From 74f73476c3755664504b7d266b5e8f91050ffed9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 11 Feb 2020 12:14:19 +0100 Subject: ALSA: usb-audio: Apply 48kHz fixed rate playback for Jabra Evolve 65 headset Jabra Evolve 65 headset appears as if supporting lower rates than 48kHz, but it actually doesn't work but with 48kHz for playback. This patch applies a workaround to enforce the 48kHz like LINE6 devices already did. The workaround is put in a unified helper function, set_fixed_rate(), to be called from both places now. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206149 Link: https://lore.kernel.org/r/20200211111419.5895-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/format.c | 33 ++++++++++++++++++++++----------- 1 file changed, 22 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/usb/format.c b/sound/usb/format.c index 9260136e4c9b..50cb183958bf 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -151,6 +151,19 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, return pcm_formats; } +static int set_fixed_rate(struct audioformat *fp, int rate, int rate_bits) +{ + kfree(fp->rate_table); + fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); + if (!fp->rate_table) + return -ENOMEM; + fp->nr_rates = 1; + fp->rate_min = rate; + fp->rate_max = rate; + fp->rates = rate_bits; + fp->rate_table[0] = rate; + return 0; +} /* * parse the format descriptor and stores the possible sample rates @@ -223,6 +236,14 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof fp->rate_min = combine_triple(&fmt[offset + 1]); fp->rate_max = combine_triple(&fmt[offset + 4]); } + + /* Jabra Evolve 65 headset */ + if (chip->usb_id == USB_ID(0x0b0e, 0x030b)) { + /* only 48kHz for playback while keeping 16kHz for capture */ + if (fp->nr_rates != 1) + return set_fixed_rate(fp, 48000, SNDRV_PCM_RATE_48000); + } + return 0; } @@ -299,17 +320,7 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0e41, 0x4248): /* Line6 Helix >= fw 2.82 */ case USB_ID(0x0e41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */ case USB_ID(0x0e41, 0x424a): /* Line6 Helix LT >= fw 2.82 */ - /* supported rates: 48Khz */ - kfree(fp->rate_table); - fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); - if (!fp->rate_table) - return -ENOMEM; - fp->nr_rates = 1; - fp->rate_min = 48000; - fp->rate_max = 48000; - fp->rates = SNDRV_PCM_RATE_48000; - fp->rate_table[0] = 48000; - return 0; + return set_fixed_rate(fp, 48000, SNDRV_PCM_RATE_48000); } return -ENODEV; -- cgit v1.2.3 From d75a170fd848f037a1e28893ad10be7a4c51f8a6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 11 Feb 2020 17:05:21 +0100 Subject: ALSA: usb-audio: Fix UAC2/3 effect unit parsing We've got a regression report about M-Audio Fast Track C400 device, and the git bisection resulted in the commit e0ccdef92653 ("ALSA: usb-audio: Clean up check_input_term()"). This commit was about the rewrite of the input terminal parser, and it's not too obvious from the change what really broke. The answer is: it's the interpretation of UAC2/3 effect units. In the original code, UAC2 effect unit is as if through UAC1 processing unit because both UAC1 PU and UAC2/3 EU share the same number (0x07). The old code went through a complex switch-case fallthrough, finally bailing out in the middle: if (protocol == UAC_VERSION_2 && hdr[2] == UAC2_EFFECT_UNIT) { /* UAC2/UAC1 unit IDs overlap here in an * uncompatible way. Ignore this unit for now. */ return 0; } ... and this special handling was missing in the new code; the new code treats UAC2/3 effect unit as if it were equivalent with the processing unit. Actually, the old code was too confusing. The effect unit has an incompatible unit description with the processing unit, so we shouldn't have dealt with EU in the same way. This patch addresses the regression by changing the effect unit handling to the own parser function. The own parser function makes the clear distinct with PU, so it improves the readability, too. The EU parser just sets the type and the id like the old kernels. Once when the proper effect unit support is added, we can revisit this parser function, but for now, let's keep this simple setup as is. Fixes: e0ccdef92653 ("ALSA: usb-audio: Clean up check_input_term()") Cc: BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206147 Link: https://lore.kernel.org/r/20200211160521.31990-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index d659fdb475e2..81b2db0edd5f 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -897,6 +897,15 @@ static int parse_term_proc_unit(struct mixer_build *state, return 0; } +static int parse_term_effect_unit(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) +{ + term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */ + term->id = id; + return 0; +} + static int