is ready. I can see that this is being called for a while when I start arecord in the foreground, until the lockup occurs. The moment I start arecord, the console output from speaker-test stops, so it's locked up by starting arecord. Previously this was working fine on kernel 4.11. However, since I updated to kernel 5.9.12 this no longer works. Has anything changed between 4.11 and 5.9.12 that could be the cause of the problem? commands used for starting playback / trace capture and for recording - ./perf record -F 99 -g -e snd_pcm:* -e sched:sched_switch -e sched:sched_wakeup -e irq:* speaker-test -D hw:CARD=ak4458ak5558aud,DEV=0 -c8 -r192000 -FS32_LE -tsine & arecord -Dhw:CARD=ak4458ak5558aud,DEV=1 -fS32_LE -c16 -r192000 /mnt/ramdisk/rec.wav I enabled the kernel options as described at https://www.kernel.org/doc/html/v5.9/sound/designs/tracepoints.html and have used perf to record a trace, which can be access here - https://kernel-debugging.s3-us-west-1.amazonaws.com/perf.data.script.211220… (exported using "perf script") The trace file shows "lockdep_recursion" at some point so I am wondering if my problem has to do with a deadlock somehow being caused. However, I also see handle_mm_fault, and do_page_fault, which has to do with memory mapping? So I'm not sure what the problem exactly is and what to do next to find out what's wrong. Thanks in advance for your help and interest and patience, Bert

3 6

[PATCH] ASoC: rt1015: reset cali_done when suspending
by Tzung-Bi Shih 25 Dec '20

25 Dec '20

Assuming the calibration state gets lost after system suspend. Resets cali_done when suspending. The rt1015_priv is alloced by kzalloc. No need to initialize cali_done to 0 in component probe callback. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- The idea inspired from previous discussion: (https://mailman.alsa-project.org/pipermail/alsa-devel/2020-December/178229.…) The device could lose the calibration state after suspend/resume. It is better to reset the flag and get chance to calibrate again. The patch applies after the series: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-December/178608.… sound/soc/codecs/rt1015.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 4a9e2eaebe30..2dcb7b0fba60 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -1055,7 +1055,6 @@ static int rt1015_probe(struct snd_soc_component *component) rt1015->component = component; rt1015->bclk_ratio = 0; - rt1015->cali_done = 0; INIT_DELAYED_WORK(&rt1015->flush_work, rt1015_flush_work); @@ -1103,6 +1102,7 @@ static int rt1015_suspend(struct snd_soc_component *component) regcache_cache_only(rt1015->regmap, true); regcache_mark_dirty(rt1015->regmap); + rt1015->cali_done = 0; return 0; } -- 2.29.2.729.g45daf8777d-goog

1 1

[PATCH] ALSA: usb-audio: Fix UBSAN warnings for MIDI jacks
by Takashi Iwai 23 Dec '20

23 Dec '20

The calculation of in_cables and out_cables bitmaps are done with the bit shift by the value from the descriptor, which is an arbitrary value, and can lead to UBSAN shift-out-of-bounds warnings. Fix it by filtering the bad descriptor values with the check of the upper bound 0x10 (the cable bitmaps are 16 bits). Reported-by: syzbot+92e45ae45543f89e8c88(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/midi.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index c8213652470c..0c23fa6d8525 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1889,6 +1889,8 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ms_ep = find_usb_ms_endpoint_descriptor(hostep); if (!ms_ep) continue; + if (ms_ep->bNumEmbMIDIJack > 0x10) + continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { if (++epidx >= MIDI_MAX_ENDPOINTS) { @@ -2141,6 +2143,8 @@ static int snd_usbmidi_detect_roland(struct snd_usb_midi *umidi, cs_desc[1] == USB_DT_CS_INTERFACE && cs_desc[2] == 0xf1 && cs_desc[3] == 0x02) { + if (cs_desc[4] > 0x10 || cs_desc[5] > 0x10) + continue; endpoint->in_cables = (1 << cs_desc[4]) - 1; endpoint->out_cables = (1 << cs_desc[5]) - 1; return snd_usbmidi_detect_endpoints(umidi, endpoint, 1); -- 2.26.2

