March 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v4 1/3] ASoC: tas571x: added missing register literals
by Petr Kulhavy 31 Mar '16

31 Mar '16

The list of TAS571x registers was incomplete. Added the missing register definitions up to the register 0x25. Added volatile and read-only register tables into tas5711_regmap_config and tas5717_regmap_config. The chip has 256 registers in total. But from address 0x29 on (0x26 to 0x28 are reserved) the register width varies between 20, 12 and 8 bytes, which the register map cannot represent. Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- v1: <initial> v2: <no change> v3: added volatile and read-only register tables into regmap configs v4: no change, the set v3 resubmitted without the already accepted patches sound/soc/codecs/tas571x.c | 28 ++++++++++++++++++++++++++++ sound/soc/codecs/tas571x.h | 22 ++++++++++++++++++++++ 2 files changed, 50 insertions(+) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index aafee9bbe01a..ef6c8d9b251a 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -57,6 +57,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_CH1_VOL_REG: case TAS571X_CH2_VOL_REG: return priv->chip->vol_reg_size; + case TAS571X_INPUT_MUX_REG: + case TAS571X_CH4_SRC_SELECT_REG: + case TAS571X_PWM_MUX_REG: + return 4; default: return 1; } @@ -259,6 +263,26 @@ static const struct snd_kcontrol_new tas5711_controls[] = { 1, 1), }; +static const struct regmap_range tas571x_readonly_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_DEV_ID_REG), +}; + +static const struct regmap_range tas571x_volatile_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG), + regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG), +}; + +static const struct regmap_access_table tas571x_write_regs = { + .no_ranges = tas571x_readonly_regs_range, + .n_no_ranges = ARRAY_SIZE(tas571x_readonly_regs_range), +}; + +static const struct regmap_access_table tas571x_volatile_regs = { + .yes_ranges = tas571x_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(tas571x_volatile_regs_range), + +}; + static const struct reg_default tas5711_reg_defaults[] = { { 0x04, 0x05 }, { 0x05, 0x40 }, @@ -278,6 +302,8 @@ static const struct regmap_config tas5711_regmap_config = { .reg_defaults = tas5711_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5711_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; static const struct tas571x_chip tas5711_chip = { @@ -332,6 +358,8 @@ static const struct regmap_config tas5717_regmap_config = { .reg_defaults = tas5717_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5717_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; /* This entry is reused for tas5719 as the software interface is identical. */ diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 0aee471232cd..cf800c364f0f 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -13,6 +13,10 @@ #define _TAS571X_H /* device registers */ +#define TAS571X_CLK_CTRL_REG 0x00 +#define TAS571X_DEV_ID_REG 0x01 +#define TAS571X_ERR_STATUS_REG 0x02 +#define TAS571X_SYS_CTRL_1_REG 0x03 #define TAS571X_SDI_REG 0x04 #define TAS571X_SDI_FMT_MASK 0x0f @@ -27,7 +31,25 @@ #define TAS571X_MVOL_REG 0x07 #define TAS571X_CH1_VOL_REG 0x08 #define TAS571X_CH2_VOL_REG 0x09 +#define TAS571X_CH3_VOL_REG 0x0a +#define TAS571X_VOL_CFG_REG 0x0e +#define TAS571X_MODULATION_LIMIT_REG 0x10 +#define TAS571X_IC_DELAY_CH1_REG 0x11 +#define TAS571X_IC_DELAY_CH2_REG 0x12 +#define TAS571X_IC_DELAY_CH3_REG 0x13 +#define TAS571X_IC_DELAY_CH4_REG 0x14 +#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */ +#define TAS571X_PWM_CH1_SDN_MASK (1<<0) +#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1) +#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2) +#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3) + +#define TAS571X_START_STOP_PERIOD_REG 0x1a #define TAS571X_OSC_TRIM_REG 0x1b +#define TAS571X_BKND_ERR_REG 0x1c +#define TAS571X_INPUT_MUX_REG 0x20 +#define TAS571X_CH4_SRC_SELECT_REG 0x21 +#define TAS571X_PWM_MUX_REG 0x25 #endif /* _TAS571X_H */ -- 1.9.1

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[alsa-devel] [PATCH] ASoC: nau8825: assign DAC Ch to match headset L/R
by John Hsu 31 Mar '16