parse_term_uac2_clock_source(struct mixer_build *state, struct usb_audio_term *term, void *p1, int id) @@ -981,8 +990,7 @@ static int __check_input_term(struct mixer_build *state, int id, UAC3_PROCESSING_UNIT); case PTYPE(UAC_VERSION_2, UAC2_EFFECT_UNIT): case PTYPE(UAC_VERSION_3, UAC3_EFFECT_UNIT): - return parse_term_proc_unit(state, term, p1, id, - UAC3_EFFECT_UNIT); + return parse_term_effect_unit(state, term, p1, id); case PTYPE(UAC_VERSION_1, UAC1_EXTENSION_UNIT): case PTYPE(UAC_VERSION_2, UAC2_EXTENSION_UNIT_V2): case PTYPE(UAC_VERSION_3, UAC3_EXTENSION_UNIT): -- cgit v1.2.3 From 93f9d1a4ac5930654c17412e3911b46ece73755a Mon Sep 17 00:00:00 2001 From: Arvind Sankar Date: Tue, 11 Feb 2020 11:22:35 -0500 Subject: ALSA: usb-audio: Apply sample rate quirk for Audioengine D1 The Audioengine D1 (0x2912:0x30c8) does support reading the sample rate, but it returns the rate in byte-reversed order. When setting sampling rate, the driver produces these warning messages: [168840.944226] usb 3-2.2: current rate 4500480 is different from the runtime rate 44100 [168854.930414] usb 3-2.2: current rate 8436480 is different from the runtime rate 48000 [168905.185825] usb 3-2.1.2: current rate 30465 is different from the runtime rate 96000 As can be seen from the hexadecimal conversion, the current rate read back is byte-reversed from the rate that was set. 44100 == 0x00ac44, 4500480 == 0x44ac00 48000 == 0x00bb80, 8436480 == 0x80bb00 96000 == 0x017700, 30465 == 0x007701 Rather than implementing a new quirk to reverse the order, just skip checking the rate to avoid spamming the log. Signed-off-by: Arvind Sankar Cc: Link: https://lore.kernel.org/r/20200211162235.1639889-1-nivedita@alum.mit.edu Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 3a5242e383b2..7f558f4b4520 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1440,6 +1440,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x1395, 0x740a): /* Sennheiser DECT */ case USB_ID(0x1901, 0x0191): /* GE B850V3 CP2114 audio interface */ case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ + case USB_ID(0x2912, 0x30c8): /* Audioengine D1 */ return true; } -- cgit v1.2.3 From 2b3b6497c38d123934de68ea82a247b557d95290 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Mon, 10 Feb 2020 16:15:14 +0800 Subject: ALSA: hda/realtek - Add more codec supported Headset Button Add supported Headset Button for ALC215/ALC285/ALC289. Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/r/948f70b4488f4cc2b629a39ce4e4be33@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4770fb3f51fb..3ee88adf57e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5701,8 +5701,11 @@ static void alc_fixup_headset_jack(struct hda_codec *codec, break; case HDA_FIXUP_ACT_INIT: switch (codec->core.vendor_id) { + case 0x10ec0215: case 0x10ec0225: + case 0x10ec0285: case 0x10ec0295: + case 0x10ec0289: case 0x10ec0299: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); -- cgit v1.2.3 From 7dafba3762d6c0083ded00a48f8c1a158bc86717 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 Feb 2020 09:10:47 +0100 Subject: ALSA: hda/realtek - Fix silent output on MSI-GL73 MSI-GL73 laptop with ALC1220 codec requires a similar workaround for Clevo laptops to enforce the DAC/mixer connection path. Set up a quirk entry for that. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204159 Cc: Link: https://lore.kernel.org/r/20200212081047.27727-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3ee88adf57e7..6c8cb4ce517e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2447,6 +2447,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_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(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), -- cgit v1.2.3 From 9f35a31283775e6f6af73fb2c95c686a4c0acac7 Mon Sep 17 00:00:00 2001 From: Alexander Tsoy Date: Thu, 13 Feb 2020 02:54:50 +0300 Subject: ALSA: usb-audio: Add clock validity quirk for Denon MC7000/MCX8000 It should be safe to ignore clock validity check result if the following conditions are met: - only one single sample rate is supported; - the terminal is directly connected to the clock source; - the clock type is internal. This is to deal with some Denon DJ controllers that always reports that clock is invalid. Tested-by: Tobias Oszlanyi Signed-off-by: Alexander Tsoy Cc: Link: https://lore.kernel.org/r/20200212235450.697348-1-alexander@tsoy.me Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 91 +++++++++++++++++++++++++++++++++++++----------------- sound/usb/clock.h | 4 +-- sound/usb/format.c | 3 +- 3 files changed, 66 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 018b1ecb5404..a48313dfa967 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -151,8 +151,34 @@ static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_i