From 977dfef40c8996b69afe23a9094d184049efb7bb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 Apr 2020 08:12:22 +0200 Subject: ALSA: hda: Match both PCI ID and SSID for driver blacklist The commit 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") added a new blacklist for the devices that are known to have empty codecs, and one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f). However, it turned out that the very same PCI SSID is used for the previous model that does have the valid HD-audio codecs and the change broke the sound on it. Since the empty codec problem appear on the certain AMD platform (PCI ID 1022:1487), this patch changes the blacklist matching to both PCI ID and SSID using pci_match_id(). Also, the entry that was removed by the previous fix for ASUS ROG Zenigh II is re-added. Link: https://lore.kernel.org/r/20200424061222.19792-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 457a2c065485..0310193ea1bd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2078,9 +2078,10 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, * some HD-audio PCI entries are exposed without any codecs, and such devices * should be ignored from the beginning. */ -static const struct snd_pci_quirk driver_blacklist[] = { - SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), - SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), +static const struct pci_device_id driver_blacklist[] = { + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1043, 0x874f) }, /* ASUS ROG Zenith II / Strix */ + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb59) }, /* MSI TRX40 Creator */ + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb60) }, /* MSI TRX40 */ {} }; @@ -2100,7 +2101,7 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; - if (snd_pci_quirk_lookup(pci, driver_blacklist)) { + if (pci_match_id(driver_blacklist, pci)) { dev_info(&pci->dev, "Skipping the blacklisted device\n"); return -ENODEV; } -- cgit v1.2.3-58-ga151 From 4285de0725b1bf73608abbcd35ad7fd3ddc0b61e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 Apr 2020 21:33:50 +0200 Subject: ALSA: pcm: oss: Place the plugin buffer overflow checks correctly The checks of the plugin buffer overflow in the previous fix by commit f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow") are put in the wrong places mistakenly, which leads to the expected (repeated) sound when the rate plugin is involved. Fix in the right places. Also, at those right places, the zero check is needed for the termination node, so added there as well, and let's get it done, finally. Fixes: f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow") Cc: Link: https://lore.kernel.org/r/20200424193350.19678-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_plugin.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 50c35ecc8953..d1760f86773c 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -211,21 +211,23 @@ static snd_pcm_sframes_t plug_client_size(struct snd_pcm_substream *plug, if (stream == SNDRV_PCM_STREAM_PLAYBACK) { plugin = snd_pcm_plug_last(plug); while (plugin && drv_frames > 0) { - if (check_size && drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) drv_frames = plugin->src_frames(plugin, drv_frames); + if (check_size && plugin->buf_frames && + drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin = plugin_prev; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_first(plug); while (plugin && drv_frames > 0) { plugin_next = plugin->next; + if (check_size && plugin->buf_frames && + drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; if (plugin->dst_frames) drv_frames = plugin->dst_frames(plugin, drv_frames); - if (check_size && drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; plugin = plugin_next; } } else @@ -251,26 +253,28 @@ static snd_pcm_sframes_t plug_slave_size(struct snd_pcm_substream *plug, plugin = snd_pcm_plug_first(plug); while (plugin && frames > 0) { plugin_next = plugin->next; + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; if (plugin->dst_frames) { frames = plugin->dst_frames(plugin, frames); if (frames < 0) return frames; } - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin = plugin_next; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_last(plug); while (plugin) { - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); if (frames < 0) return frames; } + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_prev; } } else -- cgit v1.2.3-58-ga151 From ac957e8c54115c1ed32e41e0072af3a63576cda6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 Apr 2020 21:38:43 +0200 Subject: ALSA: pcm: oss: Place the plugin buffer overflow checks correctly (for 5.7) [ This is again a forward-port of the fix applied for 5.6-base code (commit 4285de0725b1) to 5.7-base, hence neither Fixes nor Cc-to-stable tags are included here -- tiwai ] The checks of the plugin buffer overflow in the previous fix by commit f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow") are put in the wrong places mistakenly, which leads to the expected (repeated) sound when the rate plugin is involved. Fix in the right places. Also, at those right places, the zero check is needed for the termination node, so added there as well, and let's get it done, finally. Link: https://lore.kernel.org/r/20200424193843.20397-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_plugin.