1 0

Dell 0x0a58 change name
by Kailang 23 Dec '20

23 Dec '20

Hi Takashi, Dell want change platform name to 'Dell' only. BR, Kailang

2 1

Lenovo C940 speaker
by Kailang 23 Dec '20

23 Dec '20

Hi Takashi, C940 make i2s Amp(none Realtek) verb in BIOS verb table. It will enable internal speaker. This patch will solve it can't adjust speaker volume issue. BR, Kailang

2 1

[PATCH AUTOSEL 5.4 057/130] ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 93db51d06b32227319dae2ac289029ccf1b33181 ] The current driver code assumes blindly that all found sample rates for the same endpoint from the UAC2 and UAC3 descriptors can be used no matter which altsetting, but actually this was wrong: some devices accept only limited sample rates in each altsetting. For determining which altsetting supports which rate, we need to verify each sample rate and check the validity via UAC2_AS_VAL_ALT_SETTINGS. This control reports back the available altsettings as a bitmap. This patch implements the missing piece above, the verification and reconstructs the sample rate tables based on the result. An open question is how to deal with the altsettings that ended up with no valid sample rates after verification. At least, there is a device that showed this problem although the sample rates did work in the later usage (see bug link). For now, we accept such an altset as is, assuming that it's a firmware bug. Reported-by: Dylan Robinson <dylan_robinson(a)motu.com> Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1178203 Link: https://lore.kernel.org/r/20201123085347.19667-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/clock.c | 106 ++++++++++++++++++++++++++------------------- sound/usb/clock.h | 4 ++ sound/usb/format.c | 93 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 158 insertions(+), 45 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b118cf97607f3..960174485a531 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -560,16 +560,60 @@ static int get_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, return le32_to_cpu(data); } +/* + * Try to set the given sample rate: + * + * Return 0 if the clock source is read-only, the actual rate on success, + * or a negative error code. + * + * This function gets called from format.c to validate each sample rate, too. + * Hence no message is shown upon error + */ +int snd_usb_set_sample_rate_v2v3(struct snd_usb_audio *chip, + const struct audioformat *fmt, + int clock, int rate) +{ + bool writeable; + u32 bmControls; + __le32 data; + int err; + + if (fmt->protocol == UAC_VERSION_3) { + struct uac3_clock_source_descriptor *cs_desc; + + cs_desc = snd_usb_find_clock_source_v3(chip->ctrl_intf, clock); + bmControls = le32_to_cpu(cs_desc->bmControls); + } else { + struct uac_clock_source_descriptor *cs_desc; + + cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, clock); + bmControls = cs_desc->bmControls; + } + + writeable = uac_v2v3_control_is_writeable(bmControls, + UAC2_CS_CONTROL_SAM_FREQ); + if (!writeable) + return 0; + + data = cpu_to_le32(rate); + err = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + &data, sizeof(data)); + if (err < 0) + return err; + + return get_sample_rate_v2v3(chip, fmt->iface, fmt->altsetting, clock); +} + static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, - struct usb_host_interface *alts, - struct audioformat *fmt, int rate) + struct usb_host_interface *alts, + struct audioformat *fmt, int rate) { struct usb_device *dev = chip->dev; - __le32 data; - int err, cur_rate, prev_rate; + int cur_rate, prev_rate; int clock; - bool writeable; - u32 bmControls; /* First, try to find a valid clock. This may trigger * automatic clock selection if the current clock is not @@ -592,50 +636,22 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, if (prev_rate == rate) goto validation; - if (fmt->protocol == UAC_VERSION_3) { - struct uac3_clock_source_descriptor *cs_desc; - - cs_desc = snd_usb_find_clock_source_v3(chip->ctrl_intf, clock); - bmControls = le32_to_cpu(cs_desc->bmControls); - } else { - struct uac_clock_source_descriptor *cs_desc; - - cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, clock); - bmControls = cs_desc->bmControls; + cur_rate = snd_usb_set_sample_rate_v2v3(chip, fmt, clock, rate); + if (cur_rate < 0) { + usb_audio_err(chip, + "%d:%d: cannot set freq %d (v2/v3): err %d\n", + iface, fmt->altsetting, rate, cur_rate); + return cur_rate; } - writeable = uac_v2v3_control_is_writeable(bmControls, - UAC2_CS_CONTROL_SAM_FREQ); - if (writeable) { - data = cpu_to_le32(rate); - err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, - UAC2_CS_CONTROL_SAM_FREQ << 8, - snd_usb_ctrl_intf(chip) | (clock << 8), - &data, sizeof(data)); - if (err < 0) { - usb_audio_err(chip, - "%d:%d: cannot set freq %d (v2/v3): err %d\n", - iface, fmt->altsetting, rate, err); - return err; - } - - cur_rate = get_sample_rate_v2v3(chip, iface, - fmt->altsetting, clock); - } else { + if (!cur_rate) cur_rate = prev_rate; - } if (cur_rate != rate) { - if (!writeable) { - usb_audio_warn(chip, - "%d:%d: freq mismatch (RO clock): req %d, clock runs @%d\n", - iface, fmt->altsetting, rate, cur_rate); - return -ENXIO; - } - usb_audio_dbg(chip, - "current rate %d is different from the runtime rate %d\n", - 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; } /* Some devices doesn't respond to sample rate changes while the diff --git a/sound/usb/clock.h b/sound/usb/clock.h index 68df0fbe09d00..97597f5a3c18a 100644 --- a/sound/usb/clock.h +++ b/sound/usb/clock.h @@ -9,4 +9,8 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, int snd_usb_clock_find_source(struct snd_usb_audio *chip, struct audioformat *fmt, bool validate); +int snd_usb_set_sample_rate_v2v3(struct snd_usb_audio *chip, + const struct audioformat *fmt, + int clock, int rate); + #endif /* __USBAUDIO_CLOCK_H */ diff --git a/sound/usb/format.c b/sound/usb/format.c index 1f9ea513230a6..442df4578182d 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -367,6 +367,97 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, return -ENODEV; } +/* check whether the given altsetting is supported for the already set rate */ +static bool check_valid_altsetting_v2v3(struct snd_usb_audio *chip, int iface, + int altsetting) +{ + struct usb_device *dev = chip->dev; + __le64 raw_data = 0; + u64 data; + int err; + + /* we assume 64bit is enough for any altsettings */ + if (snd_BUG_ON(altsetting >= 64 - 8)) + return false; + + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_AS_VAL_ALT_SETTINGS << 8, + iface, &raw_data, sizeof(raw_data)); + if (err < 0) + return false; + + data = le64_to_cpu(raw_data); + /* first byte contains the bitmap size */ + if ((data & 0xff) * 8 < altsetting) + return false; + if (data & (1ULL << (altsetting + 8))) + return true; + + return false; +} + +/* + * Validate each sample rate with the altsetting + * Rebuild the rate table if only partial values are valid + */ +static int validate_sample_rate_table_v2v3(struct snd_usb_audio *chip, + struct audioformat *fp, + int clock) +{ + struct usb_device *dev = chip->dev; + unsigned int *table; + unsigned int nr_rates; + unsigned int rate_min = 0x7fffffff; + unsigned int rate_max = 0; + unsigned int rates = 0; + int i, err; + + table = kcalloc(fp->nr_rates, sizeof(*table), GFP_KERNEL); + if (!table) + return -ENOMEM; + + /* clear the interface altsetting at first */ + usb_set_interface(dev, fp->iface, 0); + + nr_rates = 0; + for (i = 0; i < fp->nr_rates; i++) { + err = snd_usb_set_sample_rate_v2v3(chip, fp, clock, + fp->rate_table[i]); + if (err < 0) + continue; + + if (check_valid_altsetting_v2v3(chip, fp->iface, fp->altsetting)) { + table[nr_rates++] = fp->rate_table[i]; + if (rate_min > fp->rate_table[i]) + rate_min = fp->rate_table[i]; + if (rate_max < fp->rate_table[i]) + rate_max = fp->rate_table[i]; + rates |= snd_pcm_rate_to_rate_bit(fp->rate_table[i]); + } + } + + if (!nr_rates) { + usb_audio_dbg(chip, + "No valid sample rate available for %d:%d, assuming a firmware bug\n", + fp->iface, fp->altsetting); + nr_rates = fp->nr_rates; /* continue as is */ + } + + if (fp->nr_rates == nr_rates) { + kfree(table); + return 0; + } + + kfree(fp->rate_table); + fp->rate_table = table; + fp->nr_rates = nr_rates; + fp->rate_min = rate_min; + fp->rate_max = rate_max; + fp->rates = rates; + return 0; +} + /* * parse the format descriptor and stores the possible sample rates * on the audioformat table (audio class v2 and v3). @@ -459,6 +550,8 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, * allocated, so the rates will be stored */ parse_uac2_sample_rate_range(chip, fp, nr_triplets, data); + ret = validate_sample_rate_table_v2v3(chip, fp, clock); + err_free: kfree(data); err: -- 2.27.0