return ret; } +/* + * Assume the clock is valid if clock source supports only one single sample + * rate, the terminal is connected directly to it (there is no clock selector) + * and clock type is internal. This is to deal with some Denon DJ controllers + * that always reports that clock is invalid. + */ +static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, + struct audioformat *fmt, + int source_id) +{ + if (fmt->protocol == UAC_VERSION_2) { + struct uac_clock_source_descriptor *cs_desc = + snd_usb_find_clock_source(chip->ctrl_intf, source_id); + + if (!cs_desc) + return false; + + return (fmt->nr_rates == 1 && + (fmt->clock & 0xff) == cs_desc->bClockID && + (cs_desc->bmAttributes & 0x3) != + UAC_CLOCK_SOURCE_TYPE_EXT); + } + + return false; +} + static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, - int protocol, + struct audioformat *fmt, int source_id) { int err; @@ -160,7 +186,7 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, struct usb_device *dev = chip->dev; u32 bmControls; - if (protocol == UAC_VERSION_3) { + if (fmt->protocol == UAC_VERSION_3) { struct uac3_clock_source_descriptor *cs_desc = snd_usb_find_clock_source_v3(chip->ctrl_intf, source_id); @@ -194,10 +220,14 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, return false; } - return data ? true : false; + if (data) + return true; + else + return uac_clock_source_is_valid_quirk(chip, fmt, source_id); } -static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, +static int __uac_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, int entity_id, unsigned long *visited, bool validate) { struct uac_clock_source_descriptor *source; @@ -217,7 +247,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, source = snd_usb_find_clock_source(chip->ctrl_intf, entity_id); if (source) { entity_id = source->bClockID; - if (validate && !uac_clock_source_is_valid(chip, UAC_VERSION_2, + if (validate && !uac_clock_source_is_valid(chip, fmt, entity_id)) { usb_audio_err(chip, "clock source %d is not valid, cannot use\n", @@ -248,8 +278,9 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, } cur = ret; - ret = __uac_clock_find_source(chip, selector->baCSourceID[ret - 1], - visited, validate); + ret = __uac_clock_find_source(chip, fmt, + selector->baCSourceID[ret - 1], + visited, validate); if (!validate || ret > 0 || !chip->autoclock) return ret; @@ -260,8 +291,9 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, if (i == cur) continue; - ret = __uac_clock_find_source(chip, selector->baCSourceID[i - 1], - visited, true); + ret = __uac_clock_find_source(chip, fmt, + selector->baCSourceID[i - 1], + visited, true); if (ret < 0) continue; @@ -281,14 +313,16 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, /* FIXME: multipliers only act as pass-thru element for now */ multiplier = snd_usb_find_clock_multiplier(chip->ctrl_intf, entity_id); if (multiplier) - return __uac_clock_find_source(chip, multiplier->bCSourceID, - visited, validate); + return __uac_clock_find_source(chip, fmt, + multiplier->bCSourceID, + visited, validate); return -EINVAL; } -static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, - unsigned long *visited, bool validate) +static int __uac3_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, int entity_id, + unsigned long *visited, bool validate) { struct uac3_clock_source_descriptor *source; struct uac3_clock_selector_descriptor *selector; @@ -307,7 +341,7 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, source = snd_usb_find_clock_source_v3(chip->ctrl_intf, entity_id); if (source) { entity_id = source->bClockID; - if (validate && !uac_clock_source_is_valid(chip, UAC_VERSION_3, + if (validate && !uac_clock_source_is_valid(chip, fmt, entity_id)) { usb_audio_err(chip, "clock source %d is not valid, cannot use\n", @@ -338,7 +372,8 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, } cur = ret; - ret = __uac3_clock_find_source(chip, selector->baCSourceID[ret - 1], + ret = __uac3_clock_find_source(chip, fmt, + selector->baCSourceID[ret - 1], visited, validate); if (!validate || ret > 0 || !chip->autoclock) return ret; @@ -350,8 +385,9 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, if (i == cur) continue; - ret = __uac3_clock_find_source(chip, selector->baCSourceID[i - 1], - visited, true); + ret = __uac3_clock_find_source(chip, fmt, + selector->baCSourceID[i - 1], + visited, true); if (ret < 0) continue; @@ -372,7 +408,8 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, multiplier = snd_usb_find_clock_multiplier_v3(chip->ctrl_intf, entity_id); if (multiplier) - return __uac3_clock_find_source(chip, multiplier->bCSourceID, + return __uac3_clock_find_source(chip, fmt, + multiplier->bCSourceID, visited, validate); return -EINVAL; @@ -389,18 +426,18 