31 Mar '16

The default value of DAC channel select is reverse in codec. For normal usage, switch the channel select when codec bootup. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 5 +++++ sound/soc/codecs/nau8825.h | 6 ++++++ 2 files changed, 11 insertions(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 3e7bee2..1a99a25 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -905,6 +905,11 @@ static void nau8825_init_regs(struct nau8825 *nau8825) NAU8825_RDAC_CLK_DELAY_MASK | NAU8825_RDAC_VREF_MASK, (0x2 << NAU8825_RDAC_CLK_DELAY_SFT) | (0x3 << NAU8825_RDAC_VREF_SFT)); + /* Config L/R channel */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_MASK, NAU8825_DACL_CH_SEL_L); + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACR_CTRL, + NAU8825_DACL_CH_SEL_MASK, NAU8825_DACL_CH_SEL_R); } static const struct regmap_config nau8825_regmap_config = { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 5fe009d..fcf0d0a 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -256,9 +256,15 @@ /* DACL_CTRL (0x33) */ #define NAU8825_DACL_CH_SEL_SFT 9 +#define NAU8825_DACL_CH_SEL_MASK (0x1 << NAU8825_DACL_CH_SEL_SFT) +#define NAU8825_DACL_CH_SEL_L (0x0 << NAU8825_DACL_CH_SEL_SFT) +#define NAU8825_DACL_CH_SEL_R (0x1 << NAU8825_DACL_CH_SEL_SFT) /* DACR_CTRL (0x34) */ #define NAU8825_DACR_CH_SEL_SFT 9 +#define NAU8825_DACR_CH_SEL_MASK (0x1 << NAU8825_DACR_CH_SEL_SFT) +#define NAU8825_DACR_CH_SEL_L (0x0 << NAU8825_DACR_CH_SEL_SFT) +#define NAU8825_DACR_CH_SEL_R (0x1 << NAU8825_DACR_CH_SEL_SFT) /* CLASSG_CTRL (0x50) */ #define NAU8825_CLASSG_TIMER_SFT 8 -- 2.6.4

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix double-free in snd_usb_add_audio_stream()
by Vladis Dronov 31 Mar '16

