From 28f194da4a2c4d431552025a4386edaffab181bd Mon Sep 17 00:00:00 2001 From: Jiri Slaby Date: Tue, 14 Sep 2021 11:11:22 +0200 Subject: tty: make tty_ldisc_ops::hangup return void The documentation says that the return value of tty_ldisc_ops::hangup hook is ignored. And it really is, so there is no point for its return type to be int. Switch it to void and all the hooks too. Cc: Dmitry Torokhov Cc: Wolfgang Grandegger Cc: Marc Kleine-Budde Cc: "David S. Miller" Cc: Jakub Kicinski Cc: Paul Mackerras Cc: Liam Girdwood Cc: Mark Brown Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: Peter Ujfalusi Acked-by: Dmitry Torokhov Acked-by: Mark Brown Acked-by: Marc Kleine-Budde Signed-off-by: Jiri Slaby Link: https://lore.kernel.org/r/20210914091134.17426-4-jslaby@suse.cz Signed-off-by: Greg Kroah-Hartman --- sound/soc/codecs/cx20442.c | 3 +-- sound/soc/ti/ams-delta.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 13258f3ca9aa..1af0bf5f1e2f 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -252,10 +252,9 @@ static void v253_close(struct tty_struct *tty) } /* Line discipline .hangup() */ -static int v253_hangup(struct tty_struct *tty) +static void v253_hangup(struct tty_struct *tty) { v253_close(tty); - return 0; } /* Line discipline .receive_buf() */ diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index ecd24d412a9b..b1a32545babd 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -330,10 +330,9 @@ static void cx81801_close(struct tty_struct *tty) } /* Line discipline .hangup() */ -static int cx81801_hangup(struct tty_struct *tty) +static void cx81801_hangup(struct tty_struct *tty) { cx81801_close(tty); - return 0; } /* Line discipline .receive_buf() */ -- cgit v1.2.3-71-gd317 From cd45c9bf8b43cd387e167cf166ae5c517f56d658 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Mon, 18 Oct 2021 16:33:22 +0200 Subject: ASoC: Intel: Move soc_intel_is_foo() helpers to a generic header The soc_intel_is_foo() helpers from sound/soc/intel/common/soc-intel-quirks.h are useful outside of the sound subsystem too. Move these to include/linux/platform_data/x86/soc.h, so that other code can use them too. Suggested-by: Andy Shevchenko Reviewed-by: Andy Shevchenko Acked-by: Mark Brown Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20211018143324.296961-2-hdegoede@redhat.com --- include/linux/platform_data/x86/soc.h | 65 +++++++++++++++++++++++++++++++ sound/soc/intel/common/soc-intel-quirks.h | 51 ++---------------------- 2 files changed, 68 insertions(+), 48 deletions(-) create mode 100644 include/linux/platform_data/x86/soc.h (limited to 'sound') diff --git a/include/linux/platform_data/x86/soc.h b/include/linux/platform_data/x86/soc.h new file mode 100644 index 000000000000..da05f425587a --- /dev/null +++ b/include/linux/platform_data/x86/soc.h @@ -0,0 +1,65 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Helpers for Intel SoC model detection + * + * Copyright (c) 2019, Intel Corporation. + */ + +#ifndef __PLATFORM_DATA_X86_SOC_H +#define __PLATFORM_DATA_X86_SOC_H + +#if IS_ENABLED(CONFIG_X86) + +#include +#include + +#define SOC_INTEL_IS_CPU(soc, type) \ +static inline bool soc_intel_is_##soc(void) \ +{ \ + static const struct x86_cpu_id soc##_cpu_ids[] = { \ + X86_MATCH_INTEL_FAM6_MODEL(type, NULL), \ + {} \ + }; \ + const struct x86_cpu_id *id; \ + \ + id = x86_match_cpu(soc##_cpu_ids); \ + if (id) \ + return true; \ + return false; \ +} + +SOC_INTEL_IS_CPU(byt, ATOM_SILVERMONT); +SOC_INTEL_IS_CPU(cht, ATOM_AIRMONT); +SOC_INTEL_IS_CPU(apl, ATOM_GOLDMONT); +SOC_INTEL_IS_CPU(glk, ATOM_GOLDMONT_PLUS); +SOC_INTEL_IS_CPU(cml, KABYLAKE_L); + +#else /* IS_ENABLED(CONFIG_X86) */ + +static inline bool soc_intel_is_byt(void) +{ + return false; +} + +static inline bool soc_intel_is_cht(void) +{ + return false; +} + +static inline bool soc_intel_is_apl(void) +{ + return false; +} + +static inline bool soc_intel_is_glk(void) +{ + return false; +} + +static inline bool soc_intel_is_cml(void) +{ + return false; +} +#endif /* IS_ENABLED(CONFIG_X86) */ + +#endif /* __PLATFORM_DATA_X86_SOC_H */ diff --git a/sound/soc/intel/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h index a93987ab7f4d..de4e550c5b34 100644 --- a/sound/soc/intel/common/soc-intel-quirks.h +++ b/sound/soc/intel/common/soc-intel-quirks.h @@ -9,34 +9,13 @@ #ifndef _SND_SOC_INTEL_QUIRKS_H #define _SND_SOC_INTEL_QUIRKS_H +#include + #if IS_ENABLED(CONFIG_X86) #include -#include -#include #include -#define SOC_INTEL_IS_CPU(soc, type) \ -static inline bool soc_intel_is_##soc(void) \ -{ \ - static const struct x86_cpu_id soc##_cpu_ids[] = { \ - X86_MATCH_INTEL_FAM6_MODEL(type, NULL), \ - {} \ - }; \ - const struct x86_cpu_id *id; \ - \ - id = x86_match_cpu(soc##_cpu_ids); \ - if (id) \ - return true; \ - return false; \ -} - -SOC_INTEL_IS_CPU(byt, ATOM_SILVERMONT); -SOC_INTEL_IS_CPU(cht, ATOM_AIRMONT); -SOC_INTEL_IS_CPU(apl, ATOM_GOLDMONT); -SOC_INTEL_IS_CPU(glk, ATOM_GOLDMONT_PLUS); -SOC_INTEL_IS_CPU(cml, KABYLAKE_L); - static inline bool soc_intel_is_byt_cr(struct platform_device *pdev) { /* @@ -114,30 +93,6 @@ static inline bool soc_intel_is_byt_cr(struct platform_device *pdev) return false; } -static inline bool soc_intel_is_byt(void) -{ - return false; -} - -static inline bool soc_intel_is_cht(void) -{ - return false; -} - -static inline bool soc_intel_is_apl(void) -{ - return false; -} - -static inline bool soc_intel_is_glk(void) -{ - return false; -} - -static inline bool soc_intel_is_cml(void) -{ - return false; -} #endif - #endif /* _SND_SOC_INTEL_QUIRKS_H */ +#endif /* _SND_SOC_INTEL_QUIRKS_H */ -- cgit v1.2.3-71-gd317 From f4ff6b56bc8ab2fcad6885813cd28ccc81224981 Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Mon, 18 Oct 2021 12:31:04 +0200 Subject: ASoC: cirrus: i2s: Prepare clock before using it Use clk_prepare_enable()/clk_disable_unprepare() in preparation for switch to Common Clock Framework, otherwise the following is visible: WARNING: CPU: 0 PID: 97 at drivers/clk/clk.c:1011 clk_core_enable+0x9c/0xbc Enabling unprepared mclk ... Hardware name: Cirrus Logic EDB9302 Evaluation Board ... clk_core_enable clk_core_enable_lock ep93xx_i2s_hw_params snd_soc_dai_hw_params soc_pcm_hw_params snd_pcm_hw_params snd_pcm_ioctl ... Signed-off-by: Alexander Sverdlin Acked-by: Mark Brown Link: https://lore.kernel.org/r/20211018103105.146380-2-alexander.sverdlin@gmail.com' Signed-off-by: Arnd Bergmann --- sound/soc/cirrus/ep93xx-i2s.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 0d26550d0df8..4d3179f03202 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -111,9 +111,9 @@ static void ep93xx_i2s_enable(struct ep93xx_i2s_info *info, int stream) if ((ep93xx_i2s_read_reg(info, EP93XX_I2S_TX0EN) & 0x1) == 0 && (ep93xx_i2s_read_reg(info, EP93XX_I2S_RX0EN) & 0x1) == 0) { /* Enable clocks */ - clk_enable(info->mclk); - clk_enable(info->sclk); - clk_enable(info->lrclk); + clk_prepare_enable(info->mclk); + clk_prepare_enable(info->sclk); + clk_prepare_enable(info->lrclk); /* Enable i2s */ ep93xx_i2s_write_reg(info, EP93XX_I2S_GLCTRL, 1); @@ -156,9 +156,9 @@ static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream) ep93xx_i2s_write_reg(info, EP93XX_I2S_GLCTRL, 0); /* Disable clocks */ - clk_disable(info->lrclk); - clk_disable(info->sclk); - clk_disable(info->mclk); + clk_disable_unprepare(info->lrclk); + clk_disable_unprepare(info->sclk); + clk_disable_unprepare(info->mclk); } } -- cgit v1.2.3-71-gd317 From 4f66a9ef37d3c09917a1edc065ff68b895e0b163 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Nov 2021 08:59:44 +0100 Subject: ALSA: hda: intel: More comprehensive PM runtime setup for controller driver Currently we haven't explicitly enable and allow/forbid the runtime PM at the probe and the remove phases of HD-audio controller driver, and this was the reason of a GPF mentioned in the commit e81478bbe7a1 ("ALSA: hda: fix general protection fault in azx_runtime_idle"); namely, even after the resources are released, the runtime PM might be still invoked by the bound graphics driver during the remove of the controller driver. Although we've fixed it by clearing the drvdata reference, it'd be also better to cover the runtime PM issue more properly. This patch adds a few more pm_runtime_*() calls at the probe and the remove time for setting and cleaning up the runtime PM. Particularly, now more explicitly pm_runtime_enable() and _disable() get called as well as pm_runtime_forbid() call at the remove callback, so that a use-after-free should be avoided. Reported-by: Kai Vehmanen Reviewed-by: Kai Vehmanen Tested-by: Kai Vehmanen Link: https://lore.kernel.org/r/20211110210307.1172004-1-kai.vehmanen@linux.intel.com Link: https://lore.kernel.org/r/20211115075944.6972-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fe51163f2d82..45e85180048c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1347,8 +1347,14 @@ static void azx_free(struct azx *chip) if (hda->freed) return; - if (azx_has_pm_runtime(chip) && chip->running) + if (azx_has_pm_runtime(chip) && chip->running) { pm_runtime_get_noresume(&pci->dev); + pm_runtime_disable(&pci->dev); + pm_runtime_set_suspended(&pci->dev); + pm_runtime_forbid(&pci->dev); + pm_runtime_dont_use_autosuspend(&pci->dev); + } + chip->running = 0; azx_del_card_list(chip); @@ -2322,6 +2328,8 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); pm_runtime_allow(&pci->dev); + pm_runtime_set_active(&pci->dev); + pm_runtime_enable(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- cgit v1.2.3-71-gd317 From fd23116d7b8dffa05f42a857eee6ee9cce238d24 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Nov 2021 07:54:13 +0100 Subject: ALSA: usb-audio: Use int for dB map values The values in usbmix_dB_map should be rather signed while we're using u32. As the copied target (usb_mixer_elem_info.dBmin and dBmax) is int, let's make them also int. Link: https://lore.kernel.org/r/20211116065415.11159-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer_maps.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 55eea90ee993..92c06b1bb979 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -6,8 +6,8 @@ */ struct usbmix_dB_map { - u32 min; - u32 max; + int min; + int max; }; struct usbmix_name_map { -- cgit v1.2.3-71-gd317 From 85b741c1cb6854478fd1aa13ac231e2c1baf4c4b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Nov 2021 07:54:14 +0100 Subject: ALSA: usb-audio: Add minimal-mute notion in dB mapping table Some devices do mute the volume at the minimal volume, and for such devices, we need to set SNDRV_CTL_TLVT_DB_MINMAX_MUTE to the TLV information. It corresponds to setting usb_mixer_elem_info.min_mute flag in the USB-audio driver. This patch adds a new field min_mute in usbmix_dB_map so that the mixer map entry can pass the flag. Link: https://lore.kernel.org/r/20211116065415.11159-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 1 + sound/usb/mixer_maps.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6e7bac8203ba..5b9fd07ce2a2 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -145,6 +145,7 @@ static inline void check_mapped_dB(const struct usbmix_name_map *p, if (p && p->dB) { cval->dBmin = p->dB->min; cval->dBmax = p->dB->max; + cval->min_mute = p->dB->min_mute; cval->initialized = 1; } } diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 92c06b1bb979..9d71c569b148 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -8,6 +8,7 @@ struct usbmix_dB_map { int min; int max; + bool min_mute; }; struct usbmix_name_map { -- cgit v1.2.3-71-gd317 From 02eb1d098e26f34c8f047b0b1cee6f4433a34bd1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Nov 2021 07:54:15 +0100 Subject: ALSA: usb-audio: Fix dB level of Bose Revolve+ SoundLink Bose Revolve+ SoundLink (0a57:40fa) advertises invalid dB level for the speaker volume. This patch provides the correction in the mixer map quirk table entry. Note that this requires the prerequisite change to add min_mute flag to the dB map table. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1192375 Link: https://lore.kernel.org/r/20211116065415.11159-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer_maps.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 9d71c569b148..5d391f62351b 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -337,6 +337,13 @@ static const struct usbmix_name_map bose_companion5_map[] = { { 0 } /* terminator */ }; +/* Bose Revolve+ SoundLink, correction of dB maps */ +static const struct usbmix_dB_map bose_soundlink_dB = {-8283, -0, true}; +static const struct usbmix_name_map bose_soundlink_map[] = { + { 2, NULL, .dB = &bose_soundlink_dB }, + { 0 } /* terminator */ +}; + /* Sennheiser Communications Headset [PC 8], the dB value is reported as -6 negative maximum */ static const struct usbmix_dB_map sennheiser_pc8_dB = {-9500, 0}; static const struct usbmix_name_map sennheiser_pc8_map[] = { @@ -522,6 +529,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x05a7, 0x1020), .map = bose_companion5_map, }, + { + /* Bose Revolve+ SoundLink */ + .id = USB_ID(0x05a7, 0x40fa), + .map = bose_soundlink_map, + }, { /* Corsair Virtuoso SE (wired mode) */ .id = USB_ID(0x1b1c, 0x0a3d), -- cgit v1.2.3-71-gd317 From 06764dc931848c3a9bc01a63bbf76a605408bb54 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Nov 2021 08:13:12 +0100 Subject: ALSA: jack: Add missing rwsem around snd_ctl_remove() calls snd_ctl_remove() has to be called with card->controls_rwsem held (when called after the card instantiation). This patch add the missing rwsem calls around it. Fixes: 9058cbe1eed2 ("ALSA: jack: implement kctl creating for jack devices") Link: https://lore.kernel.org/r/20211116071314.15065-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/jack.