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, * * Returns the clock source UnitID (>=0) on success, or an error. */ -int snd_usb_clock_find_source(struct snd_usb_audio *chip, int protocol, - int entity_id, bool validate) +int snd_usb_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, bool validate) { DECLARE_BITMAP(visited, 256); memset(visited, 0, sizeof(visited)); - switch (protocol) { + switch (fmt->protocol) { case UAC_VERSION_2: - return __uac_clock_find_source(chip, entity_id, visited, + return __uac_clock_find_source(chip, fmt, fmt->clock, visited, validate); case UAC_VERSION_3: - return __uac3_clock_find_source(chip, entity_id, visited, + return __uac3_clock_find_source(chip, fmt, fmt->clock, visited, validate); default: return -EINVAL; @@ -501,8 +538,7 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, * automatic clock selection if the current clock is not * valid. */ - clock = snd_usb_clock_find_source(chip, fmt->protocol, - fmt->clock, true); + clock = snd_usb_clock_find_source(chip, fmt, true); if (clock < 0) { /* We did not find a valid clock, but that might be * because the current sample rate does not match an @@ -510,8 +546,7 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, * and we will do another validation after setting the * rate. */ - clock = snd_usb_clock_find_source(chip, fmt->protocol, - fmt->clock, false); + clock = snd_usb_clock_find_source(chip, fmt, false); if (clock < 0) return clock; } @@ -577,7 +612,7 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, validation: /* validate clock after rate change */ - if (!uac_clock_source_is_valid(chip, fmt->protocol, clock)) + if (!uac_clock_source_is_valid(chip, fmt, clock)) return -ENXIO; return 0; } diff --git a/sound/usb/clock.h b/sound/usb/clock.h index 076e31b79ee0..68df0fbe09d0 100644 --- a/sound/usb/clock.h +++ b/sound/usb/clock.h @@ -6,7 +6,7 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt, int rate); -int snd_usb_clock_find_source(struct snd_usb_audio *chip, int protocol, - int entity_id, bool validate); +int snd_usb_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, bool validate); #endif /* __USBAUDIO_CLOCK_H */ diff --git a/sound/usb/format.c b/sound/usb/format.c index 50cb183958bf..9f5cb4ed3a0c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -336,8 +336,7 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, struct usb_device *dev = chip->dev; unsigned char tmp[2], *data; int nr_triplets, data_size, ret = 0, ret_l6; - int clock = snd_usb_clock_find_source(chip, fp->protocol, - fp->clock, false); + int clock = snd_usb_clock_find_source(chip, fp, false); if (clock < 0) { dev_err(&dev->dev, -- cgit v1.2.3 From 0fbb027b44e79700da80e4b8bd1c1914d4796af6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Feb 2020 07:03:49 +0100 Subject: ALSA: pcm: Fix double hw_free calls The commit 66f2d19f8116 ("ALSA: pcm: Fix memory leak at closing a stream without hw_free") tried to fix the regression wrt the missing hw_free call at closing without SNDRV_PCM_IOCTL_HW_FREE ioctl. However, the code change dropped mistakenly the state check, resulting in calling hw_free twice when SNDRV_PCM_IOCTL_HW_FRE got called beforehand. For most drivers, this is almost harmless, but the drivers like SOF show another regression now. This patch adds the state condition check before calling do_hw_free() at releasing the stream for avoiding the double hw_free calls. Fixes: 66f2d19f8116 ("ALSA: pcm: Fix memory leak at closing a stream without hw_free") Reported-by: Bard Liao Reported-by: Pierre-Louis Bossart Tested-by: Pierre-Louis Bossart Cc: Link: https://lore.kernel.org/r/s5hd0ajyprg.wl-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 336406bcb59e..d5443eeb8b63 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2594,7 +2594,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) snd_pcm_drop(substream); if (substream->hw_opened) { - do_hw_free(substream); + if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + do_hw_free(substream); substream->ops->close(substream); substream->hw_opened = 0; } -- cgit v1.2.3 From bb51e669fa49feb5904f452b2991b240ef31bc97 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 14 Feb 2020 12:13:14 +0100 Subject: ALSA: seq: Avoid concurrent access to queue flags The queue flags are represented in bit fields and the concurrent access may result in unexpected results. Although the current code should be mostly OK as it's only reading a field while writing other fields as KCSAN reported, it's safer to cover both with a proper spinlock protection. This patch fixes the possible concurrent read by protecting with q->owner_lock. Also the queue owner field is protected as well since it's the field to be protected by the lock itself. Reported-by: syzbot+65c6c92d04304d0a8efc@syzkaller.appspotmail.com Reported-by: syzbot+e60ddfa48717579799dd@syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20200214111316.26939-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_queue.