From 6ade657d6125ec3ec07f95fa51e28138aef6208f Mon Sep 17 00:00:00 2001 From: Kangjie Lu Date: Thu, 14 Mar 2019 22:58:29 -0500 Subject: ALSA: echoaudio: add a check for ioremap_nocache In case ioremap_nocache fails, the fix releases chip and returns an error code upstream to avoid NULL pointer dereference. Signed-off-by: Kangjie Lu Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index ea876b0b02b9..dc0084dc8550 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1952,6 +1952,11 @@ static int snd_echo_create(struct snd_card *card, } chip->dsp_registers = (volatile u32 __iomem *) ioremap_nocache(chip->dsp_registers_phys, sz); + if (!chip->dsp_registers) { + dev_err(chip->card->dev, "ioremap failed\n"); + snd_echo_free(chip); + return -ENOMEM; + } if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { -- cgit v1.2.3 From dcd0feac9bab901d5739de51b3f69840851f8919 Mon Sep 17 00:00:00 2001 From: Kangjie Lu Date: Thu, 14 Mar 2019 23:04:14 -0500 Subject: ALSA: sb8: add a check for request_region In case request_region fails, the fix returns an error code to avoid NULL pointer dereference. Signed-off-by: Kangjie Lu Signed-off-by: Takashi Iwai --- sound/isa/sb/sb8.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index aa2a83eb81a9..dc27a480c2d9 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -111,6 +111,10 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev) /* block the 0x388 port to avoid PnP conflicts */ acard->fm_res = request_region(0x388, 4, "SoundBlaster FM"); + if (!acard->fm_res) { + err = -EBUSY; + goto _err; + } if (port[dev] != SNDRV_AUTO_PORT) { if ((err = snd_sbdsp_create(card, port[dev], irq[dev], -- cgit v1.2.3 From 2d012c65a9ca26a0ef87ea0a42f1653dd37155f5 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 17 Mar 2019 15:49:29 +0900 Subject: ALSA: firewire-motu: use 'version' field of unit directory to identify model Current ALSA firewire-motu driver uses the value of 'model' field of unit directory in configuration ROM for modalias for MOTU FireWire models. However, as long as I checked, Pre8 and 828mk3(Hybrid) have the same value for the field (=0x100800). unit | version | model --------------- | --------- | ---------- 828mkII | 0x000003 | 0x101800 Traveler | 0x000009 | 0x107800 Pre8 | 0x00000f | 0x100800 <- 828mk3(FW) | 0x000015 | 0x106800 AudioExpress | 0x000033 | 0x104800 828mk3(Hybrid) | 0x000035 | 0x100800 <- When updating firmware for MOTU 8pre FireWire from v1.0.0 to v1.0.3, I got change of the value from 0x100800 to 0x103800. On the other hand, the value of 'version' field is fixed to 0x00000f. As a quick glance, the higher 12 bits of the value of 'version' field represent firmware version, while the lower 12 bits is unknown. By induction, the value of 'version' field represents actual model. This commit changes modalias to match the value of 'version' field, instead of 'model' field. For degug, long name of added sound card includes hexadecimal value of 'model' field. Fixes: 6c5e1ac0e144 ("ALSA: firewire-motu: add support for Motu Traveler") Signed-off-by: Takashi Sakamoto Cc: # v4.19+ Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 220e61926ea4..513291ba0ab0 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -36,7 +36,7 @@ static void name_card(struct snd_motu *motu) fw_csr_iterator_init(&it, motu->unit->directory); while (fw_csr_iterator_next(&it, &key, &val)) { switch (key) { - case CSR_VERSION: + case CSR_MODEL: version = val; break; } @@ -46,7 +46,7 @@ static void name_card(struct snd_motu *motu) strcpy(motu->card->shortname, motu->spec->name); strcpy(motu->card->mixername, motu->spec->name); snprintf(motu->card->longname, sizeof(motu->card->longname), - "MOTU %s (version:%d), GUID %08x%08x at %s, S%d", + "MOTU %s (version:%06x), GUID %08x%08x at %s, S%d", motu->spec->name, version, fw_dev->config_rom[3], fw_dev->config_rom[4], dev_name(&motu->unit->device), 100 << fw_dev->max_speed); @@ -237,20 +237,20 @@ static const struct snd_motu_spec motu_audio_express = { #define SND_MOTU_DEV_ENTRY(model, data) \ { \ .match_flags = IEEE1394_MATCH_VENDOR_ID | \ - IEEE1394_MATCH_MODEL_ID | \ - IEEE1394_MATCH_SPECIFIER_ID, \ + IEEE1394_MATCH_SPECIFIER_ID | \ + IEEE1394_MATCH_VERSION, \ .vendor_id = OUI_MOTU, \ - .model_id = model, \ .specifier_id = OUI_MOTU, \ + .version = model, \ .driver_data = (kernel_ulong_t)data, \ } static const struct ieee1394_device_id motu_id_table[] = { - SND_MOTU_DEV_ENTRY(0x101800, &motu_828mk2), - SND_MOTU_DEV_ENTRY(0x107800, &snd_motu_spec_traveler), - SND_MOTU_DEV_ENTRY(0x106800, &motu_828mk3), /* FireWire only. */ - SND_MOTU_DEV_ENTRY(0x100800, &motu_828mk3), /* Hybrid. */ - SND_MOTU_DEV_ENTRY(0x104800, &motu_audio_express), + SND_MOTU_DEV_ENTRY(0x000003, &motu_828mk2), + SND_MOTU_DEV_ENTRY(0x000009, &snd_motu_spec_traveler), + SND_MOTU_DEV_ENTRY(0x000015, &motu_828mk3), /* FireWire