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/core/jack.c b/sound/core/jack.c index 32350c6aba84..f50a1e920e1d 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -62,10 +62,13 @@ static int snd_jack_dev_free(struct snd_device *device) struct snd_card *card = device->card; struct snd_jack_kctl *jack_kctl, *tmp_jack_kctl; + down_write(&card->controls_rwsem); list_for_each_entry_safe(jack_kctl, tmp_jack_kctl, &jack->kctl_list, list) { list_del_init(&jack_kctl->list); snd_ctl_remove(card, jack_kctl->kctl); } + up_write(&card->controls_rwsem); + if (jack->private_free) jack->private_free(jack); -- cgit v1.2.3-71-gd317 From 5471e9762e1af4b7df057a96bfd46cc250979b88 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Nov 2021 08:13:13 +0100 Subject: ALSA: PCM: Add missing rwsem around snd_ctl_remove() calls snd_ctl_remove() has to be called with card->controls_rwsem held (when called after the card instantiation). This patch add the missing rwsem calls around it. Fixes: a8ff48cb7083 ("ALSA: pcm: Free chmap at PCM free callback, too") Link: https://lore.kernel.org/r/20211116071314.15065-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 6fd3677685d7..ba4a987ed1c6 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -810,7 +810,11 @@ EXPORT_SYMBOL(snd_pcm_new_internal); static void free_chmap(struct snd_pcm_str *pstr) { if (pstr->chmap_kctl) { - snd_ctl_remove(pstr->pcm->card, pstr->chmap_kctl); + struct snd_card *card = pstr->pcm->card; + + down_write(&card->controls_rwsem); + snd_ctl_remove(card, pstr->chmap_kctl); + up_write(&card->controls_rwsem); pstr->chmap_kctl = NULL; } } -- cgit v1.2.3-71-gd317 From 80bd64af75b4bb11c0329bc66c35da2ddfb66d88 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Nov 2021 08:13:14 +0100 Subject: ALSA: hda: Add missing rwsem around snd_ctl_remove() calls snd_ctl_remove() has to be called with card->controls_rwsem held (when called after the card instantiation). This patch add the missing rwsem calls around it. Fixes: d13bd412dce2 ("ALSA: hda - Manage kcontrol lists") Link: https://lore.kernel.org/r/20211116071314.15065-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0c4a337c9fc0..eda70814369b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1727,8 +1727,11 @@ void snd_hda_ctls_clear(struct hda_codec *codec) { int i; struct hda_nid_item *items = codec->mixers.list; + + down_write(&codec->card->controls_rwsem); for (i = 0; i < codec->mixers.used; i++) snd_ctl_remove(codec->card, items[i].kctl); + up_write(&codec->card->controls_rwsem); snd_array_free(&codec->mixers); snd_array_free(&codec->nids); } -- cgit v1.2.3-71-gd317 From 7206998f578d5553989bc01ea2e544b622e79539 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Nov 2021 08:24:59 +0100 Subject: ALSA: hda: Fix potential deadlock at codec unbinding When a codec is unbound dynamically via sysfs while its stream is in use, we may face a potential deadlock at the proc remove or a UAF. This happens since the hda_pcm is managed by a linked list, as it handles the hda_pcm object release via kref. When a PCM is opened at the unbinding time, the release of hda_pcm gets delayed and it ends up with the close of the PCM stream releasing the associated hda_pcm object of its own. The hda_pcm destructor contains the PCM device release that includes the removal of procfs entries. And, this removal has the sync of the close of all in-use files -- which would never finish because it's called from the PCM file descriptor itself, i.e. it's trying to shoot its foot. For addressing the deadlock above, this patch changes the way to manage and release the hda_pcm object. The kref of hda_pcm is dropped, and instead a simple refcount is introduced in hda_codec for keeping the track of the active PCM streams, and at each PCM open and close, this refcount is adjusted accordingly. At unbinding, the driver calls snd_device_disconnect() for each PCM stream, then synchronizes with the refcount finish, and finally releases the object resources. Fixes: bbbc7e8502c9 ("ALSA: hda - Allocate hda_pcm objects dynamically") Link: https://lore.kernel.org/r/20211116072459.18930-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/hda_codec.h | 8 +++++--- sound/pci/hda/hda_bind.c | 5 +++++ sound/pci/hda/hda_codec.c | 42 ++++++++++++++++++++++++++---------------- sound/pci/hda/hda_local.h | 1 + 4 files changed, 37 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 0e45963bb767..82d9daa17851 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -8,7 +8,7 @@ #ifndef __SOUND_HDA_CODEC_H #define __SOUND_HDA_CODEC_H -#include +#include #include #include #include @@ -166,8 +166,8 @@ struct hda_pcm { bool own_chmap; /* codec driver provides own channel maps */ /* private: */ struct hda_codec *codec; - struct kref kref; struct list_head list; + unsigned int disconnected:1; }; /* codec information */ @@ -187,6 +187,8 @@ struct hda_codec { /* PCM to create, set by patch_ops.build_pcms callback */ struct list_head pcm_list_head; + refcount_t pcm_ref; + wait_queue_head_t remove_sleep; /* codec specific info */ void *spec; @@ -420,7 +422,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec); static inline void snd_hda_codec_pcm_get(struct hda_pcm *pcm) { - kref_get(&pcm->kref); + refcount_inc(&pcm->codec->pcm_ref); } void snd_hda_codec_pcm_put(struct hda_pcm *pcm); diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 1c8bffc3eec6..7153bd53e189 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -156,6 +156,11 @@ static int hda_codec_driver_remove(struct device *dev) return codec->bus->core.ext_ops->hdev_detach(&codec->core); } + refcount_dec(&codec->pcm_ref); + snd_hda_codec_disconnect_pcms(codec); + wait_event(codec->remove_sleep, !refcount_read(&codec->pcm_ref)); + snd_power_sync_ref(codec->bus->card); + if (codec->patch_ops.free) codec->patch_ops.free(codec); snd_hda_codec_cleanup_for_unbind(codec); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index eda70814369b..7016b48227bf 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -703,20 +703,10 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid) /* * PCM device */ -static void release_pcm(struct kref *kref) -{ - struct hda_pcm *pcm = container_of(kref, struct hda_pcm, kref); - - if (pcm->pcm) - snd_device_free(pcm->codec->card, pcm->pcm); - clear_bit(pcm->device, pcm->codec->bus->pcm_dev_bits); - kfree(pcm->name); - kfree(pcm); -} - void snd_hda_codec_pcm_put(struct hda_pcm *pcm) { - kref_put(&pcm->kref, release_pcm); + if (refcount_dec_and_test(&pcm->codec->pcm_ref)) + wake_up(&pcm->codec->remove_sleep); } EXPORT_SYMBOL_GPL(snd_hda_codec_pcm_put); @@ -731,7 +721,6 @@ struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec, return NULL; pcm->codec = codec; - kref_init(&pcm->kref); va_start(args, fmt); pcm->name = kvasprintf(GFP_KERNEL, fmt, args); va_end(args); @@ -741,6 +730,7 @@ struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec, } list_add_tail(&pcm->list, &codec->pcm_list_head); + refcount_inc(&codec->pcm_ref); return pcm; } EXPORT_SYMBOL_GPL(snd_hda_codec_pcm_new); @@ -748,15 +738,31 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_pcm_new); /* * codec destructor */ +void snd_hda_codec_disconnect_pcms(struct hda_codec *codec) +{ + struct hda_pcm *pcm; + + list_for_each_entry(pcm, &codec->pcm_list_head, list) { + if (pcm->disconnected) + continue; + if (pcm->pcm) + snd_device_disconnect(codec->card, pcm->pcm); + snd_hda_codec_pcm_put(pcm); + pcm->disconnected = 1; + } +} + static void codec_release_pcms(struct hda_codec *codec) { struct hda_pcm *pcm, *n; list_for_each_entry_safe(pcm, n, &codec->pcm_list_head, list) { - list_del_init(&pcm->list); + list_del(&pcm->list); if (pcm->pcm) - snd_device_disconnect(codec->card, pcm->pcm); - snd_hda_codec_pcm_put(pcm); + snd_device_free(pcm->codec->card, pcm->pcm); + clear_bit(pcm->device, pcm->codec->bus->pcm_dev_bits); + kfree(pcm->name); + kfree(pcm); } } @@ -769,6 +775,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) codec->registered = 0; } + snd_hda_codec_disconnect_pcms(codec); cancel_delayed_work_sync(&codec->jackpoll_work); if (!codec->in_freeing) snd_hda_ctls_clear(codec); @@ -792,6 +799,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) remove_conn_list(codec); snd_hdac_regmap_exit(&codec->core); codec->configured = 0; + refcount_set(&codec->pcm_ref, 1); /* reset refcount */ } EXPORT_SYMBOL_GPL(snd_hda_codec_cleanup_for_unbind); @@ -958,6 +966,8 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); INIT_LIST_HEAD(&codec->conn_list); INIT_LIST_HEAD(&codec->pcm_list_head); + refcount_set(&codec->pcm_ref, 1); + init_waitqueue_head(&codec->remove_sleep); INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); codec->depop_delay = -1; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index ea8ab8b43337..4662a47add7e 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -137,6 +137,7 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, int snd_hda_codec_reset(struct hda_codec *codec); void snd_hda_codec_register(struct hda_codec *codec); void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec); +void snd_hda_codec_disconnect_pcms(struct hda_codec *codec); #define snd_hda_regmap_sync(codec) snd_hdac_regmap_sync(&(codec)->core) -- cgit v1.2.3-71-gd317 From 2c95b92ecd92e784785b1db8cccc4f0f2bfa850c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Nov 2021 08:33:58 +0100 Subject: ALSA: memalloc: Unify x86 SG-buffer handling (take#3) This is a second attempt to unify the x86-specific SG-buffer handling code with the new standard non-contiguous page handler. The first try (in commit 2d9ea39917a4) failed due to the wrong page and address calculations, hence reverted. (And the second try failed due to a copy&paste error.) Now it's corrected with the previous fix for noncontig pages, and the proper sg page iteration by this patch. After the migration, SNDRV_DMA_TYPE_DMA_SG becomes identical with SNDRV_DMA_TYPE_NONCONTIG on x86, while others still fall back to SNDRV_DMA_TYPE_DEV. Tested-by: Alex Xu (Hello71) Tested-by: Harald Arnesen Link: https://lore.kernel.org/r/20211017074859.24112-4-tiwai@suse.de Link: https://lore.kernel.org/r/20211109062235.22310-1-tiwai@suse.de Link: https://lore.kernel.org/r/20211116073358.19741-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/memalloc.h | 14 ++-- sound/core/Makefile | 1 - sound/core/memalloc.c | 53 ++++++++++++- sound/core/sgbuf.c | 201 ----------------------------------------------- 4 files changed, 56 insertions(+), 213 deletions(-) delete mode 100644 sound/core/sgbuf.c (limited to 'sound') diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 1051b84e8579..653dfffb3ac8 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -36,13 +36,6 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_CONTINUOUS 1 /* continuous no-DMA memory */ #define SNDRV_DMA_TYPE_DEV 2 /* generic device continuous */ #define SNDRV_DMA_TYPE_DEV_WC 5 /* continuous write-combined */ -#ifdef CONFIG_SND_DMA_SGBUF -#define SNDRV_DMA_TYPE_DEV_SG 3 /* generic device SG-buffer */ -#define SNDRV_DMA_TYPE_DEV_WC_SG 6 /* SG write-combined */ -#else -#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ -#define SNDRV_DMA_TYPE_DEV_WC_SG SNDRV_DMA_TYPE_DEV_WC -#endif #ifdef CONFIG_GENERIC_ALLOCATOR #define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ #else @@ -51,6 +44,13 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_VMALLOC 7 /* vmalloc'ed buffer */ #define SNDRV_DMA_TYPE_NONCONTIG 8 /* non-coherent SG buffer */ #define SNDRV_DMA_TYPE_NONCOHERENT 9 /* non-coherent buffer */ +#ifdef CONFIG_SND_DMA_SGBUF +#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_NONCONTIG +#define SNDRV_DMA_TYPE_DEV_WC_SG 6 /* SG write-combined */ +#else +#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ +#define SNDRV_DMA_TYPE_DEV_WC_SG SNDRV_DMA_TYPE_DEV_WC +#endif /* * info for buffer allocation diff --git a/sound/core/Makefile b/sound/core/Makefile index 79e1407cd0de..350d704ced98 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -19,7 +19,6 @@ snd-$(CONFIG_SND_JACK) += ctljack.o jack.o snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_misc.o \ pcm_memory.o memalloc.o snd-pcm-$(CONFIG_SND_PCM_TIMER) += pcm_timer.o -snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o snd-pcm-$(CONFIG_SND_PCM_ELD) += pcm_drm_eld.o snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 9fc971a704a9..d1fcd1d5adae 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -620,6 +620,52 @@ static const struct snd_malloc_ops snd_dma_noncontig_ops = { .get_chunk_size = snd_dma_noncontig_get_chunk_size, }; +/* x86-specific SG-buffer with WC pages */ +#ifdef CONFIG_SND_DMA_SGBUF +#define sg_wc_address(it) ((unsigned long)page_address(sg_page_iter_page(it))) + +static void *snd_dma_sg_wc_alloc(struct snd_dma_buffer *dmab, size_t size) +{ + void *p = snd_dma_noncontig_alloc(dmab, size); + struct sg_table *sgt = dmab->private_data; + struct sg_page_iter iter; + + if (!p) + return NULL; + for_each_sgtable_page(sgt, &iter, 0) + set_memory_wc(sg_wc_address(&iter), 1); + return p; +} + +static void snd_dma_sg_wc_free(struct snd_dma_buffer *dmab) +{ + struct sg_table *sgt = dmab->private_data; + struct sg_page_iter iter; + + for_each_sgtable_page(sgt, &iter, 0) + set_memory_wb(sg_wc_address(&iter), 1); + snd_dma_noncontig_free(dmab); +} + +static int snd_dma_sg_wc_mmap(struct snd_dma_buffer *dmab, + struct vm_area_struct *area) +{ + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return dma_mmap_noncontiguous(dmab->dev.dev, area, + dmab->bytes, dmab->private_data); +} + +static const struct snd_malloc_ops snd_dma_sg_wc_ops = { + .alloc = snd_dma_sg_wc_alloc, + .free = snd_dma_sg_wc_free, + .mmap = snd_dma_sg_wc_mmap, + .sync = snd_dma_noncontig_sync, + .get_addr = snd_dma_noncontig_get_addr, + .get_page = snd_dma_noncontig_get_page, + .get_chunk_size = snd_dma_noncontig_get_chunk_size, +}; +#endif /* CONFIG_SND_DMA_SGBUF */ + /* * Non-coherent pages allocator */ @@ -679,14 +725,13 @@ static const struct snd_malloc_ops *dma_ops[] = { [SNDRV_DMA_TYPE_DEV_WC] = &snd_dma_wc_ops, [SNDRV_DMA_TYPE_NONCONTIG] = &snd_dma_noncontig_ops, [SNDRV_DMA_TYPE_NONCOHERENT] = &snd_dma_noncoherent_ops, +#ifdef CONFIG_SND_DMA_SGBUF + [SNDRV_DMA_TYPE_DEV_WC_SG] = &snd_dma_sg_wc_ops, +#endif #ifdef CONFIG_GENERIC_ALLOCATOR [SNDRV_DMA_TYPE_DEV_IRAM] = &snd_dma_iram_ops, #endif /* CONFIG_GENERIC_ALLOCATOR */ #endif /* CONFIG_HAS_DMA */ -#ifdef CONFIG_SND_DMA_SGBUF - [SNDRV_DMA_TYPE_DEV_SG] = &snd_dma_sg_ops, - [SNDRV_DMA_TYPE_DEV_WC_SG] = &snd_dma_sg_ops, -#endif }; static const struct snd_malloc_ops *snd_dma_get_ops(struct snd_dma_buffer *dmab) diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c deleted file mode 100644 index 8352a5cdb19f..000000000000 --- a/sound/core/sgbuf.c +++ /dev/null @@ -1,201 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* - * Scatter-Gather buffer - * - * Copyright (c) by Takashi Iwai - */ - -#include -#include -#include -#include -#include -#include "memalloc_local.h" - -struct snd_sg_page { - void *buf; - dma_addr_t addr; -}; - -struct snd_sg_buf { - int size; /* allocated byte size */ - int pages; /* allocated pages */ - int tblsize; /* allocated table size */ - struct snd_sg_page *table; /* address table */ - struct page **page_table; /* page table (for vmap/vunmap) */ - struct device *dev; -}; - -/* table entries are align to 32 */ -#define SGBUF_TBL_ALIGN 32 -#define sgbuf_align_table(tbl) ALIGN((tbl), SGBUF_TBL_ALIGN) - -static void snd_dma_sg_free(struct snd_dma_buffer *dmab) -{ - struct snd_sg_buf *sgbuf = dmab->private_data; - struct snd_dma_buffer tmpb; - int i; - - if (!sgbuf) - return; - - vunmap(dmab->area); - dmab->area = NULL; - - tmpb.dev.type = SNDRV_DMA_TYPE_DEV; - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) - tmpb.dev.type = SNDRV_DMA_TYPE_DEV_WC; - tmpb.dev.dev = sgbuf->dev; - for (i = 0; i < sgbuf->pages; i++) { - if (!