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 59d62f05658f..1545f8fdb4db 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -205,13 +205,14 @@ static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug, plugin = snd_pcm_plug_first(plug); while (plugin && frames > 0) { plugin_next = plugin->next; + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; if (plugin->dst_frames) { frames = plugin->dst_frames(plugin, frames); if (frames < 0) return frames; } - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin = plugin_next; } return frames; @@ -225,14 +226,15 @@ static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug, plugin = snd_pcm_plug_last(plug); while (plugin && frames > 0) { - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); if (frames < 0) return frames; } + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_prev; } return frames; -- cgit v1.2.3-58-ga151 From cc18b2f4f3f1d7ed3125ac1840794f9feab0325c Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Sat, 25 Apr 2020 13:11:15 -0700 Subject: ALSA: line6: Fix POD HD500 audio playback Apparently interface 1 is control interface akin to HD500X, setting LINE6_CAP_CONTROL and choosing it as ctrl_if fixes audio playback on POD HD500. Signed-off-by: Vasily Khoruzhick Cc: Link: https://lore.kernel.org/r/20200425201115.3430-1-anarsoul@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/line6/podhd.c | 22 +++++----------------- 1 file changed, 5 insertions(+), 17 deletions(-) diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index d37db32ecd3b..e39dc85c355a 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -21,8 +21,7 @@ enum { LINE6_PODHD300, LINE6_PODHD400, - LINE6_PODHD500_0, - LINE6_PODHD500_1, + LINE6_PODHD500, LINE6_PODX3, LINE6_PODX3LIVE, LINE6_PODHD500X, @@ -318,8 +317,7 @@ static const struct usb_device_id podhd_id_table[] = { /* TODO: no need to alloc data interfaces when only audio is used */ { LINE6_DEVICE(0x5057), .driver_info = LINE6_PODHD300 }, { LINE6_DEVICE(0x5058), .driver_info = LINE6_PODHD400 }, - { LINE6_IF_NUM(0x414D, 0), .driver_info = LINE6_PODHD500_0 }, - { LINE6_IF_NUM(0x414D, 1), .driver_info = LINE6_PODHD500_1 }, + { LINE6_IF_NUM(0x414D, 0), .driver_info = LINE6_PODHD500 }, { LINE6_IF_NUM(0x414A, 0), .driver_info = LINE6_PODX3 }, { LINE6_IF_NUM(0x414B, 0), .driver_info = LINE6_PODX3LIVE }, { LINE6_IF_NUM(0x4159, 0), .driver_info = LINE6_PODHD500X }, @@ -352,23 +350,13 @@ static const struct line6_properties podhd_properties_table[] = { .ep_audio_r = 0x82, .ep_audio_w = 0x01, }, - [LINE6_PODHD500_0] = { + [LINE6_PODHD500] = { .id = "PODHD500", .name = "POD HD500", - .capabilities = LINE6_CAP_PCM + .capabilities = LINE6_CAP_PCM | LINE6_CAP_CONTROL | LINE6_CAP_HWMON, .altsetting = 1, - .ep_ctrl_r = 0x81, - .ep_ctrl_w = 0x01, - .ep_audio_r = 0x86, - .ep_audio_w = 0x02, - }, - [LINE6_PODHD500_1] = { - .id = "PODHD500", - .name = "POD HD500", - .capabilities = LINE6_CAP_PCM - | LINE6_CAP_HWMON, - .altsetting = 0, + .ctrl_if = 1, .ep_ctrl_r = 0x81, .ep_ctrl_w = 0x01, .ep_audio_r = 0x86, -- cgit v1.2.3-58-ga151 From ef0b3203c758b6b8abdb5dca651880347eae6b8c Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Mon, 27 Apr 2020 11:00:39 +0800 Subject: ALSA: hda/realtek - Two front mics on a Lenovo ThinkCenter This new Lenovo ThinkCenter has two front mics which can't be handled by PA so far, so apply the fixup ALC283_FIXUP_HEADSET_MIC to change the location for one of the mics. Cc: Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20200427030039.10121-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c1a85c8f7b69..c16f63957c5a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7420,6 +7420,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), + SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), -- cgit v1.2.3-58-ga151 From ca76282b6faffc83601c25bd2a95f635c03503ef Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Tue, 28 Apr 2020 15:38:36 +0300 Subject: ALSA: hda/hdmi: fix race in monitor detection during probe A race exists between build_pcms() and build_controls() phases of codec setup. Build_pcms() sets up notifier for jack events. If a monitor event is received before build_controls() is run, the initial jack state is lost and never reported via mixer controls. The problem can be hit at least with SOF as the controller driver. SOF calls snd_hda_codec_build_controls() in its workqueue-based probe and this can be delayed enough to hit the race condition. Fix the issue by invalidating the per-pin ELD information when build_controls() is called. The existing call to hdmi_present_sense() will update the ELD contents. This ensures initial monitor state is correctly reflected via mixer controls. BugLink: https://github.com/thesofproject/linux/issues/1687 Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200428123836.24512-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4eff16053bd5..2e2c382fe4b5 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2198,7 +2198,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + struct hdmi_eld *pin_eld = &per_pin->sink_eld; + pin_eld->eld_valid = false; hdmi_present_sense(per_pin, 0); } -- cgit v1.2.3-58-ga151 From a2f647240998aa49632fb09b01388fdf2b87acfc Mon Sep 17 00:00:00 2001 From: Wu Bo Date: Sun, 26 Apr 2020 21:17:22 +0800 Subject: ALSA: hda/hdmi: fix without unlocked before return Fix the following coccicheck warning: sound/pci/hda/patch_hdmi.c:1852:2-8: preceding lock on line 1846 After add sanity check to pass klockwork check, The spdif_mutex should be unlock before return true in check_non_pcm_per_cvt(). Fixes: 960a581e22d9 ("ALSA: hda: fix some klockwork scan warnings") Signed-off-by: Wu Bo Cc: Link: https://lore.kernel.org/r/1587907042-694161-1-git-send-email-wubo40@huawei.