31 Mar '16

Hello, Takashi, Jaroslav, all, Please, see the research and the following patch on a double-free bug in [snd-usb-audio]. 1) The upstream commits 0f886ca1, 902eb7fd and 447d6275f (many thanks to Takashi Iwai) revealed that there is a double-free bug in [snd-usb-audio] module due to alloc/free logic flaw in snd_usb_add_audio_stream() function. A double-free leads to kernel structure/list/slab corruptions and shortly to null-deref, GPF or lockup. 2.1) Let me describe what happens with the current code in usb_audio_probe(), create_fixed_stream_quirk() and snd_usb_add_audio_stream(): > [ sound/usb/card.c ] > static int usb_audio_probe(struct usb_interface *intf, > const struct usb_device_id *usb_id) > { > ... > if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { > /* need some special handlings */ > err = snd_usb_create_quirk(chip, intf, &usb_audio_driver, quirk); > if (err < 0) > goto __error; > } > ... > __error: > if (chip) { > if (!chip->num_interfaces) > snd_card_free(chip->card); Somewhere in the middle of usb_audio_probe() the function snd_usb_create_quirk() is called, and if it returns with an error and no interfaces were created, the sound card is destroyed with "snd_card_free(chip->card)". 2.2) create_fixed_stream_quirk() is called from snd_usb_create_quirk(): > [ sound/usb/quirks.c ] > static int create_fixed_stream_quirk(struct snd_usb_audio *chip, > struct usb_interface *iface, > struct usb_driver *driver, > const struct snd_usb_audio_quirk *quirk) > { > struct audioformat *fp; > struct usb_host_interface *alts; > struct usb_interface_descriptor *altsd; > ... > fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL); > ... > (*) stream = (fp->endpoint & USB_DIR_IN) > (*) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; > (*) err = snd_usb_add_audio_stream(chip, stream, fp); > (*) if (err < 0) > (*) goto error; > > if (fp->iface != get_iface_desc(&iface->altsetting[0])->bInterfaceNumber || > fp->altset_idx >= iface->num_altsetting) { > err = -EINVAL; > goto error; > } > alts = &iface->altsetting[fp->altset_idx]; > altsd = get_iface_desc(alts); > if (altsd->bNumEndpoints < 1) { > err = -EINVAL; > goto error; > } > ... > error: > kfree(fp); > kfree(rate_table); > return err; > } 2.3) *fp is allocated and passed to snd_usb_add_audio_stream() where snd_usb_init_substream() is called: > [ sound/usb/stream.c ] > int snd_usb_add_audio_stream(struct snd_usb_audio *chip, > int stream, > struct audioformat *fp) > { > struct snd_usb_stream *as; > struct snd_usb_substream *subs; > struct snd_pcm *pcm; > ... > /* create a new pcm */ > as = kzalloc(sizeof(*as), GFP_KERNEL); > ... > snd_usb_init_substream(as, stream, fp); 2.4) In turn snd_usb_init_substream() adds audioformat *fp by its &fp->list to substream's fmt_list list: > [ sound/usb/stream.c ] > static void snd_usb_init_substream(struct snd_usb_stream *as, > int stream, > struct audioformat *fp) > { > struct snd_usb_substream *subs = &as->substream[stream]; > > INIT_LIST_HEAD(&subs->fmt_list); > ... > list_add_tail(&fp->list, &subs->fmt_list); > subs->num_formats++; 2.5) Things go bad from this point in case snd_usb_add_audio_stream() or the caller go the error path. The bug is that the caller frees (see "error: kfree(fp);" code) *fp on the error path, BUT the pointer to the already-freed memory remains in the substream's fmt_list list. The double-free happens after create_fixed_stream_quirk() returns with an error and usb_audio_probe() calls snd_card_free()->...