(sgbuf->table[i].addr & ~PAGE_MASK)) - continue; /* continuous pages */ - tmpb.area = sgbuf->table[i].buf; - tmpb.addr = sgbuf->table[i].addr & PAGE_MASK; - tmpb.bytes = (sgbuf->table[i].addr & ~PAGE_MASK) << PAGE_SHIFT; - snd_dma_free_pages(&tmpb); - } - - kfree(sgbuf->table); - kfree(sgbuf->page_table); - kfree(sgbuf); - dmab->private_data = NULL; -} - -#define MAX_ALLOC_PAGES 32 - -static void *snd_dma_sg_alloc(struct snd_dma_buffer *dmab, size_t size) -{ - struct snd_sg_buf *sgbuf; - unsigned int i, pages, chunk, maxpages; - struct snd_dma_buffer tmpb; - struct snd_sg_page *table; - struct page **pgtable; - int type = SNDRV_DMA_TYPE_DEV; - pgprot_t prot = PAGE_KERNEL; - void *area; - - dmab->private_data = sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); - if (!sgbuf) - return NULL; - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) { - type = SNDRV_DMA_TYPE_DEV_WC; -#ifdef pgprot_noncached - prot = pgprot_noncached(PAGE_KERNEL); -#endif - } - sgbuf->dev = dmab->dev.dev; - pages = snd_sgbuf_aligned_pages(size); - sgbuf->tblsize = sgbuf_align_table(pages); - table = kcalloc(sgbuf->tblsize, sizeof(*table), GFP_KERNEL); - if (!table) - goto _failed; - sgbuf->table = table; - pgtable = kcalloc(sgbuf->tblsize, sizeof(*pgtable), GFP_KERNEL); - if (!pgtable) - goto _failed; - sgbuf->page_table = pgtable; - - /* allocate pages */ - maxpages = MAX_ALLOC_PAGES; - while (pages > 0) { - chunk = pages; - /* don't be too eager to take a huge chunk */ - if (chunk > maxpages) - chunk = maxpages; - chunk <<= PAGE_SHIFT; - if (snd_dma_alloc_pages_fallback(type, dmab->dev.dev, - chunk, &tmpb) < 0) { - if (!sgbuf->pages) - goto _failed; - size = sgbuf->pages * PAGE_SIZE; - break; - } - chunk = tmpb.bytes >> PAGE_SHIFT; - for (i = 0; i < chunk; i++) { - table->buf = tmpb.area; - table->addr = tmpb.addr; - if (!i) - table->addr |= chunk; /* mark head */ - table++; - *pgtable++ = virt_to_page(tmpb.area); - tmpb.area += PAGE_SIZE; - tmpb.addr += PAGE_SIZE; - } - sgbuf->pages += chunk; - pages -= chunk; - if (chunk < maxpages) - maxpages = chunk; - } - - sgbuf->size = size; - area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, prot); - if (!area) - goto _failed; - return area; - - _failed: - snd_dma_sg_free(dmab); /* free the table */ - return NULL; -} - -static dma_addr_t snd_dma_sg_get_addr(struct snd_dma_buffer *dmab, - size_t offset) -{ - struct snd_sg_buf *sgbuf = dmab->private_data; - dma_addr_t addr; - - addr = sgbuf->table[offset >> PAGE_SHIFT].addr; - addr &= ~((dma_addr_t)PAGE_SIZE - 1); - return addr + offset % PAGE_SIZE; -} - -static struct page *snd_dma_sg_get_page(struct snd_dma_buffer *dmab, - size_t offset) -{ - struct snd_sg_buf *sgbuf = dmab->private_data; - unsigned int idx = offset >> PAGE_SHIFT; - - if (idx >= (unsigned int)sgbuf->pages) - return NULL; - return sgbuf->page_table[idx]; -} - -static unsigned int snd_dma_sg_get_chunk_size(struct snd_dma_buffer *dmab, - unsigned int ofs, - unsigned int size) -{ - struct snd_sg_buf *sg = dmab->private_data; - unsigned int start, end, pg; - - start = ofs >> PAGE_SHIFT; - end = (ofs + size - 1) >> PAGE_SHIFT; - /* check page continuity */ - pg = sg->table[start].addr >> PAGE_SHIFT; - for (;;) { - start++; - if (start > end) - break; - pg++; - if ((sg->table[start].addr >> PAGE_SHIFT) != pg) - return (start << PAGE_SHIFT) - ofs; - } - /* ok, all on continuous pages */ - return size; -} - -static int snd_dma_sg_mmap(struct snd_dma_buffer *dmab, - struct vm_area_struct *area) -{ - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) - area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); - return -ENOENT; /* continue with the default mmap handler */ -} - -const struct snd_malloc_ops snd_dma_sg_ops = { - .alloc = snd_dma_sg_alloc, - .free = snd_dma_sg_free, - .get_addr = snd_dma_sg_get_addr, - .get_page = snd_dma_sg_get_page, - .get_chunk_size = snd_dma_sg_get_chunk_size, - .mmap = snd_dma_sg_mmap, -}; -- cgit v1.2.3-71-gd317 From 37c4fd0db7c961145d9d1909ecab386fdf703c26 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 17 Nov 2021 14:30:40 +0100 Subject: ALSA: hda: Do disconnect jacks at codec unbind The HD-audio codec driver remove may happen also at dynamically unbinding during operation, hence it needs manual triggers of snd_device_disconnect() calls, while it's missing for the jack objects that are associated with the codec. This patch adds the manual disconnection call for jacks when the remove happens without card->shutdown (i.e. not under the full removal). Link: https://lore.kernel.org/r/20211117133040.20272-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_bind.c | 2 ++ sound/pci/hda/hda_jack.c | 11 +++++++++++ sound/pci/hda/hda_jack.h | 1 + 3 files changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 7153bd53e189..c572fb5886d5 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -14,6 +14,7 @@ #include #include #include "hda_local.h" +#include "hda_jack.h" /* * find a matching codec id @@ -158,6 +159,7 @@ static int hda_codec_driver_remove(struct device *dev) refcount_dec(&codec->pcm_ref); snd_hda_codec_disconnect_pcms(codec); + snd_hda_jack_tbl_disconnect(codec); wait_event(codec->remove_sleep, !refcount_read(&codec->pcm_ref)); snd_power_sync_ref(codec->bus->card); diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index f29975e3e98d..7d7786df60ea 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -158,6 +158,17 @@ snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid, int dev_id) return jack; } +void snd_hda_jack_tbl_disconnect(struct hda_codec *codec) +{ + struct hda_jack_tbl *jack = codec->jacktbl.list; + int i; + + for (i = 0; i < codec->jacktbl.used; i++, jack++) { + if (!codec->bus->shutdown && jack->jack) + snd_device_disconnect(codec->card, jack->jack); + } +} + void snd_hda_jack_tbl_clear(struct hda_codec *codec) { struct hda_jack_tbl *jack = codec->jacktbl.list; diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 2abf7aac243a..ff7d289c034b 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -69,6 +69,7 @@ struct hda_jack_tbl * snd_hda_jack_tbl_get_from_tag(struct hda_codec *codec, unsigned char tag, int dev_id); +void snd_hda_jack_tbl_disconnect(struct hda_codec *codec); void snd_hda_jack_tbl_clear(struct hda_codec *codec); void snd_hda_jack_set_dirty_all(struct hda_codec *codec); -- cgit v1.2.3-71-gd317 From de2f29c4394efa64c3a5ba1b15302eb558ed4c56 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 19 Nov 2021 17:27:30 +0100 Subject: ALSA: hda: Remove redundant runtime PM calls The previous fix for more comprehensive runtime PM calls turned out to be not good as hoped; a few calls including pm_runtime_enable() and pm_runtime_disable() are rather utterly superfluous for PCI devices, even triggering a kernel error message. Better to drop those calls. Note that the problem we wanted to solve with that commit seems irrelevant with the fix itself; the original bug (a GPF at azx_remove()) was likely a regression by the recent PCI core cleanup, and the buggy PCI change has been already reverted. So basically we were scratching a wrong surface. OTOH, making the runtime PM calls symmetric for both probe and remove is more consistent, and maybe that's a sensible outcome. Fixes: 4f66a9ef37d3 ("ALSA: hda: intel: More comprehensive PM runtime setup for controller driver") Reported-by: Heiner Kallweit Link: https://lore.kernel.org/r/d9d76980-966a-e031-70d1-3254ba5be5eb@gmail.com Link: https://lore.kernel.org/r/20211119162730.24423-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 45e85180048c..221afacbc7fd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1349,8 +1349,6 @@ static void azx_free(struct azx *chip) if (azx_has_pm_runtime(chip) && chip->running) { pm_runtime_get_noresume(&pci->dev); - pm_runtime_disable(&pci->dev); - pm_runtime_set_suspended(&pci->dev); pm_runtime_forbid(&pci->dev); pm_runtime_dont_use_autosuspend(&pci->dev); } @@ -2328,8 +2326,6 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); pm_runtime_allow(&pci->dev); - pm_runtime_set_active(&pci->dev); - pm_runtime_enable(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- cgit v1.2.3-71-gd317 From 7c72665c5667d4566e594ea362c50a6007b405fb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Nov 2021 18:02:47 +0100 Subject: ALSA: led: Use restricted type for iface assignment Fix a sparse warning that complains about the inconsistent type assignment for iface, which is a restricted type of snd_ctl_elem_iface_t. Fixes: a135dfb5de15 ("ALSA: led control - add sysfs kcontrol LED marking layer") Reported-by: kernel test robot Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/202111201028.xduVYgH5-lkp@intel.com Link: https://lore.kernel.org/r/20211123170247.2962-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/control_led.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/control_led.c b/sound/core/control_led.c index a95332b2b90b..207828f30983 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -509,7 +509,7 @@ static char *parse_string(char *s, char *val, size_t val_size) return s; } -static char *parse_iface(char *s, unsigned int *val) +static char *parse_iface(char *s, snd_ctl_elem_iface_t *val) { if (!strncasecmp(s, "card", 4)) *val = SNDRV_CTL_ELEM_IFACE_CARD; -- cgit v1.2.3-71-gd317 From 6dd21ad81bf96478db3403b1bbe251c0612d0431 Mon Sep 17 00:00:00 2001 From: Thomas Gleixner Date: Wed, 24 Nov 2021 23:40:01 +0100 Subject: ALSA: hda: Make proper use of timecounter HDA uses a timecounter to read a hardware clock running at 24 MHz. The conversion factor is set with a mult value of 125 and a shift value of 0, which is not converting the hardware clock to nanoseconds, it is converting to 1/3 nanoseconds because the conversion factor from 24Mhz to nanoseconds is 125/3. The usage sites divide the "nanoseconds" value returned by timecounter_read() by 3 to get a real nanoseconds value. There is a lengthy comment in azx_timecounter_init() explaining this choice. That comment makes blatantly wrong assumptions about how timecounters work and what can overflow. The comment says: * Applying the 1/3 factor as part of the multiplication * requires at least 20 bits for a decent precision, however * overflows occur after about 4 hours or less, not a option. timecounters operate on time deltas between two readouts of a clock and use the mult/shift pair to calculate a precise nanoseconds value: delta_nsec = (delta_clock * mult) >> shift; The fractional part is also taken into account and preserved to prevent accumulated rounding errors. For details see cyclecounter_cyc2ns(). The mult/shift pair has to be chosen so that the multiplication of the maximum expected delta value does not result in a 64bit overflow. As the counter wraps around on 32bit, the maximum observable delta between two reads is (1 << 32) - 1 which is about 178.9 seconds. That in turn means the maximum multiplication factor which fits into an u32 will not cause a 64bit overflow ever because it's guaranteed that: ((1 << 32) - 1) ^ 2 < (1 << 64) The resulting correct multiplication factor is 2796202667 and the shift value is 26, i.e. 26 bit precision. The overflow of the multiplication would happen exactly at a clock readout delta of 6597069765 which is way after the wrap around of the hardware clock at around 274.8 seconds which is off from the claimed 4 hours by more than an order of magnitude. If the counter ever wraps around the last read value then the calculation is off by the number of wrap arounds times 178.9 seconds because the overflow cannot be observed. Use clocks_calc_mult_shift(), which calculates the most accurate mult/shift pair based on the given clock frequency, and remove the bogus comment along with the divisions at the readout sites. Fixes: 5d890f591d15 ("ALSA: hda: support for wallclock timestamps") Signed-off-by: Thomas Gleixner Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/871r35kwji.ffs@tglx Signed-off-by: Takashi Iwai --- sound/hda/hdac_stream.c | 14 ++++---------- sound/pci/hda/hda_controller.c | 1 - sound/soc/intel/skylake/skl-pcm.c | 1 - 3 files changed, 4 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index 9867555883c3..aa7955fdf68a 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -534,17 +534,11 @@ static void azx_timecounter_init(struct hdac_stream *azx_dev, cc->mask = CLOCKSOURCE_MASK(32); /* - * Converting from 24 MHz to ns means applying a 125/3 factor. - * To avoid any saturation issues in intermediate operations, - * the 125 factor is applied first. The division is applied - * last after reading the timecounter value. - * Applying the 1/3 factor as part of the multiplication - * requires at least 20 bits for a decent precision, however - * overflows occur after about 4 hours or less, not a option. + * Calculate the optimal mult/shift values. The counter wraps + * around after ~178.9 seconds. */ - - cc->mult = 125; /* saturation after 195 years */ - cc->shift = 0; + clocks_calc_mult_shift(&cc->mult, &cc->shift, 24000000, + NSEC_PER_SEC, 178); nsec = 0; /* audio time is elapsed time since trigger */ timecounter_init(tc, cc, nsec); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 930ae4002a81..75dcb14ff20a 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -504,7 +504,6 @@ static int azx_get_time_info(struct snd_pcm_substream *substream, snd_pcm_gettime(substream->runtime, system_ts); nsec = timecounter_read(&azx_dev->core.tc); - nsec = div_u64(nsec, 3); /* can be optimized */ if (audio_tstamp_config->report_delay) nsec = azx_adjust_codec_delay(substream, nsec); diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 9ecaf6a1e847..e4aa366d356e 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1251,7 +1251,6 @@ static int skl_platform_soc_get_time_info( snd_pcm_gettime(substream->runtime, system_ts); nsec = timecounter_read(&hstr->tc); - nsec = div_u64(nsec, 3); /* can be optimized */ if (audio_tstamp_config->report_delay) nsec = skl_adjust_codec_delay(substream, nsec); -- cgit v1.2.3-71-gd317 From 15fa179f3f45415696d376abc84e0098a9586b33 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 26 Nov 2021 15:03:53 +0100 Subject: ALSA: hda: Fill gaps in NHLT endpoint-interface MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Two key operations missings are: endpoint presence-check and retrieval of matching endpoint hardware configuration (blob). Add operations for both use cases. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20211126140355.1042684-2-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- include/sound/intel-nhlt.h | 37 ++++++++++++---- sound/hda/intel-nhlt.c | 102 +++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 131 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/include/sound/intel-nhlt.h b/include/sound/intel-nhlt.h index d0574805865f..089a760d36eb 100644 --- a/include/sound/intel-nhlt.h +++ b/include/sound/intel-nhlt.h @@ -10,6 +10,14 @@ #include +enum nhlt_link_type { + NHLT_LINK_HDA = 0, + NHLT_LINK_DSP = 1, + NHLT_LINK_DMIC = 2, + NHLT_LINK_SSP = 3, + NHLT_LINK_INVALID +}; + #if IS_ENABLED(CONFIG_ACPI) && IS_ENABLED(CONFIG_SND_INTEL_NHLT) struct wav_fmt { @@ -33,14 +41,6 @@ struct wav_fmt_ext { u8 sub_fmt[16]; } __packed; -enum nhlt_link_type { - NHLT_LINK_HDA = 0, - NHLT_LINK_DSP = 1, - NHLT_LINK_DMIC = 2, - NHLT_LINK_SSP = 3, - NHLT_LINK_INVALID -}; - enum nhlt_device_type { NHLT_DEVICE_BT = 0, NHLT_DEVICE_DMIC = 1, @@ -132,6 +132,12 @@ void intel_nhlt_free(struct nhlt_acpi_table *addr); int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt); +bool intel_nhlt_has_endpoint_type(struct nhlt_acpi_table *nhlt, u8 link_type); +struct nhlt_specific_cfg * +intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, + u32 bus_id, u8 link_type, u8 vbps, u8 bps, + u8 num_ch, u32 rate, u8 dir, u8 dev_type); + #else struct nhlt_acpi_table; @@ -150,6 +156,21 @@ static inline int intel_nhlt_get_dmic_geo(struct device *dev, { return 0; } + +static inline bool intel_nhlt_has_endpoint_type(struct nhlt_acpi_table *nhlt, + u8 link_type) +{ + return false; +} + +static inline struct nhlt_specific_cfg * +intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, + u32 bus_id, u8 link_type, u8 vbps, u8 bps, + u8 num_ch, u32 rate, u8 dir, u8 dev_type) +{ + return NULL; +} + #endif #endif diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index e2237239d922..128476aa7c61 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -110,3 +110,105 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); + +bool intel_nhlt_has_endpoint_type(struct nhlt_acpi_table *nhlt, u8 link_type) +{ + struct nhlt_endpoint *epnt; + int i; + + if (!nhlt) + return false; + + epnt = (struct nhlt_endpoint *)nhlt->desc; + for (i = 0; i < nhlt->endpoint_count; i++) { + if (epnt->linktype == link_type) + return true; + + epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); + } + return false; +} +EXPORT_SYMBOL(intel_nhlt_has_endpoint_type); + +static struct nhlt_specific_cfg * +nhlt_get_specific_cfg(struct device *dev, struct nhlt_fmt *fmt, u8 num_ch, + u32 rate, u8 vbps, u8 bps) +{ + struct nhlt_fmt_cfg *cfg = fmt->fmt_config; + struct wav_fmt *wfmt; + u16 _bps, _vbps; + int i; + + dev_dbg(dev, "Endpoint format count=%d\n", fmt->fmt_count); + + for (i = 0; i < fmt->fmt_count; i++) { + wfmt = &cfg->fmt_ext.fmt; + _bps = wfmt->bits_per_sample; + _vbps = cfg->fmt_ext.sample.valid_bits_per_sample; + + dev_dbg(dev, "Endpoint format: ch=%d fmt=%d/%d rate=%d\n", + wfmt->channels, _vbps, _bps, wfmt->samples_per_sec); + + if (wfmt->channels == num_ch && wfmt->samples_per_sec == rate && + vbps == _vbps && bps == _bps) + return &cfg->config; + + cfg = (struct nhlt_fmt_cfg *)(cfg->config.caps + cfg->config.size); + } + + return NULL; +} + +static bool nhlt_check_ep_match(struct device *dev, struct nhlt_endpoint *epnt, + u32 bus_id, u8 link_type, u8 dir, u8 dev_type) +{ + dev_dbg(dev, "Endpoint: vbus_id=%d link_type=%d dir=%d dev_type = %d\n", + epnt->virtual_bus_id, epnt->linktype, + epnt->direction, epnt->device_type); + + if ((epnt->virtual_bus_id != bus_id) || + (epnt->linktype != link_type) || + (epnt->direction != dir)) + return false; + + /* link of type DMIC bypasses device_type check */ + return epnt->linktype == NHLT_LINK_DMIC || + epnt->device_type == dev_type; +} + +struct nhlt_specific_cfg * +intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, + u32 bus_id, u8 link_type, u8 vbps, u8 bps, + u8 num_ch, u32 rate, u8 dir, u8 dev_type) +{ + struct nhlt_specific_cfg *cfg; + struct nhlt_endpoint *epnt; + struct nhlt_fmt *fmt; + int i; + + if (!nhlt) + return NULL; + + dev_dbg(dev, "Looking for configuration:\n"); + dev_dbg(dev, " vbus_id=%d link_type=%d dir=%d, dev_type=%d\n", + bus_id, link_type, dir, dev_type); + dev_dbg(dev, " ch=%d fmt=%d/%d rate=%d\n", num_ch, vbps, bps, rate); + dev_dbg(dev, "Endpoint count=%d\n", nhlt->endpoint_count); + + epnt = (struct nhlt_endpoint *)nhlt->desc; + + for (i = 0; i < nhlt->endpoint_count; i++) { + if (nhlt_check_ep_match(dev, epnt, bus_id, link_type, dir, dev_type)) { + fmt = (struct nhlt_fmt *)(epnt->config.caps + epnt->config.size); + + cfg = nhlt_get_specific_cfg(dev, fmt, num_ch, rate, vbps, bps); + if (cfg) + return cfg; + } + + epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); + } + + return NULL; +} +EXPORT_SYMBOL(intel_nhlt_get_endpoint_blob); -- cgit v1.2.3-71-gd317 From 8235a08bbc6be993c5de1de1f5d7a07110831248 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 26 Nov 2021 15:03:54 +0100 Subject: ALSA: hda: Simplify DMIC-in-NHLT check MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Only DMIC endpoint presence is relevant, not its configuration. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20211126140355.1042684-3-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/hda/intel-dsp-config.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index b9ac9e9e45a4..26f8665da689 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -384,7 +384,7 @@ static int snd_intel_dsp_check_dmic(struct pci_dev *pci) nhlt = intel_nhlt_init(&pci->dev); if (nhlt) { - if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) + if (intel_nhlt_has_endpoint_type(nhlt, NHLT_LINK_DMIC)) ret = 1; intel_nhlt_free(nhlt); } -- cgit v1.2.3-71-gd317 From 322fa4315400807c697b034b4694f0a074cc1258 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 26 Nov 2021 15:03:55 +0100 Subject: ASoC: Intel: Skylake: Use NHLT API to search for blob MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit With NHLT enriched with new search functions, remove local code in favour of them. This also fixes broken behaviour: search should be based on significant bits count rather than container size. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Acked-by: Mark Brown Link: https://lore.kernel.org/r/20211126140355.1042684-4-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/soc/intel/skylake/skl-nhlt.c | 102 --------------------------------- sound/soc/intel/skylake/skl-pcm.c | 3 + sound/soc/intel/skylake/skl-topology.c | 29 ++++++---- sound/soc/intel/skylake/skl-topology.h | 1 + sound/soc/intel/skylake/skl.h | 4 -- 5 files changed, 21 insertions(+), 118 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 64226072f0ee..2439a574ac2f 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -13,108 +13,6 @@ #include "skl.h" #include "skl-i2s.h" -static struct nhlt_specific_cfg *skl_get_specific_cfg( - struct device *dev, struct nhlt_fmt *fmt, - u8 no_ch, u32 rate, u16 bps, u8 linktype) -{ - struct nhlt_specific_cfg *sp_config; - struct wav_fmt *wfmt; - struct nhlt_fmt_cfg *fmt_config = fmt->fmt_config; - int i; - - dev_dbg(dev, "Format count =%d\n", fmt->fmt_count); - - for (i = 0; i < fmt->fmt_count; i++) { - wfmt = &fmt_config->fmt_ext.fmt; - dev_dbg(dev, "ch=%d fmt=%d s_rate=%d\n", wfmt->channels, - wfmt->bits_per_sample, wfmt->samples_per_sec); - if (wfmt->channels == no_ch && wfmt->bits_per_sample == bps) { - /* - * if link type is dmic ignore rate check as the blob is - * generic for all rates - */ - sp_config = &fmt_config->config; - if (linktype == NHLT_LINK_DMIC) - return sp_config; - - if (wfmt->samples_per_sec == rate) - return sp_config; - } - - fmt_config = (struct nhlt_fmt_cfg *)(fmt_config->config.caps + - fmt_config->config.size); - } - - return NULL; -} - -static void dump_config(struct device *dev, u32 instance_id, u8 linktype, - u8 s_fmt, u8 num_channels, u32 s_rate, u8 dirn, u16 bps) -{ - dev_dbg(dev, "Input configuration\n"); - dev_dbg(dev, "ch=%d fmt=%d s_rate=%d\n", num_channels, s_fmt, s_rate); - dev_dbg(dev, "vbus_id=%d link_type=%d\n", instance_id, linktype); - dev_dbg(dev, "bits_per_sample=%d\n", bps); -} - -static bool skl_check_ep_match(struct device *dev, struct nhlt_endpoint *epnt, - u32 instance_id, u8 link_type, u8 dirn, u8 dev_type) -{ - dev_dbg(dev, "vbus_id=%d link_type=%d dir=%d dev_type = %d\n", - epnt->virtual_bus_id, epnt->linktype, - epnt->direction, epnt->device_type); - - if ((epnt->virtual_bus_id == instance_id) && - (epnt->linktype == link_type) && - (epnt->direction == dirn)) { - /* do not check dev_type for DMIC link type */ - if (epnt->linktype == NHLT_LINK_DMIC) - return true; - - if (epnt->device_type == dev_type) - return true; - } - - return false; -} - -struct nhlt_specific_cfg -*skl_get_ep_blob(struct skl_dev *skl, u32 instance, u8 link_type, - u8 s_fmt, u8 num_ch, u32 s_rate, - u8 dirn, u8 dev_type) -{ - struct nhlt_fmt *fmt; - struct nhlt_endpoint *epnt; - struct hdac_bus *bus = skl_to_bus(skl); - struct device *dev = bus->dev; - struct nhlt_specific_cfg *sp_config; - struct nhlt_acpi_table *nhlt = skl->nhlt; - u16 bps = (s_fmt == 16) ? 16 : 32; - u8 j; - - dump_config(dev, instance, link_type, s_fmt, num_ch, s_rate, dirn, bps); - - epnt = (struct nhlt_endpoint *)nhlt->desc; - - dev_dbg(dev, "endpoint count =%d\n", nhlt->endpoint_count); - - for (j = 0; j < nhlt->endpoint_count; j++) { - if (skl_check_ep_match(dev, epnt, instance, link_type, - dirn, dev_type)) { - fmt = (struct nhlt_fmt *)(epnt->config.caps + - epnt->config.size); - sp_config = skl_get_specific_cfg(dev, fmt, num_ch, - s_rate, bps, link_type); - if (sp_config) - return sp_config; - } - - epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); - } - - return NULL; -} - static void skl_nhlt_trim_space(char *trim) { char *s = trim; diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index e4aa366d356e..4c5d209a67ba 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -317,6 +317,7 @@ static int skl_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(dai->dev, "dma_id=%d\n", dma_id); p_params.s_fmt = snd_pcm_format_width(params_format(params)); + p_params.s_cont = snd_pcm_format_physical_width(params_format(params)); p_params.ch = params_channels(params); p_params.s_freq = params_rate(params); p_params.host_dma_id = dma_id; @@ -405,6 +406,7 @@ static int skl_be_hw_params(struct snd_pcm_substream *substream, struct skl_pipe_params p_params = {0}; p_params.s_fmt = snd_pcm_format_width(params_format(params)); + p_params.s_cont = snd_pcm_format_physical_width(params_format(params)); p_params.ch = params_channels(params); p_params.s_freq = params_rate(params); p_params.stream = substream->stream; @@ -569,6 +571,7 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream, snd_soc_dai_set_tdm_slot(codec_dai, 0, stream_tag, 0, 0); p_params.s_fmt = snd_pcm_format_width(params_format(params)); + p_params.s_cont = snd_pcm_format_physical_width(params_format(params)); p_params.ch = params_channels(params); p_params.s_freq = params_rate(params); p_params.stream = substream->stream; diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 89e4231304dd..9bdf020a2b64 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -285,7 +285,7 @@ static int skl_tplg_update_be_blob(struct snd_soc_dapm_widget *w, { struct skl_module_cfg *m_cfg = w->priv; int link_type, dir; - u32 ch, s_freq, s_fmt; + u32 ch, s_freq, s_fmt, s_cont; struct nhlt_specific_cfg *cfg; u8 dev_type = skl_tplg_be_dev_type(m_cfg->dev_type); int fmt_idx = m_cfg->fmt_idx; @@ -301,7 +301,8 @@ static int skl_tplg_update_be_blob(struct snd_soc_dapm_widget *w, link_type = NHLT_LINK_DMIC; dir = SNDRV_PCM_STREAM_CAPTURE; s_freq = m_iface->inputs[0].fmt.s_freq; - s_fmt = m_iface->inputs[0].fmt.bit_depth; + s_fmt = m_iface->inputs[0].fmt.valid_bit_depth; + s_cont = m_iface->inputs[0].fmt.bit_depth; ch = m_iface->inputs[0].fmt.channels; break; @@ -310,12 +311,14 @@ static int skl_tplg_update_be_blob(struct snd_soc_dapm_widget *w, if (m_cfg->hw_conn_type == SKL_CONN_SOURCE) { dir = SNDRV_PCM_STREAM_PLAYBACK; s_freq = m_iface->outputs[0].fmt.s_freq; - s_fmt = m_iface->outputs[0].fmt.bit_depth; + s_fmt = m_iface->outputs[0].fmt.valid_bit_depth; + s_cont = m_iface->outputs[0].fmt.bit_depth; ch = m_iface->outputs[0].fmt.channels; } else { dir = SNDRV_PCM_STREAM_CAPTURE; s_freq = m_iface->inputs[0].fmt.s_freq; - s_fmt = m_iface->inputs[0].fmt.bit_depth; + s_fmt = m_iface->inputs[0].fmt.valid_bit_depth; + s_cont = m_iface->inputs[0].fmt.bit_depth; ch = m_iface->inputs[0].fmt.channels; } break; @@ -325,16 +328,17 @@ static int skl_tplg_update_be_blob(struct snd_soc_dapm_widget *w, } /* update the blob based on virtual bus_id and default params */ - cfg = skl_get_ep_blob(skl, m_cfg->vbus_id, link_type, - s_fmt, ch, s_freq, dir, dev_type); + cfg = intel_nhlt_get_endpoint_blob(skl->dev, skl->nhlt, m_cfg->vbus_id, + link_type, s_fmt, s_cont, ch, + s_freq, dir, dev_type); if (cfg) { m_cfg->formats_config[SKL_PARAM_INIT].caps_size = cfg->size; m_cfg->formats_config[SKL_PARAM_INIT].caps = (u32 *)&cfg->caps; } else { dev_err(skl->dev, "Blob NULL for id %x type %d dirn %d\n", m_cfg->vbus_id, link_type, dir); - dev_err(skl->dev, "PCM: ch %d, freq %d, fmt %d\n", - ch, s_freq, s_fmt); + dev_err(skl->dev, "PCM: ch %d, freq %d, fmt %d/%d\n", + ch, s_freq, s_fmt, s_cont); return -EIO; } @@ -1849,10 +1853,11 @@ static int skl_tplg_be_fill_pipe_params(struct snd_soc_dai *dai, pipe_fmt = &pipe->configs[pipe->pipe_config_idx].in_fmt; /* update the blob based on virtual bus_id*/ - cfg = skl_get_ep_blob(skl, mconfig->vbus_id, link_type, - pipe_fmt->bps, pipe_fmt->channels, - pipe_fmt->freq, pipe->direction, - dev_type); + cfg = intel_nhlt_get_endpoint_blob(dai->dev, skl->nhlt, + mconfig->vbus_id, link_type, + pipe_fmt->bps, params->s_cont, + pipe_fmt->channels, pipe_fmt->freq, + pipe->direction, dev_type); if (cfg) { mconfig->formats_config[SKL_PARAM_INIT].caps_size = cfg->size; mconfig->formats_config[SKL_PARAM_INIT].caps = (u32 *)&cfg->caps; diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index f0695b2ac5dd..22963634fbea 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -284,6 +284,7 @@ struct skl_pipe_params { u32 ch; u32 s_freq; u32 s_fmt; + u32 s_cont; u8 linktype; snd_pcm_format_t format; int link_index; diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 33ed274fc0cb..f55f8b3dbdc3 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -165,10 +165,6 @@ struct skl_dsp_ops { int skl_platform_unregister(struct device *dev); int skl_platform_register(struct device *dev); -struct nhlt_specific_cfg *skl_get_ep_blob(struct skl_dev *skl, u32 instance, - u8 link_type, u8 s_fmt, u8 num_ch, - u32 s_rate, u8 dirn, u8 dev_type); - int skl_nhlt_update_topology_bin(struct skl_dev *skl); int skl_init_dsp(struct skl_dev *skl); int skl_free_dsp(struct skl_dev *skl); -- cgit v1.2.3-71-gd317 From 8e7daf318d97f25e18b2fc7eb5909e34cd903575 Mon Sep 17 00:00:00 2001 From: Bixuan Cui Date: Wed, 1 Dec 2021 16:58:54 +0800 Subject: ALSA: oss: fix compile error when OSS_DEBUG is enabled Fix compile error when OSS_DEBUG is enabled: sound/core/oss/pcm_oss.c: In function 'snd_pcm_oss_set_trigger': sound/core/oss/pcm_oss.c:2055:10: error: 'substream' undeclared (first use in this function); did you mean 'csubstream'? pcm_dbg(substream->pcm, "pcm_oss: trigger = 0x%x\n", trigger); ^ Fixes: 61efcee8608c ("ALSA: oss: Use standard printk helpers") Signed-off-by: Bixuan Cui Link: https://lore.kernel.org/r/1638349134-110369-1-git-send-email-cuibixuan@linux.alibaba.com Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 82a818734a5f..bb37665ad3c2 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2052,7 +2052,7 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr int err, cmd; #ifdef OSS_DEBUG - pcm_dbg(substream->pcm, "pcm_oss: trigger = 0x%x\n", trigger); + pr_debug("pcm_oss: trigger = 0x%x\n", trigger); #endif psubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; -- cgit v1.2.3-71-gd317 From ce9778b7a0272f7c7e5bc33f537380a5d2aed6c7 Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Thu, 2 Dec 2021 15:33:35 +0800 Subject: ALSA: hda/hdmi: Consider ELD is invalid when no SAD is present There's a system that reports a bogus HDMI audio interface: $ cat eld#2.0 monitor_present 1 eld_valid 1 monitor_name connection_type DisplayPort eld_version [0x2] CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0xe430 product_id 0x690 port_id 0x0 support_hdcp 0 support_ai 0 audio_sync_delay 0 speakers [0xffff] FL/FR LFE FC RL/RR RC FLC/FRC RLC/RRC FLW/FRW FLH/FRH TC FCH sad_count 0 Since playing audio is not possible without SAD, also consider ELD is invalid for this case. Signed-off-by: Kai-Heng Feng Link: https://lore.kernel.org/r/20211202073338.1384768-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 65d2c5539919..33e5f1aa24f9 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1535,7 +1535,7 @@ static void update_eld(struct hda_codec *codec, } } - if (!eld->eld_valid || eld->eld_size <= 0) { + if (!eld->eld_valid || eld->eld_size <= 0 || eld->info.sad_count <= 0) { eld->eld_valid = false; eld->eld_size = 0; } -- cgit v1.2.3-71-gd317 From 1e583aef12aa74afd37c1418255cc4b74e023236 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 2 Dec 2021 09:38:33 +0100 Subject: ALSA: usb-audio: Drop superfluous '0' in Presonus Studio 1810c's ID The vendor ID of Presonus Studio 1810c had a superfluous '0' in its USB ID. Drop it. Fixes: 8dc5efe3d17c ("ALSA: usb-audio: Add support for Presonus Studio 1810c") Link: https://lore.kernel.org/r/20211202083833.17784-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/format.c | 2 +- sound/usb/mixer_quirks.c | 2 +- sound/usb/quirks.