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* ALSA: hda - Limit 40bit DMA for AMD HDMI controllersTakashi Iwai2014-12-141-3/+11
| | | | | | | | | | | | | | | | | commit 413cbf469a19e7662ba5025695bf5a573927105a upstream. AMD/ATI HDMI controller chip models, we already have a filter to lower to 32bit DMA, but the rest are supposed to be working with 64bit although the hardware doesn't really work with 63bit but only with 40 or 48bit DMA. In this patch, we take 40bit DMA for safety for the AMD/ATI controllers as the graphics drivers does. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Benjamin Herrenschmidt <benh@kernel.crashing.org> [bwh: Backported to 3.2: - Adjust context - s/AZX_GCAP_64OK/ICH6_GCAP_64OK/] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ASoC: sgtl5000: Fix SMALL_POP bit definitionFabio Estevam2014-12-142-3/+2
| | | | | | | | | | | | | | | | | | | | commit c251ea7bd7a04f1f2575467e0de76e803cf59149 upstream. On a mx28evk with a sgtl5000 codec we notice a loud 'click' sound to happen 5 seconds after the end of a playback. The SMALL_POP bit should fix this, but its definition is incorrect: according to the sgtl5000 manual it is bit 0 of CHIP_REF_CTRL register, not bit 1. Fix the definition accordingly and enable the bit as intended per the code comment. After applying this change, no loud 'click' sound is heard after playback Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: usb-audio: Fix device_del() sysfs warnings at disconnectTakashi Iwai2014-12-141-2/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 0725dda207e95ff25f1aa01432250323e0ec49d6 upstream. Some USB-audio devices show weird sysfs warnings at disconnecting the devices, e.g. usb 1-3: USB disconnect, device number 3 ------------[ cut here ]------------ WARNING: CPU: 0 PID: 973 at fs/sysfs/group.c:216 device_del+0x39/0x180() sysfs group ffffffff8183df40 not found for kobject 'midiC1D0' Call Trace: [<ffffffff814a3e38>] ? dump_stack+0x49/0x71 [<ffffffff8103cb72>] ? warn_slowpath_common+0x82/0xb0 [<ffffffff8103cc55>] ? warn_slowpath_fmt+0x45/0x50 [<ffffffff813521e9>] ? device_del+0x39/0x180 [<ffffffff81352339>] ? device_unregister+0x9/0x20 [<ffffffff81352384>] ? device_destroy+0x34/0x40 [<ffffffffa00ba29f>] ? snd_unregister_device+0x7f/0xd0 [snd] [<ffffffffa025124e>] ? snd_rawmidi_dev_disconnect+0xce/0x100 [snd_rawmidi] [<ffffffffa00c0192>] ? snd_device_disconnect+0x62/0x90 [snd] [<ffffffffa00c025c>] ? snd_device_disconnect_all+0x3c/0x60 [snd] [<ffffffffa00bb574>] ? snd_card_disconnect+0x124/0x1a0 [snd] [<ffffffffa02e54e8>] ? usb_audio_disconnect+0x88/0x1c0 [snd_usb_audio] [<ffffffffa015260e>] ? usb_unbind_interface+0x5e/0x1b0 [usbcore] [<ffffffff813553e9>] ? __device_release_driver+0x79/0xf0 [<ffffffff81355485>] ? device_release_driver+0x25/0x40 [<ffffffff81354e11>] ? bus_remove_device+0xf1/0x130 [<ffffffff813522b9>] ? device_del+0x109/0x180 [<ffffffffa01501d5>] ? usb_disable_device+0x95/0x1f0 [usbcore] [<ffffffffa014634f>] ? usb_disconnect+0x8f/0x190 [usbcore] [<ffffffffa0149179>] ? hub_thread+0x539/0x13a0 [usbcore] [<ffffffff810669f5>] ? sched_clock_local+0x15/0x80 [<ffffffff81066c98>] ? sched_clock_cpu+0xb8/0xd0 [<ffffffff81070730>] ? bit_waitqueue+0xb0/0xb0 [<ffffffffa0148c40>] ? usb_port_resume+0x430/0x430 [usbcore] [<ffffffffa0148c40>] ? usb_port_resume+0x430/0x430 [usbcore] [<ffffffff8105973e>] ? kthread+0xce/0xf0 [<ffffffff81059670>] ? kthread_create_on_node+0x1c0/0x1c0 [<ffffffff814a8b7c>] ? ret_from_fork+0x7c/0xb0 [<ffffffff81059670>] ? kthread_create_on_node+0x1c0/0x1c0 ---[ end trace 40b1928d1136b91e ]--- This comes from the fact that usb-audio driver may receive the disconnect callback multiple times, per each usb interface. When a device has both audio and midi interfaces, it gets called twice, and currently the driver tries to release resources at the last call. At this point, the first parent interface has been already deleted, thus deleting a child of the first parent hits such a warning. For fixing this problem, we need to call snd_card_disconnect() and cancel pending operations at the very first disconnect while the release of the whole objects waits until the last disconnect call. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=80931 Reported-and-tested-by: Tomas Gayoso <tgayoso@gmail.com> Reported-and-tested-by: Chris J Arges <chris.j.arges@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> [bwh: Backported to 3.2: adjust context] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ASoC: fsi: remove unsupported PAUSE flagKuninori Morimoto2014-12-141-2/+1