->snd_usb_audio_pcm_free()->free_substream(). As subs->fmt_list is already corrupted, iterating it with list_for_each_entry_safe() leads to any and unpredictable results. > static void free_substream(struct snd_usb_substream *subs) > { > struct audioformat *fp, *n; > ... > list_for_each_entry_safe(fp, n, &subs->fmt_list, list) { > ... > kfree(fp); 3.1) The crash is reproduceable by faking the USB device with no endpoints as described in https://bugzilla.redhat.com/show_bug.cgi?id=1283355 and https://bugzilla.redhat.com/show_bug.cgi?id=1283358. Please see as a proof the following kernel log with debug printing added to the code. First, *fp is added to fmt_list in snd_usb_init_substream(): [ 322.797223] usb 1-1: new full-speed USB device number 2 using xhci_hcd [ 322.974083] usb 1-1: config 1 interface 0 altsetting 0 has 3 endpoint descriptors, different from the interface descriptor's value: 0 [ 322.987031] usb 1-1: New USB device found, idVendor=045e, idProduct=0283 [ 322.998318] usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 323.083913] media: Linux media interface: v0.10 [ 323.231249] init_substream: add_tail() fp=ffff88003364ba80 fp->list.next=ffff8800b1e898b8 fp->list.prev=ffff8800b1e898b8 fmt_list=ffff8800b1e898b8 fmt_list.next=ffff88003364ba80 prev=ffff88003364ba80 num_formats=1 As you can see we have a correct list here with head at fmt_list=ffff8800b1e898b8 and single element fp=ffff88003364ba80. 3.2) The code finds out that there are too few endpoints present and goes the error path (to the "error:" label): [ 323.307927] usb 1-1: too few endpoints [ 323.312964] trace-before-free: substream-0 ffff8800b1e89818 numf 1 fmt_list ffff8800b1e898b8 next ffff88003364ba80 [ 323.353759] fp=ffff88003364ba80 next=ffff8800b1e898b8 prev=ffff8800b1e898b8 rate=(null) chmap=(null) [ 323.362118] struct sane As you can see the substream's fmt_list is sane at this point. 3.3) After "kfree(fp)" in the error path of create_fixed_stream_quirk() *fp is freed, BUT the pointer to the freed memory remains in fmt_list. After *fp is freed the list is corrupted and contains trash: [ 323.371752] KFREE(fp) ffff88003364ba80 [ 323.380383] trace-after-free: substream-0 ffff8800b1e89818 numf 1 fmt_list ffff8800b1e898b8 next ffff88003364ba80 [ 323.400003] fp=ffff88003364ba80 next=ffff88003364bf80 prev=ffff8800b1e898b8 rate=(null) chmap=(null) [ 323.406786] fp=ffff88003364bf80 next= (null) prev= (null) rate=(null) chmap=(null) [ 323.422211] next == NULL: FAIL, struct INSANE [ 323.436706] KFREE(rate_table) (null) 3.4) After error return from create_fixed_stream_quirk() the sound card is destroyed with "snd_card_free(chip->card)" in usb_audio_probe(). In the end free_substream() is called: [ 323.511256] usb 1-1: snd_usb_create_quirk() failed: -22 [ 323.565108] list_for_each_entry_safe(): fp=ffff88003364ba80 n=ffff88003364bf80 [ 323.586337] kfree fp ffff88003364ba80 <<< DOUBLE-FREE [ 323.588509] loop-end: fp=ffff88003364ba80 n=ffff88003364bf80 [ 323.599969] list_for_each_entry_safe(): fp=ffff88003364bf80 n=(null) [ 323.610460] kfree fp ffff88003364bf80 <<< FREEING SOMEONE ELSE'S MEMORY [ 323.613181] loop-end: fp=ffff88003364bf80 n=(null) <<< NULL-PTR DEREF ... [ 324.247113] BUG: unable to handle kernel NULL pointer dereference at (null) [ 324.247533] IP: [<ffffffffc02d40ef>] free_substream.part.0+0xef/0x100 [snd_usb_audio] [ 324.248088] PGD 0 [ 324.248088] Oops: 0000 [#1] SMP [ 324.248088] Modules linked in: snd_usb_audio(+) snd_usbmidi_lib snd_hwdep ... [ 324.248088] CPU: 2 PID: 767 Comm: systemd-udevd Not tainted 4.5.0-vladis #25 [ 324.248088] Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS rel-1.8.2-0-g33fbe13 by qemu-project.org 04/01/2014 [ 324.248088] task: ffff880034718000 ti: ffff8800b2fbc000 task.ti: ffff8800b2fbc000 [ 324.248088] RIP: 0010:[<ffffffffc02d40ef>] [<ffffffffc02d40ef>] free_substream.part.0+0xef/0x100 [snd_usb_audio] [ 324.248088] RSP: 0018:ffff8800b2fbf898 EFLAGS: 00010217 ... [ 324.248088] Call Trace: [ 324.248088] [<ffffffffc02d43fa>] snd_usb_audio_pcm_free+0x9a/0xa0 [snd_usb_audio] [ 324.248088] [<ffffffffc028d982>] snd_pcm_free+0x32/0xa0 [snd_pcm] [ 