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/format.c b/sound/usb/format.c index f5e676a51b30..405dc0bf6678 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -375,7 +375,7 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, for (rate = min; rate <= max; rate += res) { /* Filter out invalid rates on Presonus Studio 1810c */ - if (chip->usb_id == USB_ID(0x0194f, 0x010c) && + if (chip->usb_id == USB_ID(0x194f, 0x010c) && !s1810c_valid_sample_rate(fp, rate)) goto skip_rate; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index d489c1de3bae..db194ad168d0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3254,7 +3254,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_rme_controls_create(mixer); break; - case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */ + case USB_ID(0x194f, 0x010c): /* Presonus Studio 1810c */ err = snd_sc1810_init_mixer(mixer); break; case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 64e1c20311ed..ab9f3da49941 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1290,7 +1290,7 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, if (chip->usb_id == USB_ID(0x0763, 0x2012)) return fasttrackpro_skip_setting_quirk(chip, iface, altno); /* presonus studio 1810c: skip altsets incompatible with device_setup */ - if (chip->usb_id == USB_ID(0x0194f, 0x010c)) + if (chip->usb_id == USB_ID(0x194f, 0x010c)) return s1810c_skip_setting_quirk(chip, iface, altno); -- cgit v1.2.3-71-gd317 From d13a8f6d8e01a17a9fe36029e346a1f029362c9e Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Sat, 4 Dec 2021 08:28:40 +0100 Subject: ALSA: Fix some typo Some comments and include guards are not consistent with the name of the file where they can be found. This is likely some typo or cut'n'paste issues. Signed-off-by: Christophe JAILLET Link: https://lore.kernel.org/r/7b2bcbda298f02a34d46d8b6593daaaed9a09a45.1638602790.git.christophe.jaillet@wanadoo.fr Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_pcm.c | 2 +- sound/pci/hda/hda_generic.h | 2 +- sound/soc/codecs/sta350.h | 2 +- sound/soc/codecs/tlv320aic26.h | 6 +++--- sound/usb/usx2y/usbusx2y.c | 2 +- 5 files changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c index 491de1a623cb..5fee8e89790f 100644 --- a/sound/pci/ac97/ac97_pcm.c +++ b/sound/pci/ac97/ac97_pcm.c @@ -231,7 +231,7 @@ static int set_spdif_rate(struct snd_ac97 *ac97, unsigned short rate) * If the codec doesn't support VAR, the rate must be 48000 (except * for SPDIF). * - * The valid registers are AC97_PMC_MIC_ADC_RATE, + * The valid registers are AC97_PCM_MIC_ADC_RATE, * AC97_PCM_FRONT_DAC_RATE, AC97_PCM_LR_ADC_RATE. * AC97_PCM_SURR_DAC_RATE and AC97_PCM_LFE_DAC_RATE are accepted * if the codec supports them. diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index c43bd0f0338e..8e1bc8ea74fc 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -183,7 +183,7 @@ struct hda_gen_spec { struct automic_entry am_entry[MAX_AUTO_MIC_PINS]; /* for pin sensing */ - /* current status; set in hda_geneic.c */ + /* current status; set in hda_generic.c */ unsigned int hp_jack_present:1; unsigned int line_jack_present:1; unsigned int speaker_muted:1; /* current status of speaker mute */ diff --git a/sound/soc/codecs/sta350.h b/sound/soc/codecs/sta350.h index f16900e00afa..80bf56093d94 100644 --- a/sound/soc/codecs/sta350.h +++ b/sound/soc/codecs/sta350.h @@ -14,7 +14,7 @@ #ifndef _ASOC_STA_350_H #define _ASOC_STA_350_H -/* STA50 register addresses */ +/* STA350 register addresses */ #define STA350_REGISTER_COUNT 0x4D #define STA350_COEF_COUNT 62 diff --git a/sound/soc/codecs/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h index 1f2879b7a080..c86569883e0c 100644 --- a/sound/soc/codecs/tlv320aic26.h +++ b/sound/soc/codecs/tlv320aic26.h @@ -6,8 +6,8 @@ * Copyright (C) 2008 Secret Lab Technologies Ltd. */ -#ifndef _TLV320AIC16_H_ -#define _TLV320AIC16_H_ +#ifndef _TLV320AIC26_H_ +#define _TLV320AIC26_H_ /* AIC26 Registers */ #define AIC26_PAGE_ADDR(page, offset) ((page << 11) | offset << 5) @@ -88,4 +88,4 @@ enum aic26_wlen { AIC26_WLEN_32 = 3 << 10, }; -#endif /* _TLV320AIC16_H_ */ +#endif /* _TLV320AIC26_H_ */ diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 099bee662af6..52f4e6652407 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-or-later /* - * usbusy2y.c - ALSA USB US-428 Driver + * usbusx2y.c - ALSA USB US-428 Driver * 2005-04-14 Karsten Wiese Version 0.8.7.2: -- cgit v1.2.3-71-gd317 From 82cd3ba691a920007503d189989d2495a41a3a10 Mon Sep 17 00:00:00 2001 From: Bernard Zhao Date: Sun, 5 Dec 2021 17:40:46 -0800 Subject: ALSA: oss: remove useless NULL check before kfree Tis patch try to remove useless NULL check before kfree Signed-off-by: Bernard Zhao Link: https://lore.kernel.org/r/20211206014135.320720-1-bernard@vivo.com Signed-off-by: Takashi Iwai --- sound/core/info_oss.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/info_oss.c b/sound/core/info_oss.c index 1ba887c7954e..ebc714b2f46b 100644 --- a/sound/core/info_oss.c +++ b/sound/core/info_oss.c @@ -32,10 +32,8 @@ int snd_oss_info_register(int dev, int num, char *string) mutex_lock(&strings); if (string == NULL) { x = snd_sndstat_strings[num][dev]; - if (x) { - kfree(x); - x = NULL; - } + kfree(x); + x = NULL; } else { x = kstrdup(string, GFP_KERNEL); if (x == NULL) { -- cgit v1.2.3-71-gd317 From 86a9bb5bf9f610ea6baa855b4f46ecea92876ea4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 2 Dec 2021 09:40:53 +0100 Subject: ALSA: usb-audio: Drop CONFIG_PM ifdefs Practically seen, CONFIG_PM is almost mandatory. Let's drop the ugly ifdef lines and simplify the code. Link: https://lore.kernel.org/r/20211202084053.18201-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/card.c | 7 ------- sound/usb/mixer.c | 4 ---- sound/usb/mixer.h | 2 -- sound/usb/mixer_quirks.c | 2 -- sound/usb/mixer_quirks.h | 2 -- sound/usb/power.h | 10 ---------- 6 files changed, 27 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 1764b9302d46..376962291c4d 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -987,8 +987,6 @@ void snd_usb_unlock_shutdown(struct snd_usb_audio *chip) wake_up(&chip->shutdown_wait); } -#ifdef CONFIG_PM - int snd_usb_autoresume(struct snd_usb_audio *chip) { int i, err; @@ -1100,11 +1098,6 @@ err_out: atomic_dec(&chip->active); /* allow autopm after this point */ return err; } -#else -#define usb_audio_suspend NULL -#define usb_audio_resume NULL -#define usb_audio_resume NULL -#endif /* CONFIG_PM */ static const struct usb_device_id usb_audio_ids [] = { #include "quirks-table.h" diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 5b9fd07ce2a2..e8f3f8d622ec 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3629,7 +3629,6 @@ void snd_usb_mixer_disconnect(struct usb_mixer_interface *mixer) mixer->disconnected = true; } -#ifdef CONFIG_PM /* stop any bus activity of a mixer */ static void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer) { @@ -3711,7 +3710,6 @@ int snd_usb_mixer_resume(struct usb_mixer_interface *mixer) return snd_usb_mixer_activate(mixer); } -#endif void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, struct usb_mixer_interface *mixer, @@ -3720,7 +3718,5 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, list->mixer = mixer; list->id = unitid; list->dump = snd_usb_mixer_dump_cval; -#ifdef CONFIG_PM list->resume = restore_mixer_value; -#endif } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 98ea24d91d80..d43895c1ae5c 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -118,10 +118,8 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv); -#ifdef CONFIG_PM int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer); int snd_usb_mixer_resume(struct usb_mixer_interface *mixer); -#endif int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int index, int value); diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index db194ad168d0..1f9863725c7c 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3280,7 +3280,6 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) return err; } -#ifdef CONFIG_PM void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer) { switch (mixer->chip->usb_id) { @@ -3289,7 +3288,6 @@ void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer) break; } } -#endif void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, int unitid) diff --git a/sound/usb/mixer_quirks.h b/sound/usb/mixer_quirks.h index 52be26db558f..4ba01ba3fe8b 100644 --- a/sound/usb/mixer_quirks.h +++ b/sound/usb/mixer_quirks.h @@ -14,9 +14,7 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, struct usb_mixer_elem_info *cval, int unitid, struct snd_kcontrol *kctl); -#ifdef CONFIG_PM void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer); -#endif #endif /* SND_USB_MIXER_QUIRKS_H */ diff --git a/sound/usb/power.h b/sound/usb/power.h index 6004231a7c75..396e3e51440a 100644 --- a/sound/usb/power.h +++ b/sound/usb/power.h @@ -21,17 +21,7 @@ struct snd_usb_power_domain * snd_usb_find_power_domain(struct usb_host_interface *ctrl_iface, unsigned char id); -#ifdef CONFIG_PM int snd_usb_autoresume(struct snd_usb_audio *chip); void snd_usb_autosuspend(struct snd_usb_audio *chip); -#else -static inline int snd_usb_autoresume(struct snd_usb_audio *chip) -{ - return 0; -} -static inline void snd_usb_autosuspend(struct snd_usb_audio *chip) -{ -} -#endif #endif /* __USBAUDIO_POWER_H */ -- cgit v1.2.3-71-gd317 From c7d58971dbea0888b6328ed0ea61089a6d62253a Mon Sep 17 00:00:00 2001 From: Kees Cook Date: Mon, 6 Dec 2021 22:29:41 -0800 Subject: ALSA: mixart: Reduce size of mixart_timer_notify The mixart_timer_notify structure was larger than could be represented by the mixart_msg_data array storage. Adjust the size to as large as possible to fix the warning seen with -Warray-bounds builds: sound/pci/mixart/mixart_core.c: In function 'snd_mixart_threaded_irq': sound/pci/mixart/mixart_core.c:447:50: error: array subscript 'struct mixart_timer_notify[0]' is partly outside array bounds of 'u32[128]' {aka 'unsigned int[128]'} [-Werror=array-bounds] 447 | for(i=0; istream_count; i++) { | ^~ sound/pci/mixart/mixart_core.c:328:12: note: while referencing 'mixart_msg_data' 328 | static u32 mixart_msg_data[MSG_DEFAULT_SIZE / 4]; | ^~~~~~~~~~~~~~~ Signed-off-by: Kees Cook Link: https://lore.kernel.org/r/20211207062941.2413679-1-keescook@chromium.org Signed-off-by: Takashi Iwai --- sound/pci/mixart/mixart_core.c | 3 +-- sound/pci/mixart/mixart_core.h | 10 +++++++++- 2 files changed, 10 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index fb8895af0363..853083dd4bad 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -23,8 +23,6 @@ #define MSG_DESCRIPTOR_SIZE 0x24 #define MSG_HEADER_SIZE (MSG_DESCRIPTOR_SIZE + 4) -#define MSG_DEFAULT_SIZE 512 - #define MSG_TYPE_MASK 0x00000003 /* mask for following types */ #define MSG_TYPE_NOTIFY 0 /* embedded -> driver (only notification, do not get_msg() !) */ #define MSG_TYPE_COMMAND 1 /* driver <-> embedded (a command has no answer) */ @@ -444,6 +442,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) struct mixart_timer_notify *notify; notify = (struct mixart_timer_notify *)mixart_msg_data; + BUILD_BUG_ON(sizeof(notify) > sizeof(mixart_msg_data)); for(i=0; istream_count; i++) { u32 buffer_id = notify->streams[i].buffer_id; diff --git a/sound/pci/mixart/mixart_core.h b/sound/pci/mixart/mixart_core.h index fbf4731a276d..2f0e29ed5d63 100644 --- a/sound/pci/mixart/mixart_core.h +++ b/sound/pci/mixart/mixart_core.h @@ -49,6 +49,7 @@ enum mixart_message_id { MSG_CLOCK_SET_PROPERTIES = 0x200002, }; +#define MSG_DEFAULT_SIZE 512 struct mixart_msg { @@ -251,10 +252,17 @@ struct mixart_sample_pos u32 sample_pos_low_part; } __attribute__((packed)); +/* + * This structure is limited by the size of MSG_DEFAULT_SIZE. Instead of + * having MIXART_MAX_STREAM_PER_CARD * MIXART_MAX_CARDS many streams, + * this is capped to have a total size below MSG_DEFAULT_SIZE. + */ +#define MIXART_MAX_TIMER_NOTIFY_STREAMS \ + ((MSG_DEFAULT_SIZE - sizeof(u32)) / sizeof(struct mixart_sample_pos)) struct mixart_timer_notify { u32 stream_count; - struct mixart_sample_pos streams[MIXART_MAX_STREAM_PER_CARD * MIXART_MAX_CARDS]; + struct mixart_sample_pos streams[MIXART_MAX_TIMER_NOTIFY_STREAMS]; } __attribute__((packed)); -- cgit v1.2.3-71-gd317 From a98478f825862ddc1686a3335f9f1cc278fc5733 Mon Sep 17 00:00:00 2001 From: Anders Roxell Date: Tue, 7 Dec 2021 12:00:53 +0100 Subject: ALSA: ppc: beep: fix clang -Wimplicit-fallthrough Clang warns: sound/ppc/beep.c:103:2: warning: unannotated fall-through between switch labels [-Wimplicit-fallthrough] case SND_TONE: break; ^ sound/ppc/beep.c:103:2: note: insert 'break;' to avoid fall-through case SND_TONE: break; ^ break; 1 warning generated. Clang is more pedantic than GCC, which does not warn when failing through to a case that is just break or return. Clang's version is more in line with the kernel's own stance in deprecated.rst. Add athe missing break to silence the warning. Reported-by: Naresh Kamboju Signed-off-by: Anders Roxell Link: https://lore.kernel.org/r/20211207110053.695712-1-anders.roxell@linaro.org Signed-off-by: Takashi Iwai --- sound/ppc/beep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/ppc/beep.c b/sound/ppc/beep.c index 0f4bce1c0d4f..bf289783eafd 100644 --- a/sound/ppc/beep.c +++ b/sound/ppc/beep.c @@ -99,7 +99,7 @@ static int snd_pmac_beep_event(struct input_dev *dev, unsigned int type, return -1; switch (code) { - case SND_BELL: if (hz) hz = 1000; + case SND_BELL: if (hz) hz = 1000; break; case SND_TONE: break; default: return -1; } -- cgit v1.2.3-71-gd317 From 403c521003a1364fd2d7c01a2a1f66ed025fb94a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 Dec 2021 16:33:23 +0100 Subject: ALSA: mixart: Add sanity check for timer notify streams The miXart timer notification is a variable length, and if a hardware is screwed up, we may access over the actual data size. Let's add a sanity check and bail out if an invalid value is received. Link: https://lore.kernel.org/r/20211207153323.27098-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/mixart/mixart_core.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 853083dd4bad..a047ed0f84e9 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -443,6 +443,8 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) notify = (struct mixart_timer_notify *)mixart_msg_data; BUILD_BUG_ON(sizeof(notify) > sizeof(mixart_msg_data)); + if (snd_BUG_ON(notify->stream_count > ARRAY_SIZE(notify->streams))) + break; for(i=0; istream_count; i++) { u32 buffer_id = notify->streams[i].buffer_id; -- cgit v1.2.3-71-gd317 From 6fadb494a638d8b8a55864ecc6ac58194f03f327 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 Dec 2021 17:51:46 +0100 Subject: ALSA: seq: Set upper limit of processed events Currently ALSA sequencer core tries to process the queued events as much as possible when they become dispatchable. If applications try to queue too massive events to be processed at the very same timing, the sequencer core would still try to process such all events, either in the interrupt context or via some notifier; in either away, it might be a cause of RCU stall or such problems. As a potential workaround for those problems, this patch adds the upper limit of the amount of events to be processed. The remaining events are processed in the next batch, so they won't be lost. For the time being, it's limited up to 1000 events per queue, which should be high enough for any normal usages. Reported-by: Zqiang Reported-by: syzbot+bb950e68b400ab4f65f8@syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20211102033222.3849-1-qiang.zhang1211@gmail.com Link: https://lore.kernel.org/r/20211207165146.2888-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_queue.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index d6c02dea976c..bc933104c3ee 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -235,12 +235,15 @@ struct snd_seq_queue *snd_seq_queue_find_name(char *name) /* -------------------------------------------------------- */ +#define MAX_CELL_PROCESSES_IN_QUEUE 1000 + void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) { unsigned long flags; struct snd_seq_event_cell *cell; snd_seq_tick_time_t cur_tick; snd_seq_real_time_t cur_time; + int processed = 0; if (q == NULL) return; @@ -263,6 +266,8 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + if (++processed >= MAX_CELL_PROCESSES_IN_QUEUE) + goto out; /* the rest processed at the next batch */ } /* Process time queue... */ @@ -272,14 +277,19 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + if (++processed >= MAX_CELL_PROCESSES_IN_QUEUE) + goto out; /* the rest processed at the next batch */ } + out: /* free lock */ spin_lock_irqsave(&q->check_lock, flags); if (q->check_again) { q->check_again = 0; - spin_unlock_irqrestore(&q->check_lock, flags); - goto __again; + if (processed < MAX_CELL_PROCESSES_IN_QUEUE) { + spin_unlock_irqrestore(&q->check_lock, flags); + goto __again; + } } q->check_blocked = 0; spin_unlock_irqrestore(&q->check_lock, flags); -- cgit v1.2.3-71-gd317 From 808709d7675dc0707a9fd6a08077c2b29dca0d60 Mon Sep 17 00:00:00 2001 From: Jason Wang Date: Sun, 12 Dec 2021 15:04:22 +0800 Subject: ALSA: sparc: no need to initialise statics to 0 Static variables do not need to be initialised to 0, because compiler will initialise all uninitialised statics to 0. Thus, remove the unneeded initializations. Signed-off-by: Jason Wang Link: https://lore.kernel.org/r/20211212070422.281924-1-wangborong@cdjrlc.com Signed-off-by: Takashi Iwai --- sound/sparc/dbri.