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index caf68bf42f13..20c552cf8398 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -392,6 +392,7 @@ int snd_seq_queue_check_access(int queueid, int client) int snd_seq_queue_set_owner(int queueid, int client, int locked) { struct snd_seq_queue *q = queueptr(queueid); + unsigned long flags; if (q == NULL) return -EINVAL; @@ -401,8 +402,10 @@ int snd_seq_queue_set_owner(int queueid, int client, int locked) return -EPERM; } + spin_lock_irqsave(&q->owner_lock, flags); q->locked = locked ? 1 : 0; q->owner = client; + spin_unlock_irqrestore(&q->owner_lock, flags); queue_access_unlock(q); queuefree(q); @@ -539,15 +542,17 @@ void snd_seq_queue_client_termination(int client) unsigned long flags; int i; struct snd_seq_queue *q; + bool matched; for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) { if ((q = queueptr(i)) == NULL) continue; spin_lock_irqsave(&q->owner_lock, flags); - if (q->owner == client) + matched = (q->owner == client); + if (matched) q->klocked = 1; spin_unlock_irqrestore(&q->owner_lock, flags); - if (q->owner == client) { + if (matched) { if (q->timer->running) snd_seq_timer_stop(q->timer); snd_seq_timer_reset(q->timer); @@ -739,6 +744,8 @@ void snd_seq_info_queues_read(struct snd_info_entry *entry, int i, bpm; struct snd_seq_queue *q; struct snd_seq_timer *tmr; + bool locked; + int owner; for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) { if ((q = queueptr(i)) == NULL) @@ -750,9 +757,14 @@ void snd_seq_info_queues_read(struct snd_info_entry *entry, else bpm = 0; + spin_lock_irq(&q->owner_lock); + locked = q->locked; + owner = q->owner; + spin_unlock_irq(&q->owner_lock); + snd_iprintf(buffer, "queue %d: [%s]\n", q->queue, q->name); - snd_iprintf(buffer, "owned by client : %d\n", q->owner); - snd_iprintf(buffer, "lock status : %s\n", q->locked ? "Locked" : "Free"); + snd_iprintf(buffer, "owned by client : %d\n", owner); + snd_iprintf(buffer, "lock status : %s\n", locked ? "Locked" : "Free"); snd_iprintf(buffer, "queued time events : %d\n", snd_seq_prioq_avail(q->timeq)); snd_iprintf(buffer, "queued tick events : %d\n", snd_seq_prioq_avail(q->tickq)); snd_iprintf(buffer, "timer state : %s\n", tmr->running ? "Running" : "Stopped"); -- cgit v1.2.3 From dc7497795e014d84699c3b8809ed6df35352dd74 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 14 Feb 2020 12:13:15 +0100 Subject: ALSA: seq: Fix concurrent access to queue current tick/time snd_seq_check_queue() passes the current tick and time of the given queue as a pointer to snd_seq_prioq_cell_out(), but those might be updated concurrently by the seq timer update. Fix it by retrieving the current tick and time via the proper helper functions at first, and pass those values to snd_seq_prioq_cell_out() later in the loops. snd_seq_timer_get_cur_time() takes a new argument and adjusts with the current system time only when it's requested so; this update isn't needed for snd_seq_check_queue(), as it's called either from the interrupt handler or right after queuing. Also, snd_seq_timer_get_cur_tick() is changed to read the value in the spinlock for the concurrency, too. Reported-by: syzbot+fd5e0eaa1a32999173b2@syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20200214111316.26939-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 4 ++-- sound/core/seq/seq_queue.c | 9 ++++++--- sound/core/seq/seq_timer.c | 13 ++++++++++--- sound/core/seq/seq_timer.h | 3 ++- 4 files changed, 20 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 6d9592f0ae1d..cc93157fa950 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -580,7 +580,7 @@ static int update_timestamp_of_queue(struct snd_seq_event *event, event->queue = queue; event->flags &= ~SNDRV_SEQ_TIME_STAMP_MASK; if (real_time) { - event->time.time = snd_seq_timer_get_cur_time(q->timer); + event->time.time = snd_seq_timer_get_cur_time(q->timer, true); event->flags |= SNDRV_SEQ_TIME_STAMP_REAL; } else { event->time.tick = snd_seq_timer_get_cur_tick(q->timer); @@ -1659,7 +1659,7 @@ static int snd_seq_ioctl_get_queue_status(struct snd_seq_client *client, tmr = queue->timer; status->events = queue->tickq->cells + queue->timeq->cells; - status->time = snd_seq_timer_get_cur_time(tmr); + status->time = snd_seq_timer_get_cur_time(tmr, true); status->tick = snd_seq_timer_get_cur_tick(tmr); status->running = tmr->running; diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index 20c552cf8398..71a6ea62c3be 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -238,6 +238,8 @@ 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; if (q == NULL) return; @@ -254,17 +256,18 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) __again: /* Process tick queue... */ + cur_tick = snd_seq_timer_get_cur_tick(q->timer); for (;;) { - cell = snd_seq_prioq_cell_out(q->tickq, - &q->timer->tick.cur_tick); + cell = snd_seq_prioq_cell_out(q->tickq, &cur_tick); if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); } /* Process time queue... */ + cur_time = snd_seq_timer_get_cur_time(q->timer, false); for (;;) { - cell = snd_seq_prioq_cell_out(q->timeq, &q->timer->cur_time); + cell = snd_seq_prioq_cell_out(q->timeq, &cur_time); if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index be59b59c9be4..1645e4142e30 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -428,14 +428,15 @@ int snd_seq_timer_continue(struct snd_seq_timer *tmr) } /* return current 'real' time. use timeofday() to get better granularity. */ -snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr) +snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr, + bool adjust_ktime) { snd_seq_real_time_t cur_time; unsigned long flags; spin_lock_irqsave(&tmr->lock, flags); cur_time = tmr->cur_time; - if (tmr->running) { + if (adjust_ktime && tmr->running) { struct timespec64 tm; ktime_get_ts64(&tm); @@ -452,7 +453,13 @@ snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr) high PPQ values) */ snd_seq_tick_time_t snd_seq_timer_get_cur_tick(struct snd_seq_timer *tmr) { - return tmr->tick.cur_tick; + snd_seq_tick_time_t cur_tick; + unsigned long flags; + + spin_lock_irqsave(&tmr->lock, flags); + cur_tick = tmr->tick.cur_tick; + spin_unlock_irqrestore(&tmr->lock, flags); + return cur_tick; } diff --git a/sound/core/seq/seq_timer.h b/sound/core/seq/seq_timer.h index 66c3e344eae3..4bec57df8158 100644 --- a/sound/core/seq/seq_timer.h +++ b/sound/core/seq/seq_timer.h @@ -120,7 +120,8 @@ int snd_seq_timer_set_tempo_ppq(struct snd_seq_timer *tmr, int tempo, int ppq); int snd_seq_timer_set_position_tick(struct snd_seq_timer *tmr, snd_seq_tick_time_t position); int snd_seq_timer_set_position_time(struct snd_seq_timer *tmr, snd_seq_real_time_t position); int snd_seq_timer_set_skew(struct snd_seq_timer *tmr, unsigned int skew, unsigned int base); -snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr); +snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr, + bool adjust_ktime); snd_seq_tick_time_t snd_seq_timer_get_cur_tick(struct snd_seq_timer *tmr); extern int seq_default_timer_class; -- cgit v1.2.3 From a655e2b107d463ce2745188ce050d07daed09a71 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 17 Feb 2020 16:19:47 +0100 Subject: ALSA: hda/realtek - Apply quirk for MSI GP63, too The same quirk that was applied to MSI GL73 is needed for MSI GP63, too. Adding the entry with the SSID 1462:1228. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206503 Cc: Link: https://lore.kernel.org/r/20200217151947.17528-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6c8cb4ce517e..82485e06dde1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2447,6 +2447,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_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(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), -- cgit v1.2.3 From cc5049ae4d457194796f854eb2e38b9727ad8c2d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2020 09:09:15 +0100 Subject: ALSA: hda/realtek - Apply quirk for yet another MSI laptop MSI GP65 laptop with SSID 1462:1293 requires the same quirk as other MSI models. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204159 Cc: Link: https://lore.kernel.org/r/20200218080915.3433-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 82485e06dde1..477589e7ec1d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2449,6 +2449,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), -- cgit v1.2.3 From 44eeb081b8630bb3ad3cd381d1ae1831463e48bb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2020 10:14:09 +0100 Subject: ALSA: hda: Use scnprintf() for printing texts for sysfs/procfs Some code in HD-audio driver calls snprintf() in a loop and still expects that the return value were actually written size, while snprintf() returns the expected would-be length instead. When the given buffer limit were small, this leads to a buffer overflow. Use scnprintf() for addressing those issues. It returns the actually written size unlike snprintf(). Cc: Link: https://lore.kernel.org/r/20200218091409.27162-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/hda/hdmi_chmap.