only. */ + SND_MOTU_DEV_ENTRY(0x000035, &motu_828mk3), /* Hybrid. */ + SND_MOTU_DEV_ENTRY(0x000033, &motu_audio_express), { } }; MODULE_DEVICE_TABLE(ieee1394, motu_id_table); -- cgit v1.2.3 From 721f1e6c1fd137e7e2053d8e103b666faaa2d50c Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 18 Mar 2019 13:45:43 +0100 Subject: ALSA: hda - add Lenovo IdeaCentre B550 to the power_save_blacklist Another machine which does not like the power saving (noise): https://bugzilla.redhat.com/show_bug.cgi?id=1689623 Also, reorder the Lenovo C50 entry to keep the table sorted. Reported-by: hs.guimaraes@outlook.com Signed-off-by: Jaroslav Kysela Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e5c49003e75f..4f502c92061f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2144,10 +2144,12 @@ static struct snd_pci_quirk power_save_blacklist[] = { SND_PCI_QUIRK(0x8086, 0x2057, "Intel NUC5i7RYB", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1520902 */ SND_PCI_QUIRK(0x8086, 0x2068, "Intel NUC7i3BNB", 0), - /* https://bugzilla.redhat.com/show_bug.cgi?id=1572975 */ - SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugzilla.kernel.org/show_bug.cgi?id=198611 */ SND_PCI_QUIRK(0x17aa, 0x2227, "Lenovo X1 Carbon 3rd Gen", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1689623 */ + SND_PCI_QUIRK(0x17aa, 0x367b, "Lenovo IdeaCentre B550", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1572975 */ + SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), {} }; #endif /* CONFIG_PM */ -- cgit v1.2.3 From b4748e7ab731e436cf5db4786358ada5dd2db6dd Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Sun, 17 Mar 2019 23:21:24 +0000 Subject: ALSA: opl3: fix mismatch between snd_opl3_drum_switch definition and declaration The function snd_opl3_drum_switch declaration in the header file has the order of the two arguments on_off and vel swapped when compared to the definition arguments of vel and on_off. Fix this by swapping them around to match the definition. This error predates the git history, so no idea when this error was introduced. Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/drivers/opl3/opl3_voice.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/opl3/opl3_voice.h b/sound/drivers/opl3/opl3_voice.h index 5b02bd49fde4..4e4ecc21760b 100644 --- a/sound/drivers/opl3/opl3_voice.h +++ b/sound/drivers/opl3/opl3_voice.h @@ -41,7 +41,7 @@ void snd_opl3_timer_func(struct timer_list *t); /* Prototypes for opl3_drums.c */ void snd_opl3_load_drums(struct snd_opl3 *opl3); -void snd_opl3_drum_switch(struct snd_opl3 *opl3, int note, int on_off, int vel, struct snd_midi_channel *chan); +void snd_opl3_drum_switch(struct snd_opl3 *opl3, int note, int vel, int on_off, struct snd_midi_channel *chan); /* Prototypes for opl3_oss.c */ #if IS_ENABLED(CONFIG_SND_SEQUENCER_OSS) -- cgit v1.2.3 From 744c67ffeb06f2d2493f4049ba0bd19698ce0adf Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Tue, 19 Mar 2019 09:28:43 +0800 Subject: ALSA: hda - Don't trigger jackpoll_work in azx_resume The commit 3baffc4a84d7 (ALSA: hda/intel: Refactoring PM code) changed the behaviour of azx_resume(), it triggers the jackpoll_work after applying this commit. This change introduced a new issue, all codecs are runtime active after S3, and will not call runtime_suspend() automatically. The root cause is the jackpoll_work calls snd_hda_power_up/down_pm, and it calls up_pm before snd_hdac_enter_pm is called, while calls the down_pm in the middle of enter_pm and leave_pm is called. This makes the dev->power.usage_count unbalanced after S3. To fix it, let azx_resume() don't trigger jackpoll_work as before it did. Fixes: 3baffc4a84d7 ("ALSA: hda/intel: Refactoring PM code") Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4f502c92061f..ece256a3b48f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -947,7 +947,7 @@ static void __azx_runtime_suspend(struct azx *chip) display_power(chip, false); } -static void __azx_runtime_resume(struct azx *chip) +static void __azx_runtime_resume(struct azx *chip, bool from_rt) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hdac_bus *bus = azx_bus(chip); @@ -964,7 +964,7 @@ static void __azx_runtime_resume(struct azx *chip) azx_init_pci(chip); hda_intel_init_chip(chip, true); - if (status) { + if (status && from_rt) { list_for_each_codec(codec, &chip->bus) if (status & (1 << codec->addr)) schedule_delayed_work(&codec->jackpoll_work, @@ -1016,7 +1016,7 @@ static int azx_resume(struct device *dev) chip->msi = 0; if (azx_acquire_irq(chip, 1) < 0) return -EIO; - __azx_runtime_resume(chip); + __azx_runtime_resume(chip, false); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); @@ -1081,7 +1081,7 @@ static int azx_runtime_resume(struct device *dev) chip = card->private_data; if (!azx_has_pm_runtime(chip)) return 