324.248088] [<ffffffffc028da02>] snd_pcm_dev_free+0x12/0x20 [snd_pcm] [ 324.248088] [<ffffffffc027d279>] __snd_device_free+0x29/0x70 [snd] [ 324.248088] [<ffffffffc027d660>] snd_device_free_all+0x30/0x60 [snd] [ 324.248088] [<ffffffffc02777a4>] release_card_device+0x34/0x90 [snd] [ 324.248088] [<ffffffff815ae2b2>] device_release+0x32/0x90 [ 324.248088] [<ffffffff81455f8a>] kobject_release+0x7a/0x190 [ 324.248088] [<ffffffff81455e47>] kobject_put+0x27/0x50 [ 324.248088] [<ffffffff815ae5f7>] put_device+0x17/0x20 [ 324.248088] [<ffffffffc0277b57>] snd_card_free+0x67/0x90 [snd] [ 324.248088] [<ffffffffc02c4f14>] usb_audio_probe+0x754/0x9d0 [snd_usb_audio] ... 4.1) The suggested patch consists of 2 changes. First, I suppose we should move the code in create_fixed_stream_quirk() marked as "(*)" (see above) after "if (altsd->bNumEndpoints < 1)" check. This way no allocations is done if we go an error path. I've checked that fp->iface and fp->altset_idx are not changed in snd_usb_add_audio_stream() and functions called from it, so it is safe to swap these 2 pieces of code. 4.2) The problem with double-free still remains. I've verified that all the callers of snd_usb_add_audio_stream() free *fp in case of error: $ git grep snd_usb_add_audio_stream sound/usb/quirks.c: err = snd_usb_add_audio_stream(chip, stream, fp); > * err = snd_usb_add_audio_stream(chip, stream, fp); > * if (err < 0) > * goto error; > * error: > * kfree(fp); > * kfree(rate_table); > * return err; sound/usb/quirks.c: err = snd_usb_add_audio_stream(chip, stream, fp); > * err = snd_usb_add_audio_stream(chip, stream, fp); > * if (err < 0) { > * kfree(fp); > * return err; > * } sound/usb/stream.c: err = snd_usb_add_audio_stream(chip, stream, fp); > * err = snd_usb_add_audio_stream(chip, stream, fp); > * if (err < 0) { > * kfree(fp->rate_table); > * kfree(fp->chmap); > * kfree(fp); > * return err; > * } This means that snd_usb_add_audio_stream() should remove *fp from the substream's fmt_list list on the error path, if it was already added. Such places are: > int snd_usb_add_audio_stream(struct snd_usb_audio *chip, int stream, struct audioformat *fp) > { > struct snd_usb_stream *as; > struct snd_usb_substream *subs; > struct snd_pcm *pcm; > int err; > > list_for_each_entry(as, &chip->pcm_list, list) { > subs = &as->substream[stream]; > ... > list_add_tail(&fp->list, &subs->fmt_list); <<< ADDING fp HERE > subs->num_formats++; > return 0; <<< NO ERROR PATH HERE > } > } > > /* look for an empty stream */ > list_for_each_entry(as, &chip->pcm_list, list) { > subs = &as->substream[stream]; > ... > snd_usb_init_substream(as, stream, fp); <<< ADDING fp HERE, IF add_chmap() FAILS > return add_chmap(as->pcm, stream, subs); <<< fp SHOULD BE REMOVED FROM fmt_list > } > > /* create a new pcm */ > as = kzalloc(sizeof(*as), GFP_KERNEL); > err = snd_pcm_new(chip->card, "USB Audio", chip->pcm_devs, > ... > if (err < 0) { <<< fp IS NOT ADDED YET HERE, SO FINE > kfree(as); > return err; > } > ... > snd_usb_init_substream(as, stream, fp); <<< ADDING fp HERE > ... <<< IF add_chmap() FAILS fp SHOULD > return add_chmap(pcm, stream, &as->substream[stream]); <<< BE REMOVED FROM fmt_list > } add_chmap() itself does not add anything to fmt_list list, so we indeed need to remove only the single list element from the list. Having all the above in mind, the patch follows. 4.3) How to handle possible error paths after successful call to snd_usb_add_audio_stream() in create_fixed_stream_quirk() is d iscussable. Properly it should be like the below, but I believe it is overcomplication here would and stick to a simple error_after_add_audio_stream: label: > error2: > snd_usb_del_audio_stream(...something...); > error: > kfree(fp); > kfree(rate_table); > return err; Best regards, Vladis Dronov | Red Hat, Inc. | Product Security Engineer