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Sdca support
by GitHub pull_request - opened 23 Dec '20

23 Dec '20

alsa-project/alsa-ucm-conf pull request #71 was opened from libinyang: Add supports for SDW sdca codec devices Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/71 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/71.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH AUTOSEL 4.4 37/38] ALSA: hda/hdmi: packet buffer index must be set before reading value
by Sasha Levin 23 Dec '20

23 Dec '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 46c3bbd9827952f92e250fa6ee30a797a4c4e17e ] The check for infoframe transmit status in hdmi_infoframe_uptodate() makes the assumption that packet buffer index is set to zero. Align code with specification and explicitly set the index before AC_VERB_GET_HDMI_DIP_XMIT. The packet index setting affects both DIP-Data and DIP-XmitCtrl verbs. There are no known cases where the old implementation has caused driver to work incorrectly. This change is purely based on code review against the specification (HDA spec rev1.0a). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20201211131613.3271407-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b249b1b857464..bab2b06c62b1c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1055,11 +1055,11 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, u8 val; int i; + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) != AC_DIPXMIT_BEST) return false; - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); for (i = 0; i < size; i++) { val = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_DATA, 0); -- 2.27.0

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[PATCH AUTOSEL 4.4 20/38] ALSA: usb-audio: Don't call usb_set_interface() at trigger callback
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4974b7950929e4a28d4eaee48e4ad07f168ac132 ] The PCM trigger callback is atomic, hence we must not call a function like usb_set_interface() there. Calling it from there would lead to a kernel Oops. Fix it by moving the usb_set_interface() call to set_sync_endpoint(). Also, apply the snd_usb_set_interface_quirk() for consistency, too. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/pcm.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 366813f1a5f80..59f3a9492d74b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -242,21 +242,6 @@ static int start_endpoints(struct snd_usb_substream *subs) !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { struct snd_usb_endpoint *ep = subs->sync_endpoint; - if (subs->data_endpoint->iface != subs->sync_endpoint->iface || - subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { - err = usb_set_interface(subs->dev, - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting); - if (err < 0) { - clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); - dev_err(&subs->dev->dev, - "%d:%d: cannot set interface (%d)\n", - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting, err); - return -EIO; - } - } - dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; @@ -489,6 +474,19 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, subs->data_endpoint->sync_master = subs->sync_endpoint; + if (subs->data_endpoint->iface != subs->sync_endpoint->iface || + subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { + err = usb_set_interface(subs->dev, + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + if (err < 0) + return err; + dev_dbg(&dev->dev, "setting usb interface %d:%d\n", + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + snd_usb_set_interface_quirk(dev); + } + return 0; } -- 2.27.0

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