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 6b84f66e4af4..3881e1c1b08a 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -688,7 +688,7 @@ static void dbri_cmdsend(struct snd_dbri *dbri, s32 *cmd, int len) { u32 dvma_addr = (u32)dbri->dma_dvma; s32 tmp, addr; - static int wait_id = 0; + static int wait_id; wait_id++; wait_id &= 0xffff; /* restrict it to a 16 bit counter. */ @@ -1926,7 +1926,7 @@ static void dbri_process_interrupt_buffer(struct snd_dbri *dbri) static irqreturn_t snd_dbri_interrupt(int irq, void *dev_id) { struct snd_dbri *dbri = dev_id; - static int errcnt = 0; + static int errcnt; int x; if (dbri == NULL) @@ -2591,7 +2591,7 @@ static int dbri_probe(struct platform_device *op) struct snd_dbri *dbri; struct resource *rp; struct snd_card *card; - static int dev = 0; + static int dev; int irq; int err; -- cgit v1.2.3-71-gd317 From 78977fd5b11cc90668c0dec6109d2f6572c9601c Mon Sep 17 00:00:00 2001 From: Xiaoke Wang Date: Mon, 13 Dec 2021 18:52:32 +0800 Subject: ALSA: sound/isa/gus: check the return value of kstrdup() kstrdup() returns NULL when some internal memory errors happen, it is better to check the return value of it. Otherwise, we may not to be able to catch some memory errors in time. Signed-off-by: Xiaoke Wang Link: https://lore.kernel.org/r/tencent_1E3950293AC22395ACFE99404C985D738309@qq.com Signed-off-by: Takashi Iwai --- sound/isa/gus/gus_mem.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index ff9480f249fe..4c691dbf2721 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -199,6 +199,10 @@ struct snd_gf1_mem_block *snd_gf1_mem_alloc(struct snd_gf1_mem * alloc, int owne memcpy(&block.share_id, share_id, sizeof(block.share_id)); block.owner = owner; block.name = kstrdup(name, GFP_KERNEL); + if (block.name == NULL) { + snd_gf1_mem_lock(alloc, 1); + return NULL; + } nblock = snd_gf1_mem_xalloc(alloc, &block); snd_gf1_mem_lock(alloc, 1); return nblock; @@ -237,13 +241,13 @@ int snd_gf1_mem_init(struct snd_gus_card * gus) block.ptr = 0; block.size = 1024; block.name = kstrdup("InterWave LFOs", GFP_KERNEL); - if (snd_gf1_mem_xalloc(alloc, &block) == NULL) + if (block.name == NULL || snd_gf1_mem_xalloc(alloc, &block) == NULL) return -ENOMEM; } block.ptr = gus->gf1.default_voice_address; block.size = 4; block.name = kstrdup("Voice default (NULL's)", GFP_KERNEL); - if (snd_gf1_mem_xalloc(alloc, &block) == NULL) + if (block.name == NULL || snd_gf1_mem_xalloc(alloc, &block) == NULL) return -ENOMEM; #ifdef CONFIG_SND_DEBUG snd_card_ro_proc_new(gus->card, "gusmem", gus, snd_gf1_mem_info_read); -- cgit v1.2.3-71-gd317 From c2f51415401cb8e9b7991e828ae12ab2972f2ca7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Dec 2021 14:24:43 +0100 Subject: ALSA: gus: Fix erroneous memory allocation snd_gf1_mem_xalloc() returns NULL incorrectly when the memory chunk is allocated in the middle of the chain. This patch corrects the return value to treat it properly. Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20211213132444.22385-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/isa/gus/gus_mem.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index 4c691dbf2721..5e3ff3137dd7 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -44,7 +44,7 @@ static struct snd_gf1_mem_block *snd_gf1_mem_xalloc(struct snd_gf1_mem * alloc, else nblock->prev->next = nblock; mutex_unlock(&alloc->memory_mutex); - return NULL; + return nblock; } pblock = pblock->next; } -- cgit v1.2.3-71-gd317 From dec242b6a8380c08e41e02fb54f1282894fb45cc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Dec 2021 15:15:12 +0100 Subject: ALSA: gus: Fix memory leaks at memory allocator error paths When snd_gf1_mem_xalloc() returns NULL, the current code still leaves the formerly allocated block.name string but returns an error immediately. This patch does code-refactoring to move the kstrdup() call itself into snd_gf1_mem_xalloc() and deals with the resource free in the helper code by itself for fixing those memory leaks. Suggested-by: Jaroslav Kysela Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20211213132444.22385-2-tiwai@suse.de Link: https://lore.kernel.org/r/20211213141512.27359-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/isa/gus/gus_mem.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index 5e3ff3137dd7..3e56c01c4544 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -24,8 +24,9 @@ void snd_gf1_mem_lock(struct snd_gf1_mem * alloc, int xup) } } -static struct snd_gf1_mem_block *snd_gf1_mem_xalloc(struct snd_gf1_mem * alloc, - struct snd_gf1_mem_block * block) +static struct snd_gf1_mem_block * +snd_gf1_mem_xalloc(struct snd_gf1_mem *alloc, struct snd_gf1_mem_block *block, + const char *name) { struct snd_gf1_mem_block *pblock, *nblock; @@ -33,6 +34,12 @@ static struct snd_gf1_mem_block *snd_gf1_mem_xalloc(struct snd_gf1_mem * alloc, if (nblock == NULL) return NULL; *nblock = *block; + nblock->name = kstrdup(name, GFP_KERNEL); + if (!nblock->name) { + kfree(nblock); + return NULL; + } + pblock = alloc->first; while (pblock) { if (pblock->ptr > nblock->ptr) { @@ -198,12 +205,7 @@ struct snd_gf1_mem_block *snd_gf1_mem_alloc(struct snd_gf1_mem * alloc, int owne if (share_id != NULL) memcpy(&block.share_id, share_id, sizeof(block.share_id)); block.owner = owner; - block.name = kstrdup(name, GFP_KERNEL); - if (block.name == NULL) { - snd_gf1_mem_lock(alloc, 1); - return NULL; - } - nblock = snd_gf1_mem_xalloc(alloc, &block); + nblock = snd_gf1_mem_xalloc(alloc, &block, name); snd_gf1_mem_lock(alloc, 1); return nblock; } @@ -240,14 +242,12 @@ int snd_gf1_mem_init(struct snd_gus_card * gus) if (gus->gf1.enh_mode) { block.ptr = 0; block.size = 1024; - block.name = kstrdup("InterWave LFOs", GFP_KERNEL); - if (block.name == NULL || snd_gf1_mem_xalloc(alloc, &block) == NULL) + if (!snd_gf1_mem_xalloc(alloc, &block, "InterWave LFOs")) return -ENOMEM; } block.ptr = gus->gf1.default_voice_address; block.size = 4; - block.name = kstrdup("Voice default (NULL's)", GFP_KERNEL); - if (block.name == NULL || snd_gf1_mem_xalloc(alloc, &block) == NULL) + if (!snd_gf1_mem_xalloc(alloc, &block, "Voice default (NULL's)")) return -ENOMEM; #ifdef CONFIG_SND_DEBUG snd_card_ro_proc_new(gus->card, "gusmem", gus, snd_gf1_mem_info_read); -- cgit v1.2.3-71-gd317 From 12054f0ce8be7d2003ec068ab27c9eb608397b98 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Dec 2021 17:11:27 -0600 Subject: ALSA/ASoC: hda: move/rename snd_hdac_ext_stop_streams to hdac_stream.c MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit snd_hdac_ext_stop_streams() has really nothing to do with the extension, it just loops over the bus streams. Move it to the hdac_stream layer and rename to remove the 'ext' prefix and add the precision that the chip will also be stopped. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Kai Vehmanen Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20211216231128.344321-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 1 + include/sound/hdaudio_ext.h | 1 - sound/hda/ext/hdac_ext_stream.c | 17 ----------------- sound/hda/hdac_stream.c | 16 ++++++++++++++++ sound/soc/intel/skylake/skl.c | 4 ++-- 5 files changed, 19 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 22af68b01426..6a90ce405e60 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -558,6 +558,7 @@ int snd_hdac_stream_set_params(struct hdac_stream *azx_dev, void snd_hdac_stream_start(struct hdac_stream *azx_dev, bool fresh_start); void snd_hdac_stream_clear(struct hdac_stream *azx_dev); void snd_hdac_stream_stop(struct hdac_stream *azx_dev); +void snd_hdac_stop_streams_and_chip(struct hdac_bus *bus); void snd_hdac_stream_reset(struct hdac_stream *azx_dev); void snd_hdac_stream_sync_trigger(struct hdac_stream *azx_dev, bool set, unsigned int streams, unsigned int reg); diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index d4e31ea16aba..56ea5cde5e63 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -92,7 +92,6 @@ void snd_hdac_ext_stream_decouple_locked(struct hdac_bus *bus, struct hdac_ext_stream *azx_dev, bool decouple); void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, struct hdac_ext_stream *azx_dev, bool decouple); -void snd_hdac_ext_stop_streams(struct hdac_bus *bus); int snd_hdac_ext_stream_set_spib(struct hdac_bus *bus, struct hdac_ext_stream *stream, u32 value); diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index 37154ed43bd5..c09652da43ff 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -475,23 +475,6 @@ int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_bus *bus, } EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_get_spbmaxfifo); - -/** - * snd_hdac_ext_stop_streams - stop all stream if running - * @bus: HD-audio core bus - */ -void snd_hdac_ext_stop_streams(struct hdac_bus *bus) -{ - struct hdac_stream *stream; - - if (bus->chip_init) { - list_for_each_entry(stream, &bus->stream_list, list) - snd_hdac_stream_stop(stream); - snd_hdac_bus_stop_chip(bus); - } -} -EXPORT_SYMBOL_GPL(snd_hdac_ext_stop_streams); - /** * snd_hdac_ext_stream_drsm_enable - enable DMA resume for a stream * @bus: HD-audio core bus diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index aa7955fdf68a..f3582012d22f 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -142,6 +142,22 @@ void snd_hdac_stream_stop(struct hdac_stream *azx_dev) } EXPORT_SYMBOL_GPL(snd_hdac_stream_stop); +/** + * snd_hdac_stop_streams_and_chip - stop all streams and chip if running + * @bus: HD-audio core bus + */ +void snd_hdac_stop_streams_and_chip(struct hdac_bus *bus) +{ + struct hdac_stream *stream; + + if (bus->chip_init) { + list_for_each_entry(stream, &bus->stream_list, list) + snd_hdac_stream_stop(stream); + snd_hdac_bus_stop_chip(bus); + } +} +EXPORT_SYMBOL_GPL(snd_hdac_stop_streams_and_chip); + /** * snd_hdac_stream_reset - reset a stream * @azx_dev: HD-audio core stream to reset diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 5b1a15e39912..148ddf4cace0 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -439,7 +439,7 @@ static int skl_free(struct hdac_bus *bus) skl->init_done = 0; /* to be sure */ - snd_hdac_ext_stop_streams(bus); + snd_hdac_stop_streams_and_chip(bus); if (bus->irq >= 0) free_irq(bus->irq, (void *)bus); @@ -1096,7 +1096,7 @@ static void skl_shutdown(struct pci_dev *pci) if (!skl->init_done) return; - snd_hdac_ext_stop_streams(bus); + snd_hdac_stop_streams_and_chip(bus); list_for_each_entry(s, &bus->stream_list, list) { stream = stream_to_hdac_ext_stream(s); snd_hdac_ext_stream_decouple(bus, stream, false); -- cgit v1.2.3-71-gd317 From 0f7e5ee62f4c24ca9db58351c86653cc3ee0bd0e Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Dec 2021 17:11:28 -0600 Subject: ALSA: HDA: hdac_ext_stream: use consistent prefixes for variables The existing code maximizes confusion by using 'stream' and 'hstream' variables of different types. Examples: struct hdac_stream *stream; struct hdac_ext_stream *stream; struct hdac_stream *hstream; struct hdac_ext_stream *hstream; with some additional copy/paste remains: struct hdac_ext_stream *azx_dev; This patch suggests a consistent naming across all 'hdac_ext_stream' functions. The convention is: struct hdac_stream *hstream; struct hdac_ext_stream *hext_stream; No functionality change - just renaming of variables and more consistent indentation. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Kai Vehmanen Reviewed-by: Rander Wang Link: https://lore.kernel.org/r/20211216231128.344321-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- include/sound/hdaudio_ext.h | 26 +++--- sound/hda/ext/hdac_ext_stream.c | 199 ++++++++++++++++++++-------------------- 2 files changed, 113 insertions(+), 112 deletions(-) (limited to 'sound') diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 56ea5cde5e63..77123c3e4095 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -78,35 +78,35 @@ struct hdac_ext_stream { container_of(s, struct hdac_ext_stream, hstream) void snd_hdac_ext_stream_init(struct hdac_bus *bus, - struct hdac_ext_stream *stream, int idx, - int direction, int tag); + struct hdac_ext_stream *hext_stream, int idx, + int direction, int tag); int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx, - int num_stream, int dir); + int num_stream, int dir); void snd_hdac_stream_free_all(struct hdac_bus *bus); void snd_hdac_link_free_all(struct hdac_bus *bus); struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream, int type); -void snd_hdac_ext_stream_release(struct hdac_ext_stream *azx_dev, int type); +void snd_hdac_ext_stream_release(struct hdac_ext_stream *hext_stream, int type); void snd_hdac_ext_stream_decouple_locked(struct hdac_bus *bus, - struct hdac_ext_stream *azx_dev, bool decouple); + struct hdac_ext_stream *hext_stream, bool decouple); void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, struct hdac_ext_stream *azx_dev, bool decouple); int snd_hdac_ext_stream_set_spib(struct hdac_bus *bus, - struct hdac_ext_stream *stream, u32 value); + struct hdac_ext_stream *hext_stream, u32 value); int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_bus *bus, - struct hdac_ext_stream *stream); + struct hdac_ext_stream *hext_stream); void snd_hdac_ext_stream_drsm_enable(struct hdac_bus *bus, bool enable, int index); int snd_hdac_ext_stream_set_dpibr(struct hdac_bus *bus, - struct hdac_ext_stream *stream, u32 value); -int snd_hdac_ext_stream_set_lpib(struct hdac_ext_stream *stream, u32 value); + struct hdac_ext_stream *hext_stream, u32 value); +int snd_hdac_ext_stream_set_lpib(struct hdac_ext_stream *hext_stream, u32 value); -void snd_hdac_ext_link_stream_start(struct hdac_ext_stream *hstream); -void snd_hdac_ext_link_stream_clear(struct hdac_ext_stream *hstream); -void snd_hdac_ext_link_stream_reset(struct hdac_ext_stream *hstream); -int snd_hdac_ext_link_stream_setup(struct hdac_ext_stream *stream, int fmt); +void snd_hdac_ext_link_stream_start(struct hdac_ext_stream *hext_stream); +void snd_hdac_ext_link_stream_clear(struct hdac_ext_stream *hext_stream); +void snd_hdac_ext_link_stream_reset(struct hdac_ext_stream *hext_stream); +int snd_hdac_ext_link_stream_setup(struct hdac_ext_stream *hext_stream, int fmt); struct hdac_ext_link { struct hdac_bus *bus; diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index c09652da43ff..d2b5724b463f 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -18,7 +18,7 @@ /** * snd_hdac_ext_stream_init - initialize each stream (aka device) * @bus: HD-audio core bus - * @stream: HD-audio ext core stream object to initialize + * @hext_stream: HD-audio ext core stream object to initialize * @idx: stream index number * @direction: stream direction (SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE) * @tag: the tag id to assign @@ -27,34 +27,34 @@ * invoke hdac stream initialization routine */ void snd_hdac_ext_stream_init(struct hdac_bus *bus, - struct hdac_ext_stream *stream, - int idx, int direction, int tag) + struct hdac_ext_stream *hext_stream, + int idx, int direction, int tag) { if (bus->ppcap) { - stream->pphc_addr = bus->ppcap + AZX_PPHC_BASE + + hext_stream->pphc_addr = bus->ppcap + AZX_PPHC_BASE + AZX_PPHC_INTERVAL * idx; - stream->pplc_addr = bus->ppcap + AZX_PPLC_BASE + + hext_stream->pplc_addr = bus->ppcap + AZX_PPLC_BASE + AZX_PPLC_MULTI * bus->num_streams + AZX_PPLC_INTERVAL * idx; } if (bus->spbcap) { - stream->spib_addr = bus->spbcap + AZX_SPB_BASE + + hext_stream->spib_addr = bus->spbcap + AZX_SPB_BASE + AZX_SPB_INTERVAL * idx + AZX_SPB_SPIB; - stream->fifo_addr = bus->spbcap + AZX_SPB_BASE + + hext_stream->fifo_addr = bus->spbcap + AZX_SPB_BASE + AZX_SPB_INTERVAL * idx + AZX_SPB_MAXFIFO; } if (bus->drsmcap) - stream->dpibr_addr = bus->drsmcap + AZX_DRSM_BASE + + hext_stream->dpibr_addr = bus->drsmcap + AZX_DRSM_BASE + AZX_DRSM_INTERVAL * idx; - stream->decoupled = false; - snd_hdac_stream_init(bus, &stream->hstream, idx, direction, tag); + hext_stream->decoupled = false; + snd_hdac_stream_init(bus, &hext_stream->hstream, idx, direction, tag); } EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init); @@ -67,18 +67,18 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init); * @dir: direction of streams */ int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx, - int num_stream, int dir) + int num_stream, int dir) { int stream_tag = 0; int i, tag, idx = start_idx; for (i = 0; i < num_stream; i++) { - struct hdac_ext_stream *stream = - kzalloc(sizeof(*stream), GFP_KERNEL); - if (!stream) + struct hdac_ext_stream *hext_stream = + kzalloc(sizeof(*hext_stream), GFP_KERNEL); + if (!hext_stream) return -ENOMEM; tag = ++stream_tag; - snd_hdac_ext_stream_init(bus, stream, idx, dir, tag); + snd_hdac_ext_stream_init(bus, hext_stream, idx, dir, tag); idx++; } @@ -95,22 +95,22 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init_all); void snd_hdac_stream_free_all(struct hdac_bus *bus) { struct hdac_stream *s, *_s; - struct hdac_ext_stream *stream; + struct hdac_ext_stream *hext_stream; list_for_each_entry_safe(s, _s, &bus->stream_list, list) { - stream = stream_to_hdac_ext_stream(s); - snd_hdac_ext_stream_decouple(bus, stream, false); + hext_stream = stream_to_hdac_ext_stream(s); + snd_hdac_ext_stream_decouple(bus, hext_stream, false); list_del(&s->list); - kfree(stream); + kfree(hext_stream); } } EXPORT_SYMBOL_GPL(snd_hdac_stream_free_all); void snd_hdac_ext_stream_decouple_locked(struct hdac_bus *bus, - struct hdac_ext_stream *stream, + struct hdac_ext_stream *hext_stream, bool decouple) { - struct hdac_stream *hstream = &stream->hstream; + struct hdac_stream *hstream = &hext_stream->hstream; u32 val; int mask = AZX_PPCTL_PROCEN(hstream->index); @@ -121,76 +121,76 @@ void snd_hdac_ext_stream_decouple_locked(struct hdac_bus *bus, else if (!decouple && val) snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, mask, 0); - stream->decoupled = decouple; + hext_stream->decoupled = decouple; } EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_decouple_locked); /** * snd_hdac_ext_stream_decouple - decouple the hdac stream * @bus: HD-audio core bus - * @stream: HD-audio ext core stream object to initialize + * @hext_stream: HD-audio ext core stream object to initialize * @decouple: flag to decouple */ void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, - struct hdac_ext_stream *stream, bool decouple) + struct hdac_ext_stream *hext_stream, bool decouple) { spin_lock_irq(&bus->reg_lock); - snd_hdac_ext_stream_decouple_locked(bus, stream, decouple); + snd_hdac_ext_stream_decouple_locked(bus, hext_stream, decouple); spin_unlock_irq(&bus->reg_lock); } EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_decouple); /** * snd_hdac_ext_link_stream_start - start a stream - * @stream: HD-audio ext core stream to start + * @hext_stream: HD-audio ext core stream to start */ -void snd_hdac_ext_link_stream_start(struct hdac_ext_stream *stream) +void snd_hdac_ext_link_stream_start(struct hdac_ext_stream *hext_stream) { - snd_hdac_updatel(stream->pplc_addr, AZX_REG_PPLCCTL, + snd_hdac_updatel(hext_stream->pplc_addr, AZX_REG_PPLCCTL, AZX_PPLCCTL_RUN, AZX_PPLCCTL_RUN); } EXPORT_SYMBOL_GPL(snd_hdac_ext_link_stream_start); /** * snd_hdac_ext_link_stream_clear - stop a stream DMA - * @stream: HD-audio ext core stream to stop + * @hext_stream: HD-audio ext core stream to stop */ -void snd_hdac_ext_link_stream_clear(struct hdac_ext_stream *stream) +void snd_hdac_ext_link_stream_clear(struct hdac_ext_stream *hext_stream) { - snd_hdac_updatel(stream->pplc_addr, AZX_REG_PPLCCTL, AZX_PPLCCTL_RUN, 0); + snd_hdac_updatel(hext_stream->pplc_addr, AZX_REG_PPLCCTL, AZX_PPLCCTL_RUN, 0); } EXPORT_SYMBOL_GPL(snd_hdac_ext_link_stream_clear); /** * snd_hdac_ext_link_stream_reset - reset a stream - * @stream: HD-audio ext core stream to reset + * @hext_stream: HD-audio ext core stream to reset */ -void snd_hdac_ext_link_stream_reset(struct hdac_ext_stream *stream) +void snd_hdac_ext_link_stream_reset(struct hdac_ext_stream *hext_stream) { unsigned char val; int timeout; - snd_hdac_ext_link_stream_clear(stream); + snd_hdac_ext_link_stream_clear(hext_stream); - snd_hdac_updatel(stream->pplc_addr, AZX_REG_PPLCCTL, + snd_hdac_updatel(hext_stream->pplc_addr, AZX_REG_PPLCCTL, AZX_PPLCCTL_STRST, AZX_PPLCCTL_STRST); udelay(3); timeout = 50; do { - val = readl(stream->pplc_addr + AZX_REG_PPLCCTL) & + val = readl(hext_stream->pplc_addr + AZX_REG_PPLCCTL) & AZX_PPLCCTL_STRST; if (val) break; udelay(3); } while (--timeout); val &= ~AZX_PPLCCTL_STRST; - writel(val, stream->pplc_addr + AZX_REG_PPLCCTL); + writel(val, hext_stream->pplc_addr + AZX_REG_PPLCCTL); udelay(3); timeout = 50; /* waiting for hardware to report that the stream is out of reset */ do { - val = readl(stream->pplc_addr + AZX_REG_PPLCCTL) & AZX_PPLCCTL_STRST; + val = readl(hext_stream->pplc_addr + AZX_REG_PPLCCTL) & AZX_PPLCCTL_STRST; if (!val) break; udelay(3); @@ -201,24 +201,24 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_link_stream_reset); /** * snd_hdac_ext_link_stream_setup - set up the SD for streaming - * @stream: HD-audio ext core stream to set up + * @hext_stream: HD-audio ext core stream to set up * @fmt: stream format */ -int snd_hdac_ext_link_stream_setup(struct hdac_ext_stream *stream, int fmt) +int snd_hdac_ext_link_stream_setup(struct hdac_ext_stream *hext_stream, int fmt) { - struct hdac_stream *hstream = &stream->hstream; + struct hdac_stream *hstream = &hext_stream->hstream; unsigned int val; /* make sure the run bit is zero for SD */ - snd_hdac_ext_link_stream_clear(stream); + snd_hdac_ext_link_stream_clear(hext_stream); /* program the stream_tag */ - val = readl(stream->pplc_addr + AZX_REG_PPLCCTL); + val = readl(hext_stream->pplc_addr + AZX_REG_PPLCCTL); val = (val & ~AZX_PPLCCTL_STRM_MASK) | (hstream->stream_tag << AZX_PPLCCTL_STRM_SHIFT); - writel(val, stream->pplc_addr + AZX_REG_PPLCCTL); + writel(val, hext_stream->pplc_addr + AZX_REG_PPLCCTL); /* program the stream format */ - writew(fmt, stream->pplc_addr + AZX_REG_PPLCFMT); + writew(fmt, hext_stream->pplc_addr + AZX_REG_PPLCFMT); return 0; } @@ -230,7 +230,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_link_stream_setup); * @stream: stream id */ void snd_hdac_ext_link_set_stream_id(struct hdac_ext_link *link, - int stream) + int stream) { snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, (1 << stream), 1 << stream); } @@ -250,10 +250,10 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_link_clear_stream_id); static struct hdac_ext_stream * hdac_ext_link_stream_assign(struct hdac_bus *bus, - struct snd_pcm_substream *substream) + struct snd_pcm_substream *substream) { struct hdac_ext_stream *res = NULL; - struct hdac_stream *stream = NULL; + struct hdac_stream *hstream = NULL; if (!bus->ppcap) { dev_err(bus->dev, "stream type not supported\n"); @@ -261,22 +261,22 @@ hdac_ext_link_stream_assign(struct hdac_bus *bus, } spin_lock_irq(&bus->reg_lock); - list_for_each_entry(stream, &bus->stream_list, list) { - struct hdac_ext_stream *hstream = container_of(stream, - struct hdac_ext_stream, - hstream); - if (stream->direction != substream->stream) + list_for_each_entry(hstream, &bus->stream_list, list) { + struct hdac_ext_stream *hext_stream = container_of(hstream, + struct hdac_ext_stream, + hstream); + if (hstream->direction != substream->stream) continue; /* check if decoupled stream and not in use is available */ - if (hstream->decoupled && !hstream->link_locked) { - res = hstream; + if (hext_stream->decoupled && !hext_stream->link_locked) { + res = hext_stream; break; } - if (!hstream->link_locked) { - snd_hdac_ext_stream_decouple_locked(bus, hstream, true); - res = hstream; + if (!hext_stream->link_locked) { + snd_hdac_ext_stream_decouple_locked(bus, hext_stream, true); + res = hext_stream; break; } } @@ -290,10 +290,10 @@ hdac_ext_link_stream_assign(struct hdac_bus *bus, static struct hdac_ext_stream * hdac_ext_host_stream_assign(struct hdac_bus *bus, - struct snd_pcm_substream *substream) + struct snd_pcm_substream *substream) { struct hdac_ext_stream *res = NULL; - struct hdac_stream *stream = NULL; + struct hdac_stream *hstream = NULL; if (!bus->ppcap) { dev_err(bus->dev, "stream type not supported\n"); @@ -301,17 +301,17 @@ hdac_ext_host_stream_assign(struct hdac_bus *bus, } spin_lock_irq(&bus->reg_lock); - list_for_each_entry(stream, &bus->stream_list, list) { - struct hdac_ext_stream *hstream = container_of(stream, - struct hdac_ext_stream, - hstream); - if (stream->direction != substream->stream) + list_for_each_entry(hstream, &bus->stream_list, list) { + struct hdac_ext_stream *hext_stream = container_of(hstream, + struct hdac_ext_stream, + hstream); + if (hstream->direction != substream->stream) continue; - if (!stream->opened) { - if (!hstream->decoupled) - snd_hdac_ext_stream_decouple_locked(bus, hstream, true); - res = hstream; + if (!hstream->opened) { + if (!hext_stream->decoupled) + snd_hdac_ext_stream_decouple_locked(bus, hext_stream, true); + res = hext_stream; break; } } @@ -346,16 +346,17 @@ struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream, int type) { - struct hdac_ext_stream *hstream = NULL; - struct hdac_stream *stream = NULL; + struct hdac_ext_stream *hext_stream = NULL; + struct hdac_stream *hstream = NULL; switch (type) { case HDAC_EXT_STREAM_TYPE_COUPLED: - stream = snd_hdac_stream_assign(bus, substream); - if (stream) - hstream = container_of(stream, - struct hdac_ext_stream, hstream); - return hstream; + hstream = snd_hdac_stream_assign(bus, substream); + if (hstream) + hext_stream = container_of(hstream, + struct hdac_ext_stream, + hstream); + return hext_stream; case HDAC_EXT_STREAM_TYPE_HOST: return hdac_ext_host_stream_assign(bus, substream); @@ -371,34 +372,34 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_assign); /** * snd_hdac_ext_stream_release - release the assigned stream - * @stream: HD-audio ext core stream to release + * @hext_stream: HD-audio ext core stream to release * @type: type of stream (coupled, host or link stream) * * Release the stream that has been assigned by snd_hdac_ext_stream_assign(). */ -void snd_hdac_ext_stream_release(struct hdac_ext_stream *stream, int type) +void snd_hdac_ext_stream_release(struct hdac_ext_stream *hext_stream, int type) { - struct hdac_bus *bus = stream->hstream.bus; + struct hdac_bus *bus = hext_stream->hstream.bus; switch (type) { case HDAC_EXT_STREAM_TYPE_COUPLED: - snd_hdac_stream_release(&stream->hstream); + snd_hdac_stream_release(&hext_stream->hstream); break; case HDAC_EXT_STREAM_TYPE_HOST: spin_lock_irq(&bus->reg_lock); - if (stream->decoupled && !stream->link_locked) - snd_hdac_ext_stream_decouple_locked(bus, stream, false); + if (hext_stream->decoupled && !hext_stream->link_locked) + snd_hdac_ext_stream_decouple_locked(bus, hext_stream, false); spin_unlock_irq(&bus->reg_lock); - snd_hdac_stream_release(&stream->hstream); + snd_hdac_stream_release(&hext_stream->hstream); break; case HDAC_EXT_STREAM_TYPE_LINK: spin_lock_irq(&bus->reg_lock); - if (stream->decoupled && !stream->hstream.opened) - snd_hdac_ext_stream_decouple_locked(bus, stream, false); - stream->link_locked = 0; - stream->link_substream = NULL; + if (hext_stream->decoupled && !hext_stream->hstream.opened) + snd_hdac_ext_stream_decouple_locked(bus, hext_stream, false); + hext_stream->link_locked = 0; + hext_stream->link_substream = NULL; spin_unlock_irq(&bus->reg_lock); break; @@ -437,11 +438,11 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_spbcap_enable); /** * snd_hdac_ext_stream_set_spib - sets the spib value of a stream * @bus: HD-audio core bus - * @stream: hdac_ext_stream + * @hext_stream: hdac_ext_stream * @value: spib value to set */ int snd_hdac_ext_stream_set_spib(struct hdac_bus *bus, - struct hdac_ext_stream *stream, u32 value) + struct hdac_ext_stream *hext_stream, u32 value) { if (!bus->spbcap) { @@ -449,7 +450,7 @@ int snd_hdac_ext_stream_set_spib(struct hdac_bus *bus, return -EINVAL; } - writel(value, stream->spib_addr); + writel(value, hext_stream->spib_addr); return 0; } @@ -458,12 +459,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_set_spib); /** * snd_hdac_ext_stream_get_spbmaxfifo - gets the spib value of a stream * @bus: HD-audio core bus - * @stream: hdac_ext_stream + * @hext_stream: hdac_ext_stream * * Return maxfifo for the stream */ int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_bus *bus, - struct hdac_ext_stream *stream) + struct hdac_ext_stream *hext_stream) { if (!bus->spbcap) { @@ -471,7 +472,7 @@ int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_bus *bus, return -EINVAL; } - return readl(stream->fifo_addr); + return readl(hext_stream->fifo_addr); } EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_get_spbmaxfifo); @@ -503,11 +504,11 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_drsm_enable); /** * snd_hdac_ext_stream_set_dpibr - sets the dpibr value of a stream * @bus: HD-audio core bus - * @stream: hdac_ext_stream + * @hext_stream: hdac_ext_stream * @value: dpib value to set */ int snd_hdac_ext_stream_set_dpibr(struct hdac_bus *bus, - struct hdac_ext_stream *stream, u32 value) + struct hdac_ext_stream *hext_stream, u32 value) { if (!bus->drsmcap) { @@ -515,7 +516,7 @@ int snd_hdac_ext_stream_set_dpibr(struct hdac_bus *bus, return -EINVAL; } - writel(value, stream->dpibr_addr); + writel(value, hext_stream->dpibr_addr); return 0; } @@ -523,12 +524,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_set_dpibr); /** * snd_hdac_ext_stream_set_lpib - sets the lpib value of a stream - * @stream: hdac_ext_stream + * @hext_stream: hdac_ext_stream * @value: lpib value to set */ -int snd_hdac_ext_stream_set_lpib(struct hdac_ext_stream *stream, u32 value) +int snd_hdac_ext_stream_set_lpib(struct hdac_ext_stream *hext_stream, u32 value) { - snd_hdac_stream_writel(&stream->hstream, SD_LPIB, value); + snd_hdac_stream_writel(&hext_stream->hstream, SD_LPIB, value); return 0; } -- cgit v1.2.3-71-gd317 From 5dcdc4600c3a7773a7b901d6b7eb29340be95cf6 Mon Sep 17 00:00:00 2001 From: Yang Guang Date: Sat, 18 Dec 2021 09:54:16 +0800 Subject: ALSA: hda: use swap() to make code cleaner Use the macro 'swap()' defined in 'include/linux/minmax.h' to avoid opencoding it. Reported-by: Zeal Robot Signed-off-by: David Yang Signed-off-by: Yang Guang Link: https://lore.kernel.org/r/ebc9db44b802dfc88e1538629b517e000acb27b3.1639790796.git.yang.guang5@zte.com.cn Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_auto_parser.