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[alsa-devel] [PATCH V3 0/2] ASoC: share more product information with user space
by han.lu＠intel.com 31 Mar '16

31 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Share more product information, so user space utils such as PA and UCM can distinguish different products. 1. Add a core API. 2. Apply the API to bytcr-rt5640 driver. changes on V3: 1. Split the core API and the API call to two patches 2. Replace misused strcat() with snprintf() 3. Code and comment fix Lu, Han (2): ASoC: core: add API for registering DMI card names ASoC: bytcr-rt5640: register DMI names to card include/sound/soc.h | 3 ++ sound/soc/intel/boards/bytcr_rt5640.c | 15 ++++++++++ sound/soc/soc-core.c | 53 +++++++++++++++++++++++++++++++++++ 3 files changed, 71 insertions(+) -- 2.5.0

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[alsa-devel] [PATCH v2] ALSA: hda - Bind with i915 only when Intel graphics is present
by Takashi Iwai 31 Mar '16

31 Mar '16

On Skylake and onwards, the HD-audio controller driver needs to bind with i915 for having the control of power well audio domain before actually probing the codec. This leads to the load of i915 driver from the audio driver side. But, there are systems that have no Intel graphics but Nvidia or AMD GPU, although they still use HD-audio bus for the onboard audio codecs. On these, loading the i915 driver is nothing but a useless memory and CPU consumption. A simple way to avoid it is just to look for the Intel graphics PCI entry beforehand, and try to bind with i915 only when such an entry is found. Currently, it assumes the PCI display class. If another class appears, this needs to be extended (although it's very unlikely). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Check also PCI vendor id to match only with Intel Now get out of a series, as this should be applied to 4.6, too. sound/hda/hdac_i915.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index d0da2508823e..6800e0c5a38f 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -287,6 +287,18 @@ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops } EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier); +/* check whether intel graphics is present */ +static bool i915_gfx_present(void) +{ + static struct pci_device_id ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), + .class = PCI_BASE_CLASS_DISPLAY << 16, + .class_mask = 0xff << 16 }, + {} + }; + return pci_dev_present(ids); +} + /** * snd_hdac_i915_init - Initialize i915 audio component * @bus: HDA core bus @@ -309,6 +321,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (WARN_ON(hdac_acomp)) return -EBUSY; + if (!i915_gfx_present()) + return -ENODEV; + acomp = kzalloc(sizeof(*acomp), GFP_KERNEL); if (!acomp) return -ENOMEM; -- 2.7.4

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[alsa-devel] ALSA license
by Li, Steve 31 Mar '16

31 Mar '16

I am trying to use ALSA in a commercial application. I see that in the introduction page, ALSA is licensed under GPL and LGPL. Can you tell me which part of ALSA is licensed under LGPL? What is the best way to find out this information? Thanks Steve ______________________________________________________________________ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com ______________________________________________________________________

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[alsa-devel] [PATCH] ALSA: hda: add AMD Polaris-10/11 AZ PCI IDs with proper driver caps
by Maruthi Srinivas Bayyavarapu 31 Mar '16

31 Mar '16

This commit fixes garbled audio on Polaris-10/11 variants Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> Acked-by: Christian König <christian.koenig(a)amd.com> --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2624cfe..b680be0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2361,6 +2361,10 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaae8), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaae0), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaaf0), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA }, /* VIA GFX VT7122/VX900 */ -- 1.9.1

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[alsa-devel] [PATCH] ASoC: dpcm: prevent error for paths including static links
by Piotr Stankiewicz 31 Mar '16

31 Mar '16

According to the DPCM documentation using static codec <-> codec links along dynamic ones is a valid use case, however this causes errors of the following form to be printed to the kernel log: ASoC: can't get [playback|caputure] BE for <widget name> ASoC: no BE found for <widget name> This happens when setting up, e.g. a route starting with a dynamic DAI, which passes through a static DAI (say, "Some FE" -> "Some BE" -> "Codec to Modem link"). All DAPM widgets involved in that path will be passed to dpcm_add_paths in order to establish FE <-> BE connections, which will result in dpcm_get_be trying to locate back-ends for dynamic, as well as static links, causing a spurious error to be logged for the latter. This patch changes dpcm_get_be to look up a non front-end runtime for a stream widget, and renames it accordingly. Like this both dynamic and static links can be handled properly in dpcm_add_paths, i.e. an actual missing BE can be distinguished from a normal link. Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1af4f23..ff5ea7e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1220,47 +1220,47 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) } } -/* get BE for DAI widget and stream */ -static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, +/* get runtime for DAI widget and stream */ +static struct snd_soc_pcm_runtime *dpcm_get_rtd(struct snd_soc_card *card, struct snd_soc_dapm_widget *widget, int stream) { - struct snd_soc_pcm_runtime *be; + struct snd_soc_pcm_runtime *rtd; int i; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - list_for_each_entry(be, &card->rtd_list, list) { + list_for_each_entry(rtd, &card->rtd_list, list) { - if (!be->dai_link->no_pcm) + if (rtd->dai_link->dynamic) continue; - if (be->cpu_dai->playback_widget == widget) - return be; + if (rtd->cpu_dai->playback_widget == widget) + return rtd; - for (i = 0; i < be->num_codecs; i++) { - struct snd_soc_dai *dai = be->codec_dais[i]; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *dai = rtd->codec_dais[i]; if (dai->playback_widget == widget) - return be; + return rtd; } } } else { - list_for_each_entry(be, &card->rtd_list, list) { + list_for_each_entry(rtd, &card->rtd_list, list) { - if (!be->dai_link->no_pcm) + if (rtd->dai_link->dynamic) continue; - if (be->cpu_dai->capture_widget == widget) - return be; + if (rtd->cpu_dai->capture_widget == widget) + return rtd; - for (i = 0; i < be->num_codecs; i++) { - struct snd_soc_dai *dai = be->codec_dais[i]; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *dai = rtd->codec_dais[i]; if (dai->capture_widget == widget) - return be; + return rtd; } } } - dev_err(card->dev, "ASoC: can't get %s BE for %s\n", + dev_err(card->dev, "ASoC: can't get %s runtime for %s\n", stream ? "capture" : "playback", widget->name); return NULL; } @@ -1367,15 +1367,15 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; } - /* is there a valid BE rtd for this widget */ - be = dpcm_get_be(card, list->widgets[i], stream); + /* is there a valid rtd for this widget */ + be = dpcm_get_rtd(card, list->widgets[i], stream); if (!be) { - dev_err(fe->dev, "ASoC: no BE found for %s\n", + dev_err(fe->dev, "ASoC: no runtime found for %s\n", list->widgets[i]->name); continue; } - /* make sure BE is a real BE */ + /* check if the runtime is a BE */ if (!be->dai_link->no_pcm) continue; -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: nau8825: Fix jack detection across suspend
by Ben Zhang 31 Mar '16