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_eld.c | 2 +- sound/pci/hda/hda_sysfs.c | 4 ++-- 4 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c index 5fd6d575e123..aad5c4bf4d34 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -250,7 +250,7 @@ void snd_hdac_print_channel_allocation(int spk_alloc, char *buf, int buflen) for (i = 0, j = 0; i < ARRAY_SIZE(cea_speaker_allocation_names); i++) { if (spk_alloc & (1 << i)) - j += snprintf(buf + j, buflen - j, " %s", + j += scnprintf(buf + j, buflen - j, " %s", cea_speaker_allocation_names[i]); } buf[j] = '\0'; /* necessary when j == 0 */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5dc42f932739..53e7732ef752 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4022,7 +4022,7 @@ void snd_print_pcm_bits(int pcm, char *buf, int buflen) for (i = 0, j = 0; i < ARRAY_SIZE(bits); i++) if (pcm & (AC_SUPPCM_BITS_8 << i)) - j += snprintf(buf + j, buflen - j, " %d", bits[i]); + j += scnprintf(buf + j, buflen - j, " %d", bits[i]); buf[j] = '\0'; /* necessary when j == 0 */ } diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index bb46c89b7f63..136477ed46ae 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -360,7 +360,7 @@ static void hdmi_print_pcm_rates(int pcm, char *buf, int buflen) for (i = 0, j = 0; i < ARRAY_SIZE(alsa_rates); i++) if (pcm & (1 << i)) - j += snprintf(buf + j, buflen - j, " %d", + j += scnprintf(buf + j, buflen - j, " %d", alsa_rates[i]); buf[j] = '\0'; /* necessary when j == 0 */ diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 0607ed5d1959..eb8ec109d7ad 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -222,7 +222,7 @@ static ssize_t init_verbs_show(struct device *dev, int i, len = 0; mutex_lock(&codec->user_mutex); snd_array_for_each(&codec->init_verbs, i, v) { - len += snprintf(buf + len, PAGE_SIZE - len, + len += scnprintf(buf + len, PAGE_SIZE - len, "0x%02x 0x%03x 0x%04x\n", v->nid, v->verb, v->param); } @@ -272,7 +272,7 @@ static ssize_t hints_show(struct device *dev, int i, len = 0; mutex_lock(&codec->user_mutex); snd_array_for_each(&codec->hints, i, hint) { - len += snprintf(buf + len, PAGE_SIZE - len, + len += scnprintf(buf + len, PAGE_SIZE - len, "%s = %s\n", hint->key, hint->val); } mutex_unlock(&codec->user_mutex); -- cgit v1.2.3 From c37c0ab029569a75fd180edb03d411e7a28a936f Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Wed, 19 Feb 2020 13:23:06 +0800 Subject: ALSA: hda/realtek - Fix a regression for mute led on Lenovo Carbon X1 Need to chain the THINKPAD_ACPI, otherwise the mute led will not work. Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") Cc: Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20200219052306.24935-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 477589e7ec1d..31bee0512334 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6684,6 +6684,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC285_FIXUP_SPEAKER2_TO_DAC1] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_speaker2_to_dac1, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, -- cgit v1.2.3 From 3fb83cbee1de58fcd5d22f1db89460bb7c08b6e8 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 4 Mar 2020 22:02:41 +0800 Subject: ASoC: wm8741: Fix typo in Kconfig prompt Fix trivial copy-n-paste mistake. Signed-off-by: Axel Lin Link: https://lore.kernel.org/r/20200304140241.340-1-axel.lin@ingics.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7e90f5d83097..ea912439e446 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1406,7 +1406,7 @@ config SND_SOC_WM8737 depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8741 - tristate "Wolfson Microelectronics WM8737 DAC" + tristate "Wolfson Microelectronics WM8741 DAC" depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8750 -- cgit v1.2.3 From 78def224f59c05d00e815be946ec229719ccf377 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 20 Feb 2020 15:21:54 +0800 Subject: ALSA: hda/realtek - Add Headset Mic supported Dell desktop platform supported headset Mic. Add pin verb to enable headset Mic. This platform only support fixed type headset for Iphone type. Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/r/b9da28d772ef43088791b0f3675929e7@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 31bee0512334..128c7e9df6de 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7117,6 +7117,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0935, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x097e, "Dell Precision", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x097d, "Dell Precision", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x098d, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x09bf, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- cgit v1.2.3 From 76f7dec08fd64e9e3ad0810a1a8a60b0a846d348 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Mon, 10 Feb 2020 16:30:26 +0800 Subject: ALSA: hda/realtek - Add Headset Button supported for ThinkPad X1 ThinkPad want to support Headset Button control. This patch will enable it. Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/r/7f0b7128f40f41f6b5582ff610adc33d@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 128c7e9df6de..a8d66c1e4991 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5920,7 +5920,7 @@ enum { ALC289_FIXUP_DUAL_SPK, ALC294_FIXUP_SPK2_TO_DAC1, ALC294_FIXUP_ASUS_DUAL_SPK, - + ALC285_FIXUP_THINKPAD_HEADSET_JACK, }; static const struct hda_fixup alc269_fixups[] = { @@ -7042,7 +7042,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_SPK2_TO_DAC1 }, - + [ALC285_FIXUP_THINKPAD_HEADSET_JACK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_jack, + .chained = true, + .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1 + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7278,8 +7283,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_SPEAKER2_TO_DAC1), - SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_SPEAKER2_TO_DAC1), + SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), + SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), -- cgit v1.2.3 From 0d45e86d2267d5bdf7bbb631499788da1c27ceb2 Mon Sep 17 00:00:00 2001 From: Christian Lachner Date: Sun, 23 Feb 2020 10:24:16 +0100 Subject: ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Master The Gigabyte X570 Aorus Master motherboard with ALC1220 codec requires a similar workaround for Clevo laptops to enforce the DAC/mixer connection path. Set up a quirk entry for that. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205275 Signed-off-by: Christian Lachner Cc: Link: https://lore.kernel.org/r/20200223092416.15016-2-gladiac@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a8d66c1e4991..508d2be3d43d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2447,6 +2447,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_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(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), -- cgit v1.2.3 From 8b33a134a9cc2a501f8fc731d91caef39237d495 Mon Sep 17 00:00:00 2001 From: Jian-Hong Pan Date: Tue, 25 Feb 2020 15:29:21 +0800 Subject: ALSA: hda/realtek - Enable the headset of ASUS B9450FA with ALC294 A headset on the laptop like ASUS B9450FA does not work, until quirk ALC294_FIXUP_ASUS_HPE is applied. Signed-off-by: Jian-Hong Pan Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/r/20200225072920.109199-1-jian-hong@endlessm.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 508d2be3d43d..0ac06ff1a17c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5922,6 +5922,7 @@ enum { ALC294_FIXUP_SPK2_TO_DAC1, ALC294_FIXUP_ASUS_DUAL_SPK, ALC285_FIXUP_THINKPAD_HEADSET_JACK, + ALC294_FIXUP_ASUS_HPE, }; static const struct hda_fixup alc269_fixups[] = { @@ -7049,6 +7050,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1 }, + [ALC294_FIXUP_ASUS_HPE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Set EAPD high */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x0f }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x7774 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7214,6 +7226,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), -- cgit v1.2.3 From 95dbf14b236f3147f716cd159bd29461916c610e Mon Sep 17 00:00:00 2001 From: Thomas Bogendoerfer Date: Fri, 6 Mar 2020 11:58:37 +0100 Subject: ALSA: sgio2audio: Remove usage of dropped hw_params/hw_free functions Commit ee88f4ebe575 ("ALSA: mips: Use managed buffer allocation") removed superfluous hw_params/hw_free callbacks, but forgot to remove them where they were used. Fixes: ee88f4ebe575 ("ALSA: mips: Use managed buffer allocation") Signed-off-by: Thomas Bogendoerfer Link: https://lore.kernel.org/r/20200306105837.31523-1-tsbogend@alpha.franken.de Signed-off-by: Takashi Iwai --- sound/mips/sgio2audio.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 9f60a5037f8b..5bf1ea150f26 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -649,8 +649,6 @@ snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream) static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .open = snd_sgio2audio_playback1_open, .close = snd_sgio2audio_pcm_close, - .hw_params = snd_sgio2audio_pcm_hw_params, - .hw_free = snd_sgio2audio_pcm_hw_free, .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, @@ -659,8 +657,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = { static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = { .open = snd_sgio2audio_playback2_open, .close = snd_sgio2audio_pcm_close, - .hw_params = snd_sgio2audio_pcm_hw_params, - .hw_free = snd_sgio2audio_pcm_hw_free, .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, @@ -669,8 +665,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = { static const struct snd_pcm_ops snd_sgio2audio_capture_ops = { .open = snd_sgio2audio_capture_open, .close = snd_sgio2audio_pcm_close, - .hw_params = snd_sgio2audio_pcm_hw_params, - .hw_free = snd_sgio2audio_pcm_hw_free, .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, -- cgit v1.2.3