0; - __azx_runtime_resume(chip); + __azx_runtime_resume(chip, true); /* disable controller Wake Up event*/ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & -- cgit v1.2.3 From b5a236c175b0d984552a5f7c9d35141024c2b261 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Tue, 19 Mar 2019 09:28:44 +0800 Subject: ALSA: hda - Enforces runtime_resume after S3 and S4 for each codec Recently we found the audio jack detection stop working after suspend on many machines with Realtek codec. Sometimes the audio selection dialogue didn't show up after users plugged headhphone/headset into the headset jack, sometimes after uses plugged headphone/headset, then click the sound icon on the upper-right corner of gnome-desktop, it also showed the speaker rather than the headphone. The root cause is that before suspend, the codec already call the runtime_suspend since this codec is not used by any apps, then in resume, it will not call runtime_resume for this codec. But for some realtek codec (so far, alc236, alc255 and alc891) with the specific BIOS, if it doesn't run runtime_resume after suspend, all codec functions including jack detection stop working anymore. This problem existed for a long time, but it was not exposed, that is because when problem happens, if users play sound or open sound-setting to check audio device, this will trigger calling to runtime_resume (via snd_hda_power_up), then the codec starts working again before users notice this problem. Since we don't know how many codec and BIOS combinations have this problem, to fix it, let the driver call runtime_resume for all codecs in pm_resume, maybe for some codecs, this is not needed, but it is harmless. After a codec is runtime resumed, if it is not used by any apps, it will be runtime suspended soon and furthermore we don't run suspend frequently, this change will not add much power consumption. Fixes: cc72da7d4d06 ("ALSA: hda - Use standard runtime PM for codec power-save control") Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5f2005098a60..ec0b8595eb4d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2939,6 +2939,20 @@ static int hda_codec_runtime_resume(struct device *dev) #endif /* CONFIG_PM */ #ifdef CONFIG_PM_SLEEP +static int hda_codec_force_resume(struct device *dev) +{ + int ret; + + /* The get/put pair below enforces the runtime resume even if the + * device hasn't been used at suspend time. This trick is needed to + * update the jack state change during the sleep. + */ + pm_runtime_get_noresume(dev); + ret = pm_runtime_force_resume(dev); + pm_runtime_put(dev); + return ret; +} + static int hda_codec_pm_suspend(struct device *dev) { dev->power.power_state = PMSG_SUSPEND; @@ -2948,7 +2962,7 @@ static int hda_codec_pm_suspend(struct device *dev) static int hda_codec_pm_resume(struct device *dev) { dev->power.power_state = PMSG_RESUME; - return pm_runtime_force_resume(dev); + return hda_codec_force_resume(dev); } static int hda_codec_pm_freeze(struct device *dev) @@ -2960,13 +2974,13 @@ static int hda_codec_pm_freeze(struct device *dev) static int hda_codec_pm_thaw(struct device *dev) { dev->power.power_state = PMSG_THAW; - return pm_runtime_force_resume(dev); + return hda_codec_force_resume(dev); } static int hda_codec_pm_restore(struct device *dev) { dev->power.power_state = PMSG_RESTORE; - return pm_runtime_force_resume(dev); + return hda_codec_force_resume(dev); } #endif /* CONFIG_PM_SLEEP */ -- cgit v1.2.3 From 667a8f73753908c4d0171e52b71774f9be5d6713 Mon Sep 17 00:00:00 2001 From: Jian-Hong Pan Date: Fri, 15 Mar 2019 17:51:09 +0800 Subject: ALSA: hda/realtek: Enable headset MIC of Acer AIO with ALC286 Some Acer AIO desktops like Veriton Z6860G, Z4860G and Z4660G cannot record sound from headset MIC. This patch adds the ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk to fix this issue. Fixes: 9f8aefed9623 ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4660G") Fixes: b72f936f6b32 ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4860G/Z6860G") Signed-off-by: Jian-Hong Pan Reviewed-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 384719d5c44e..191830d4fa40 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5687,6 +5687,7 @@ enum { ALC225_FIXUP_DELL_WYSE_AIO_MIC_NO_PRESENCE, ALC225_FIXUP_WYSE_AUTO_MUTE, ALC225_FIXUP_WYSE_DISABLE_MIC_VREF, + ALC286_FIXUP_ACER_AIO_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -6685,6 +6686,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC286_FIXUP_ACER_AIO_HEADSET_MIC] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x4f }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x5029 }, + { } + }, + .chained = true, + .chain_id = ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6701,9 +6712,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), - SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), -- cgit v1.2.3