31 Mar '16

Jack plug status is rechecked at resume to handle plug/unplug in S3 when the chip has no power. Suspend/resume callbacks are moved from the i2c dev_pm_ops to snd_soc_codec_driver. soc_resume_deferred is a delayed work which may trigger nau8825_set_bias_level. The bias change races against dev_pm_ops, causing jack detection issues. soc_resume_deferred ensures bias change and snd_soc_codec_driver suspend/resume are sequenced correctly. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 126 +++++++++++++++++++++++++-------------------- 1 file changed, 71 insertions(+), 55 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 3e7bee2..cb08a35 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -343,9 +343,12 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, 0), - /* ADC for button press detection */ - SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, - NAU8825_SAR_ADC_EN_SFT, 0), + /* ADC for button press detection. A dapm supply widget is used to + * prevent dapm_power_widgets keeping the codec at SND_SOC_BIAS_ON + * during suspend. + */ + SND_SOC_DAPM_SUPPLY("SAR", NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACL", 2, NAU8825_REG_RDAC, 12, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACR", 2, NAU8825_REG_RDAC, 13, 0, NULL, 0), @@ -607,6 +610,16 @@ static bool nau8825_is_jack_inserted(struct regmap *regmap) static void nau8825_restart_jack_detection(struct regmap *regmap) { + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + /* this will restart the entire jack detection process including MIC/GND * switching and create interrupts. We have to go from 0 to 1 and back * to 0 to restart. @@ -728,7 +741,10 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) struct regmap *regmap = nau8825->regmap; int active_irq, clear_irq = 0, event = 0, event_mask = 0; - regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + if (regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq)) { + dev_err(nau8825->dev, "failed to read irq status\n"); + return IRQ_NONE; + } if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == NAU8825_JACK_EJECTION_DETECTED) { @@ -1198,33 +1214,74 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, return ret; } } - - ret = regcache_sync(nau8825->regmap); - if (ret) { - dev_err(codec->dev, - "Failed to sync cache: %d\n", ret); - return ret; - } } - break; case SND_SOC_BIAS_OFF: if (nau8825->mclk_freq) clk_disable_unprepare(nau8825->mclk); - - regcache_mark_dirty(nau8825->regmap); break; } return 0; } +#ifdef CONFIG_PM +static int nau8825_suspend(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + disable_irq(nau8825->irq); + regcache_cache_only(nau8825->regmap, true); + regcache_mark_dirty(nau8825->regmap); + + return 0; +} + +static int nau8825_resume(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + /* The chip may lose power and reset in S3. regcache_sync restores + * register values including configurations for sysclk, irq, and + * jack/button detection. + */ + regcache_cache_only(nau8825->regmap, false); + regcache_sync(nau8825->regmap); + + /* Check the jack plug status directly. If the headset is unplugged + * during S3 when the chip has no power, there will be no jack + * detection irq even after the nau8825_restart_jack_detection below, + * because the chip just thinks no headset has ever been plugged in. + */ + if (!nau8825_is_jack_inserted(nau8825->regmap)) { + nau8825_eject_jack(nau8825); + snd_soc_jack_report(nau8825->jack, 0, SND_JACK_HEADSET); + } + + enable_irq(nau8825->irq); + + /* Run jack detection to check the type (OMTP or CTIA) of the headset + * if there is one. This