| | | | | | | | | | | | commit c1b9b9b1ad2df6144ca3fbe6989f7bd9ea5c5562 upstream. FSI doesn't support PAUSE. Remove SNDRV_PCM_INFO_PAUSE flags from snd_pcm_hardware info Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org> [bwh: Backported to 3.2: adjust context] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: pcm: Zero-clear reserved fields of PCM status ioctl in compat modeTakashi Iwai2014-12-141-0/+2
| | | | | | | | | | | | | commit 317168d0c766defd14b3d0e9c2c4a9a258b803ee upstream. In compat mode, we copy each field of snd_pcm_status struct but don't touch the reserved fields, and this leaves uninitialized values there. Meanwhile the native ioctl does zero-clear the whole structure, so we should follow the same rule in compat mode, too. Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: emu10k1: Fix deadlock in synth voice lookupTakashi Iwai2014-12-141-4/+2
| | | | | | | | | | | | | | | | | | commit 95926035b187cc9fee6fb61385b7da9c28123f74 upstream. The emu10k1 voice allocator takes voice_lock spinlock. When there is no empty stream available, it tries to release a voice used by synth, and calls get_synth_voice. The callback function, snd_emu10k1_synth_get_voice(), however, also takes the voice_lock, thus it deadlocks. The fix is simply removing the voice_lock holds in snd_emu10k1_synth_get_voice(), as this is always called in the spinlock context. Reported-and-tested-by: Arthur Marsh <arthur.marsh@internode.on.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: pcm: fix fifo_size frame calculationClemens Ladisch2014-11-051-3/+5
| | | | | | | | | | | | | | | commit a9960e6a293e6fc3ed414643bb4e4106272e4d0a upstream. The calculated frame size was wrong because snd_pcm_format_physical_width() actually returns the number of bits, not bytes. Use snd_pcm_format_size() instead, which not only returns bytes, but also simplifies the calculation. Fixes: 8bea869c5e56 ("ALSA: PCM midlevel: improve fifo_size handling") Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: hda - Fix COEF setups for ALC1150 codecTakashi Iwai2014-11-051-0/+2
| | | | | | | | | | | | | commit acf08081adb5e8fe0519eb97bb49797ef52614d6 upstream. ALC1150 codec seems to need the COEF- and PLL-setups just like its compatible ALC882 codec. Some machines (e.g. SunMicro X10SAT) show the problem like too low output volumes unless the COEF setup is applied. Reported-and-tested-by: Dana Goyette <danagoyette@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: hda/realtek - Avoid setting wrong COEF on ALC269 & coTakashi Iwai2014-09-131-5/+12
| | | | | | | | | | | | | | | | | | | | | commit f3ee07d8b6e061bf34a7167c3f564e8da4360a99 upstream. ALC269 & co have many vendor-specific setups with COEF verbs. However, some verbs seem specific to some codec versions and they result in the codec stalling. Typically, such a case can be avoided by checking the return value from reading a COEF. If the return value is -1, it implies that the COEF is invalid, thus it shouldn't be written. This patch adds the invalid COEF checks in appropriate places accessing ALC269 and its variants. The patch actually fixes the resume problem on Acer AO725 laptop. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181 Tested-by: Francesco Muzio <muziofg@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> [bwh: Backported to 3.2: adjust context] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ASoC: pxa-ssp: drop SNDRV_PCM_FMTBIT_S24_LEDaniel Mack2014-09-131-3/+1