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2e2c382fe4b5..93760a3564cf 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1848,8 +1848,10 @@ static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) /* Add sanity check to pass klockwork check. * This should never happen. */ - if (WARN_ON(spdif == NULL)) + if (WARN_ON(spdif == NULL)) { + mutex_unlock(&codec->spdif_mutex); return true; + } non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO); mutex_unlock(&codec->spdif_mutex); return non_pcm; -- cgit v1.2.3-58-ga151 From 5ce00760a84848d008554c693ceb6286f4d9c509 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 29 Apr 2020 21:02:03 +0200 Subject: ALSA: opti9xx: shut up gcc-10 range warning gcc-10 points out a few instances of suspicious integer arithmetic leading to value truncation: sound/isa/opti9xx/opti92x-ad1848.c: In function 'snd_opti9xx_configure': sound/isa/opti9xx/opti92x-ad1848.c:322:43: error: overflow in conversion from 'int' to 'unsigned char' changes value from '(int)snd_opti9xx_read(chip, 3) & -256 | 240' to '240' [-Werror=overflow] 322 | (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~ sound/isa/opti9xx/opti92x-ad1848.c:351:3: note: in expansion of macro 'snd_opti9xx_write_mask' 351 | snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); | ^~~~~~~~~~~~~~~~~~~~~~ sound/isa/opti9xx/miro.c: In function 'snd_miro_configure': sound/isa/opti9xx/miro.c:873:40: error: overflow in conversion from 'int' to 'unsigned char' changes value from '(int)snd_miro_read(chip, 3) & -256 | 240' to '240' [-Werror=overflow] 873 | (snd_miro_read(chip, reg) & ~(mask)) | ((value) & (mask))) | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~ sound/isa/opti9xx/miro.c:1010:3: note: in expansion of macro 'snd_miro_write_mask' 1010 | snd_miro_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); | ^~~~~~~~~~~~~~~~~~~ These are all harmless here as only the low 8 bit are passed down anyway. Change the macros to inline functions to make the code more readable and also avoid the warning. Strictly speaking those functions also need locking to make the read/write pair atomic, but it seems unlikely that anyone would still run into that issue. Fixes: 1841f613fd2e ("[ALSA] Add snd-miro driver") Signed-off-by: Arnd Bergmann Link: https://lore.kernel.org/r/20200429190216.85919-1-arnd@arndb.de Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/miro.c | 9 ++++++--- sound/isa/opti9xx/opti92x-ad1848.c | 9 ++++++--- 2 files changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index e764816a8f7a..b039429e6871 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -867,10 +867,13 @@ static void snd_miro_write(struct snd_miro *chip, unsigned char reg, spin_unlock_irqrestore(&chip->lock, flags); } +static inline void snd_miro_write_mask(struct snd_miro *chip, + unsigned char reg, unsigned char value, unsigned char mask) +{ + unsigned char oldval = snd_miro_read(chip, reg); -#define snd_miro_write_mask(chip, reg, value, mask) \ - snd_miro_write(chip, reg, \ - (snd_miro_read(chip, reg) & ~(mask)) | ((value) & (mask))) + snd_miro_write(chip, reg, (oldval & ~mask) | (value & mask)); +} /* * Proc Interface diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index d06b29693c85..0e6d20e49158 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -317,10 +317,13 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, } -#define snd_opti9xx_write_mask(chip, reg, value, mask) \ - snd_opti9xx_write(chip, reg, \ - (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) +static inline void snd_opti9xx_write_mask(struct snd_opti9xx *chip, + unsigned char reg, unsigned char value, unsigned char mask) +{ + unsigned char oldval = snd_opti9xx_read(chip, reg); + snd_opti9xx_write(chip, reg, (oldval & ~mask) | (value & mask)); +} static int snd_opti9xx_configure(struct snd_opti9xx *chip, long port, -- cgit v1.2.3-58-ga151 From 547d2c9cf4f1f72adfecacbd5b093681fb0e8b3e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Apr 2020 14:47:55 +0200 Subject: ALSA: usb-audio: Correct a typo of NuPrime DAC-10 USB ID The USB vendor ID of NuPrime DAC-10 is not 16b0 but 16d0. Fixes: f656891c6619 ("ALSA: usb-audio: add more quirks for DSD interfaces") Cc: Link: https://lore.kernel.org/r/20200430124755.15940-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 351ba214a9d3..848a4cc25bed 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1687,7 +1687,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */ - case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ + case USB_ID(0x16d0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ -- cgit v1.2.3-58-ga151