handles the case where a different type of + * headset is plugged in during S3. This triggers an IRQ iff a headset + * is already plugged in. + */ + nau8825_restart_jack_detection(nau8825->regmap); + + return 0; +} +#else +#define nau8825_suspend NULL +#define nau8825_resume NULL +#endif + static struct snd_soc_codec_driver nau8825_codec_driver = { .probe = nau8825_codec_probe, .set_sysclk = nau8825_set_sysclk, .set_pll = nau8825_set_pll, .set_bias_level = nau8825_set_bias_level, .suspend_bias_off = true, + .suspend = nau8825_suspend, + .resume = nau8825_resume, .controls = nau8825_controls, .num_controls = ARRAY_SIZE(nau8825_controls), @@ -1334,16 +1391,6 @@ static int nau8825_setup_irq(struct nau8825 *nau8825) regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); - /* Chip needs one FSCLK cycle in order to generate interrupts, - * as we cannot guarantee one will be provided by the system. Turning - * master mode on then off enables us to generate that FSCLK cycle - * with a minimum of contention on the clock bus. - */ - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); - ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL, nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, "nau8825", nau8825); @@ -1411,36 +1458,6 @@ static int nau8825_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM_SLEEP -static int nau8825_suspend(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - disable_irq(client->irq); - regcache_cache_only(nau8825->regmap, true); - regcache_mark_dirty(nau8825->regmap); - - return 0; -} - -static int nau8825_resume(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - regcache_cache_only(nau8825->regmap, false); - regcache_sync(nau8825->regmap); - enable_irq(client->irq); - - return 0; -} -#endif - -static const struct dev_pm_ops nau8825_pm = { - SET_SYSTEM_SLEEP_PM_OPS(nau8825_suspend, nau8825_resume) -}; - static const struct i2c_device_id nau8825_i2c_ids[] = { { "nau8825", 0 }, { } @@ -1467,7 +1484,6 @@ static struct i2c_driver nau8825_driver = { .name = "nau8825", .of_match_table = of_match_ptr(nau8825_of_ids), .acpi_match_table = ACPI_PTR(nau8825_acpi_match), - .pm = &nau8825_pm, }, .probe = nau8825_i2c_probe, .remove = nau8825_i2c_remove, -- 2.8.0.rc3.226.g39d4020

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[alsa-devel] [PATCH] ASoC: intel: add function stub when ACPI is not enabled
by Randy Dunlap 31 Mar '16

31 Mar '16

From: Randy Dunlap <rdunlap(a)infradead.org> Add function stub for "sst_acpi_find_name_from_hid()" when CONFIG_ACPI is not enabled so that the driver will build successfully. This fixes the following build errors: (loadable module) ERROR: "sst_acpi_find_name_from_hid" [sound/soc/intel/boards/snd-soc-sst-bytcr-rt5640.ko] undefined! (or built-in) bytcr_rt5640.c:(.text+0x26fc52): undefined reference to `sst_acpi_find_name_from_hid' Reported-by: Borislav Petkov <bp(a)alien8.de> Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/intel/common/sst-acpi.h | 9 +++++++++ 1 file changed, 9 insertions(+) --- linux-next-20160308.orig/sound/soc/intel/common/sst-acpi.h +++ linux-next-20160308/sound/soc/intel/common/sst-acpi.h @@ -12,10 +12,19 @@ * */ +#include <linux/kconfig.h> +#include <linux/stddef.h> #include <linux/acpi.h> /* translation fron HID to I2C name, needed for DAI codec_name */ +#if IS_ENABLED(CONFIG_ACPI) const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]); +#else +inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) +{ + return NULL; +} +#endif /* acpi match */ struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines);

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