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 4a854475a0e6..82c492b05667 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -92,14 +92,10 @@ static int compare_input_type(const void *ap, const void *bp) */ static void reorder_outputs(unsigned int nums, hda_nid_t *pins) { - hda_nid_t nid; - switch (nums) { case 3: case 4: - nid = pins[1]; - pins[1] = pins[2]; - pins[2] = nid; + swap(pins[1], pins[2]); break; } } -- cgit v1.2.3-71-gd317 From 6c3a0c39130c9f29d52269cca7cf29c0e1c8d966 Mon Sep 17 00:00:00 2001 From: Ville Syrjälä Date: Wed, 22 Dec 2021 16:53:50 +0200 Subject: ALSA: hda/hdmi: Disable silent stream on GLK MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The silent stream stuff recurses back into i915 audio component .get_power() from the .pin_eld_notify() hook. On GLK this will deadlock as i915 may already be holding the relevant modeset locks during .pin_eld_notify() and the GLK audio vs. CDCLK workaround will try to grab the same locks from .get_power(). Until someone comes up with a better fix just disable the silent stream support on GLK. Cc: stable@vger.kernel.org Cc: Harsha Priya Cc: Emmanuel Jillela Cc: Kai Vehmanen Cc: Takashi Iwai Closes: https://gitlab.freedesktop.org/drm/intel/-/issues/2623 Fixes: 951894cf30f4 ("ALSA: hda/hdmi: Add Intel silent stream support") Signed-off-by: Ville Syrjälä Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20211222145350.24342-1-ville.syrjala@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 33e5f1aa24f9..4ac2a28a3167 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2947,7 +2947,8 @@ static int parse_intel_hdmi(struct hda_codec *codec) /* Intel Haswell and onwards; audio component with eld notifier */ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, - const int *port_map, int port_num, int dev_num) + const int *port_map, int port_num, int dev_num, + bool send_silent_stream) { struct hdmi_spec *spec; int err; @@ -2980,7 +2981,7 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, * Enable silent stream feature, if it is enabled via * module param or Kconfig option */ - if (enable_silent_stream) + if (send_silent_stream) spec->send_silent_stream = true; return parse_intel_hdmi(codec); @@ -2988,12 +2989,18 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, static int patch_i915_hsw_hdmi(struct hda_codec *codec) { - return intel_hsw_common_init(codec, 0x08, NULL, 0, 3); + return intel_hsw_common_init(codec, 0x08, NULL, 0, 3, + enable_silent_stream); } static int patch_i915_glk_hdmi(struct hda_codec *codec) { - return intel_hsw_common_init(codec, 0x0b, NULL, 0, 3); + /* + * Silent stream calls audio component .get_power() from + * .pin_eld_notify(). On GLK this will deadlock in i915 due + * to the audio vs. CDCLK workaround. + */ + return intel_hsw_common_init(codec, 0x0b, NULL, 0, 3, false); } static int patch_i915_icl_hdmi(struct hda_codec *codec) @@ -3004,7 +3011,8 @@ static int patch_i915_icl_hdmi(struct hda_codec *codec) */ static const int map[] = {0x0, 0x4, 0x6, 0x8, 0xa, 0xb}; - return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 3); + return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 3, + enable_silent_stream); } static int patch_i915_tgl_hdmi(struct hda_codec *codec) @@ -3016,7 +3024,8 @@ static int patch_i915_tgl_hdmi(struct hda_codec *codec) static const int map[] = {0x4, 0x6, 0x8, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf}; int ret; - ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 4); + ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 4, + enable_silent_stream); if (!ret) { struct hdmi_spec *spec = codec->spec; -- cgit v1.2.3-71-gd317 From 4d5a628d96532607b2e01e507f951ab19a33fc12 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Thu, 23 Dec 2021 09:34:23 +0200 Subject: ALSA: hda: Add AlderLake-N PCI ID Add HD Audio PCI ID for Intel AlderLake-N. Add rules to snd_intel_dsp_find_config() to choose DSP-based SOF driver for ADL-N systems with PCH-DMIC or Soundwire codecs, and plain HDA driver for the rest (DSP not used). Signed-off-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20211223073424.1738125-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/intel-dsp-config.c | 4 ++++ sound/pci/hda/hda_intel.c | 3 +++ 2 files changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 26f8665da689..b5f9b8d00e0b 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -355,6 +355,10 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x51cc, }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x54c8, + }, #endif }; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 221afacbc7fd..4987353ee770 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2486,6 +2486,9 @@ static const struct pci_device_id azx_ids[] = { /* Alderlake-M */ { PCI_DEVICE(0x8086, 0x51cc), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Alderlake-N */ + { PCI_DEVICE(0x8086, 0x54c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, -- cgit v1.2.3-71-gd317 From ca1ece24d9bc5bd1d5257494654bb2b73942ddea Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Thu, 23 Dec 2021 09:34:24 +0200 Subject: ALSA: hda: Add new AlderLake-P variant PCI ID Add HD Audio PCI ID for a variant of Intel AlderLake-P. Use same driver match rules as for existing AlderLake-P devices. Signed-off-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20211223073424.1738125-2-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/intel-dsp-config.c | 4 ++++ sound/pci/hda/hda_intel.c | 2 ++ 2 files changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index b5f9b8d00e0b..8a92d661410c 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -355,6 +355,10 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x51cc, }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x51cd, + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x54c8, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4987353ee770..de0c2dfb8b03 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2483,6 +2483,8 @@ static const struct pci_device_id azx_ids[] = { /* Alderlake-P */ { PCI_DEVICE(0x8086, 0x51c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x51cd), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Alderlake-M */ { PCI_DEVICE(0x8086, 0x51cc), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, -- cgit v1.2.3-71-gd317 From d278dc9151a034674b31ffeda24cdfb0073570f3 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 23 Dec 2021 17:23:49 +0530 Subject: ALSA: hda/tegra: Fix Tegra194 HDA reset failure HDA regression is recently reported on Tegra194 based platforms. This happens because "hda2codec_2x" reset does not really exist in Tegra194 and it causes probe failure. All the HDA based audio tests fail at the moment. This underlying issue is exposed by commit c045ceb5a145 ("reset: tegra-bpmp: Handle errors in BPMP response") which now checks return code of BPMP command response. Fix this issue by skipping unavailable reset on Tegra194. Cc: stable@vger.kernel.org Signed-off-by: Sameer Pujar Reviewed-by: Dmitry Osipenko Link: https://lore.kernel.org/r/1640260431-11613-2-git-send-email-spujar@nvidia.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_tegra.c | 43 ++++++++++++++++++++++++++++++++++--------- 1 file changed, 34 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index ea700395bef4..773f4903550a 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -68,14 +68,20 @@ */ #define TEGRA194_NUM_SDO_LINES 4 +struct hda_tegra_soc { + bool has_hda2codec_2x_reset; +}; + struct hda_tegra { struct azx chip; struct device *dev; - struct reset_control *reset; + struct reset_control_bulk_data resets[3]; struct clk_bulk_data clocks[3]; + unsigned int nresets; unsigned int nclocks; void __iomem *regs; struct work_struct probe_work; + const struct hda_tegra_soc *soc; }; #ifdef CONFIG_PM @@ -170,7 +176,7 @@ static int __maybe_unused hda_tegra_runtime_resume(struct device *dev) int rc; if (!chip->running) { - rc = reset_control_assert(hda->reset); + rc = reset_control_bulk_assert(hda->nresets, hda->resets); if (rc) return rc; } @@ -187,7 +193,7 @@ static int __maybe_unused hda_tegra_runtime_resume(struct device *dev) } else { usleep_range(10, 100); - rc = reset_control_deassert(hda->reset); + rc = reset_control_bulk_deassert(hda->nresets, hda->resets); if (rc) return rc; } @@ -427,9 +433,17 @@ static int hda_tegra_create(struct snd_card *card, return 0; } +static const struct hda_tegra_soc tegra30_data = { + .has_hda2codec_2x_reset = true, +}; + +static const struct hda_tegra_soc tegra194_data = { + .has_hda2codec_2x_reset = false, +}; + static const struct of_device_id hda_tegra_match[] = { - { .compatible = "nvidia,tegra30-hda" }, - { .compatible = "nvidia,tegra194-hda" }, + { .compatible = "nvidia,tegra30-hda", .data = &tegra30_data }, + { .compatible = "nvidia,tegra194-hda", .data = &tegra194_data }, {}, }; MODULE_DEVICE_TABLE(of, hda_tegra_match); @@ -449,6 +463,8 @@ static int hda_tegra_probe(struct platform_device *pdev) hda->dev = &pdev->dev; chip = &hda->chip; + hda->soc = of_device_get_match_data(&pdev->dev); + err = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, 0, &card); if (err < 0) { @@ -456,11 +472,20 @@ static int hda_tegra_probe(struct platform_device *pdev) return err; } - hda->reset = devm_reset_control_array_get_exclusive(&pdev->dev); - if (IS_ERR(hda->reset)) { - err = PTR_ERR(hda->reset); + hda->resets[hda->nresets++].id = "hda"; + hda->resets[hda->nresets++].id = "hda2hdmi"; + /* + * "hda2codec_2x" reset is not present on Tegra194. Though DT would + * be updated to reflect this, but to have backward compatibility + * below is necessary. + */ + if (hda->soc->has_hda2codec_2x_reset) + hda->resets[hda->nresets++].id = "hda2codec_2x"; + + err = devm_reset_control_bulk_get_exclusive(&pdev->dev, hda->nresets, + hda->resets); + if (err) goto out_free; - } hda->clocks[hda->nclocks++].id = "hda"; hda->clocks[hda->nclocks++].id = "hda2hdmi"; -- cgit v1.2.3-71-gd317 From c1933008679586b20437280463110c967d66f865 Mon Sep 17 00:00:00 2001 From: Christian Lachner Date: Mon, 3 Jan 2022 15:05:17 +0100 Subject: ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Master after reboot from Windows This patch addresses an issue where after rebooting from Windows into Linux there would be no audio output. It turns out that the Realtek Audio driver on Windows changes some coeffs which are not being reset/reinitialized when rebooting the machine. As a result, there is no audio output until these coeffs are being reset to their initial state. This patch takes care of that by setting known-good (initial) values to the coeffs. We initially relied upon alc1220_fixup_clevo_p950() to fix some pins in the connection list. However, it also sets coef 0x7 which does not need to be touched. Furthermore, to prevent mixing device-specific quirks I introduced a new alc1220_fixup_gb_x570() which is heavily based on alc1220_fixup_clevo_p950() but does not set coeff 0x7 and fixes the coeffs that are actually needed instead. This new alc1220_fixup_gb_x570() is believed to also work for other boards, like the Gigabyte X570 Aorus Extreme and the newer Gigabyte Aorus X570S Master. However, as there is no way for me to test these I initially only enable this new behaviour for the mainboard I have which is the Gigabyte X570(non-S) Aorus Master. I tested this patch on the 5.15 branch as well as on master and it is working well for me. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205275 Signed-off-by: Christian Lachner Fixes: 0d45e86d2267d ("ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Master") Cc: Link: https://lore.kernel.org/r/20220103140517.30273-2-gladiac@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 30 +++++++++++++++++++++++++++++- 1 file changed, 29 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2f1727faec69..2eea70605fd3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1924,6 +1924,7 @@ enum { ALC887_FIXUP_ASUS_BASS, ALC887_FIXUP_BASS_CHMAP, ALC1220_FIXUP_GB_DUAL_CODECS, + ALC1220_FIXUP_GB_X570, ALC1220_FIXUP_CLEVO_P950, ALC1220_FIXUP_CLEVO_PB51ED, ALC1220_FIXUP_CLEVO_PB51ED_PINS, @@ -2113,6 +2114,29 @@ static void alc1220_fixup_gb_dual_codecs(struct hda_codec *codec, } } +static void alc1220_fixup_gb_x570(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + static const hda_nid_t conn1[] = { 0x0c }; + static const struct coef_fw gb_x570_coefs[] = { + WRITE_COEF(0x1a, 0x01c1), + WRITE_COEF(0x1b, 0x0202), + WRITE_COEF(0x43, 0x3005), + {} + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_override_conn_list(codec, 0x14, ARRAY_SIZE(conn1), conn1); + snd_hda_override_conn_list(codec, 0x1b, ARRAY_SIZE(conn1), conn1); + break; + case HDA_FIXUP_ACT_INIT: + alc_process_coef_fw(codec, gb_x570_coefs); + break; + } +} + static void alc1220_fixup_clevo_p950(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -2415,6 +2439,10 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc1220_fixup_gb_dual_codecs, }, + [ALC1220_FIXUP_GB_X570] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc1220_fixup_gb_x570, + }, [ALC1220_FIXUP_CLEVO_P950] = { .type = HDA_FIXUP_FUNC, .v.func = alc1220_fixup_clevo_p950, @@ -2517,7 +2545,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x13fe, 0x1009, "Advantech MIT-W101", ALC886_FIXUP_EAPD), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), - SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_GB_X570), SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), -- cgit v1.2.3-71-gd317 From 8cd07657177006b67cc1610e4466cc75ad781c05 Mon Sep 17 00:00:00 2001 From: "Christian A. Ehrhardt" Date: Fri, 31 Dec 2021 14:12:21 +0100 Subject: ALSA: hda/cs8409: Increase delay during jack detection Commit c8b4f0865e82 reduced delays related to cs42l42 jack detection. However, the change was too aggressive. As a result internal speakers on DELL Inspirion 3501 are not detected. Increase the delay in cs42l42_run_jack_detect() a bit. Fixes: c8b4f0865e82 ("ALSA: hda/cs8409: Remove unnecessary delays") Signed-off-by: Christian A. Ehrhardt Link: https://lore.kernel.org/r/20211231131221.itwotyfk5qomn7n6@cae.in-ulm.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cs8409.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 31ff11ab868e..07eab788b145 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -628,8 +628,8 @@ static void cs42l42_run_jack_detect(struct sub_codec *cs42l42) cs8409_i2c_write(cs42l42, 0x1b74, 0x07); cs8409_i2c_write(cs42l42, 0x131b, 0xFD); cs8409_i2c_write(cs42l42, 0x1120, 0x80); - /* Wait ~100us*/ - usleep_range(100, 200); + /* Wait ~20ms*/ + usleep_range(20000, 25000); cs8409_i2c_write(cs42l42, 0x111f, 0x77); cs8409_i2c_write(cs42l42, 0x1120, 0xc0); } -- cgit v1.2.3-71-gd317 From 57f234248ff925d88caedf4019ec84e6ecb83909 Mon Sep 17 00:00:00 2001 From: "Christian A. Ehrhardt" Date: Fri, 31 Dec 2021 14:44:32 +0100 Subject: ALSA: hda/cs8409: Fix Jack detection after resume The suspend code unconditionally sets ->hp_jack_in and ->mic_jack_in to zero but without reporting this status change to the HDA core. To compensate for this, always assume a status change on the first unsol event after boot or resume. Fixes: 424e531b47f8 ("ALSA: hda/cs8409: Ensure Type Detection is only run on startup when necessary") Signed-off-by: Christian A. Ehrhardt Link: https://lore.kernel.org/r/20211231134432.atwmuzeceqiklcoa@cae.in-ulm.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cs8409-tables.c | 3 +++ sound/pci/hda/patch_cs8409.c | 5 ++++- sound/pci/hda/patch_cs8409.h | 1 + 3 files changed, 8 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c index 0fb0a428428b..df0b4522babf 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -252,6 +252,7 @@ struct sub_codec cs8409_cs42l42_codec = { .init_seq_num = ARRAY_SIZE(cs42l42_init_reg_seq), .hp_jack_in = 0, .mic_jack_in = 0, + .force_status_change = 1, .paged = 1, .suspended = 1, .no_type_dect = 0, @@ -443,6 +444,7 @@ struct sub_codec dolphin_cs42l42_0 = { .init_seq_num = ARRAY_SIZE(dolphin_c0_init_reg_seq), .hp_jack_in = 0, .mic_jack_in = 0, + .force_status_change = 1, .paged = 1, .suspended = 1, .no_type_dect = 0, @@ -456,6 +458,7 @@ struct sub_codec dolphin_cs42l42_1 = { .init_seq_num = ARRAY_SIZE(dolphin_c1_init_reg_seq), .hp_jack_in = 0, .mic_jack_in = 0, + .force_status_change = 1, .paged = 1, .suspended = 1, .no_type_dect = 1, diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 07eab788b145..9319ca879d01 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -636,7 +636,9 @@ static void cs42l42_run_jack_detect(struct sub_codec *cs42l42) static int cs42l42_handle_tip_sense(struct sub_codec *cs42l42, unsigned int reg_ts_status) { - int status_changed = 0; + int status_changed = cs42l42->force_status_change; + + cs42l42->force_status_change = 0; /* TIP_SENSE INSERT/REMOVE */ switch (reg_ts_status) { @@ -786,6 +788,7 @@ static void cs42l42_suspend(struct sub_codec *cs42l42) cs42l42->last_page = 0; cs42l42->hp_jack_in = 0; cs42l42->mic_jack_in = 0; + cs42l42->force_status_change = 1; /* Put CS42L42 into Reset */ gpio_data = snd_hda_codec_read(codec, CS8409_PIN_AFG, 0, AC_VERB_GET_GPIO_DATA, 0); diff --git a/sound/pci/hda/patch_cs8409.h b/sound/pci/hda/patch_cs8409.h index ade2b838590c..d0b725c7285b 100644 --- a/sound/pci/hda/patch_cs8409.h +++ b/sound/pci/hda/patch_cs8409.h @@ -305,6 +305,7 @@ struct sub_codec { unsigned int hp_jack_in:1; unsigned int mic_jack_in:1; + unsigned int force_status_change:1; unsigned int suspended:1; unsigned int paged:1; unsigned int last_page; -- cgit v1.2.3-71-gd317