| | | | | | | | | | | | commit 9301503af016eb537ccce76adec0c1bb5c84871e upstream. This mode is unsupported, as the DMA controller can't do zero-padding of samples. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Johannes Stezenbach <js@sig21.net> Signed-off-by: Mark Brown <broonie@linaro.org> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: virtuoso: add Xonar Essence STX II supportClemens Ladisch2014-09-133-4/+13
| | | | | | | | | | | | | commit f42bb22243d2ae264d721b055f836059fe35321f upstream. Just add the PCI ID for the STX II. It appears to work the same as the STX, except for the addition of the not-yet-supported daughterboard. Tested-by: Mario <fugazzi99@gmail.com> Tested-by: corubba <corubba@gmx.de> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: virtuoso: Xonar DSX supportSergiu Giurgiu2014-09-132-1/+6
| | | | | | | | | | | | | | | | | | | commit 4492363251235c4499a2d073c5f09121ea23d39d upstream. This patch adds support for ASUS - Xonar DSX sound cards. Tested on openSUSE 12.2 with kernel: Linux 3.4.6-2.10-desktop #1 SMP PREEMPT Thu Jul 26 09:36:26 UTC 2012 (641c197) x86_64 x86_64 x86_64 GNU/Linux Works: - play sounds - adjust volume on master channel. - mute . Since Xonar DS uses the same chip, everything that works for DS should work for DSX as well. Signed-off-by: Sergiu Giurgiu <sgiurgiu11@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ASoC: samsung: Correct I2S DAI suspend/resume opsSylwester Nawrocki2014-09-131-10/+6
| | | | | | | | | | | | | | commit d3d4e5247b013008a39e4d5f69ce4c60ed57f997 upstream. We should save/restore relevant I2S registers regardless of the dai->active flag, otherwise some settings are being lost after system suspend/resume cycle. E.g. I2S slave mode set only during dai initialization is not preserved and the device ends up in master mode after system resume. Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com> Signed-off-by: Mark Brown <broonie@linaro.org> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: control: Make sure that id->index does not overflowLars-Peter Clausen2014-07-111-0/+3
| | | | | | | | | | | | | | | | commit 883a1d49f0d77d30012f114b2e19fc141beb3e8e upstream. The ALSA control code expects that the range of assigned indices to a control is continuous and does not overflow. Currently there are no checks to enforce this. If a control with a overflowing index range is created that control becomes effectively inaccessible and unremovable since snd_ctl_find_id() will not be able to find it. This patch adds a check that makes sure that controls with a overflowing index range can not be created. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: control: Handle numid overflowLars-Peter Clausen2014-07-111-0/+4
| | | | | | | | | | | | | | | | | | commit ac902c112d90a89e59916f751c2745f4dbdbb4bd upstream. Each control gets automatically assigned its numids when the control is created. The allocation is done by incrementing the numid by the amount of allocated numids per allocation. This means that excessive creation and destruction of controls (e.g. via SNDRV_CTL_IOCTL_ELEM_ADD/REMOVE) can cause the id to eventually overflow. Currently when this happens for the control that caused the overflow kctl->id.numid + kctl->count will also over flow causing it to be smaller than kctl->id.numid. Most of the code assumes that this is something that can not happen, so we need to make sure that it won't happen Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: control: Don't access controls outside of protected regionsLars-Peter Clausen2014-07-111-5/+10
| | | | | | | | | | | | | commit fd9f26e4eca5d08a27d12c0933fceef76ed9663d upstream. A control that is visible on the card->controls list can be freed at any time. This means we must not access any of its memory while not holding the controls_rw_lock. Otherwise we risk a use after free access. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: control: Fix replacing user controlsLars-Peter Clausen2014-07-111-16/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 82262a46627bebb0febcc26664746c25cef08563 upstream. There are two issues with the current implementation for replacing user controls. The first is that the code does not check if the control is actually a user control and neither does it check if the control is owned by the process that tries to remove it. That allows userspace applications to remove arbitrary controls, which can cause a user after free if a for example a driver does not expect a control to be removed from under its feed. The second issue is that on one hand when a control is replaced the user_ctl_count limit is not checked and on the other hand the user_ctl_count is increased (even though the number of user controls does not change). This allows userspace, once the user_ctl_count limit as been reached, to repeatedly replace a control until user_ctl_count overflows. Once that happens new controls can be added effectively bypassing the user_ctl_count limit. Both issues can be fixed by instead of open-coding the removal of the control that is to be replaced to use snd_ctl_remove_user_ctl(). This function does proper permission checks as well as decrements user_ctl_count after the control has been removed. Note that by using snd_ctl_remove_user_ctl() the check which returns -EBUSY at beginning of the function if the control already exists is removed. This is not a problem though since the check is quite useless, because the lock that is protecting the control list is released between the check and before adding the new control to the list, which means that it is possible that a different control with the same settings is added to the list after the check. Luckily there is another check that is done while holding the lock in snd_ctl_add(), so we'll rely on that to make sure that the same control is not added twice. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: control: Protect user controls against concurrent accessLars-Peter Clausen2014-07-112-6/+26
| | | | | | | | | | | | | | | | | commit 07f4d9d74a04aa7c72c5dae0ef97565f28f17b92 upstream. The user-control put and get handlers as well as the tlv do not protect against concurrent access from multiple threads. Since the state of the control is not updated atomically it is possible that either two write operations or a write and a read operation race against each other. Both can lead to arbitrary memory disclosure. This patch introduces a new lock that protects user-controls from concurrent access. Since applications typically access controls sequentially than in parallel a single lock per card should be fine. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: hda - Add quirk for external mic on Lifebook U904David Henningsson2014-07-111-0/+9
| | | | | | | | | | | | | | | | | | commit 2041d56464a067461d7cc21734a0f024587ed2ff upstream. According to the bug reporter (Данило Шеган), the external mic starts to work and has proper jack detection if only pin 0x19 is marked properly as an external headset mic. AlsaInfo at https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1328587/+attachment/4128991/+files/AlsaInfo.txt BugLink: https://bugs.launchpad.net/bugs/1328587 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> [bwh: Backported to 3.2: - s/struct hda_pintbl/struct alc_pincfg/ - s/HDA_FIXUP_PINS/ALC_FIXUP_PINS/ Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: hda/realtek - Add support of ALC891 codecKailang Yang2014-07-111-0/+1
| | | | | | | | | | commit b6c5fbad16aa5026f508093a8d651c25e1cb6179 upstream. New codec support for ALC891. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: ice1712: Fix boundary checks in PCM pointer opsTakashi Iwai2014-04-301-5/+10
| | | | | | | | | | | | | | | | commit 4f8e940095536bc002a81666a4107a581c84e9b9 upstream. PCM pointer callbacks in ice1712 driver check the buffer size boundary wrongly between bytes and frames. This leads to PCM core warnings like: snd_pcm_update_hw_ptr0: 105 callbacks suppressed ALSA pcm_lib.c:352 BUG: pcmC3D0c:0, pos = 5461, buffer size = 5461, period size = 2730 This patch fixes these checks to be placed after the proper unit conversions. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: hda - Enable beep for ASUS 1015EW. Trevor King2014-04-301-0/+1
| | | | | | | | | | | | | | | commit a4b7f21d7b42b33609df3f86992a8deff80abfaf upstream. The `lspci -nnvv` output contains (wrapped for line length): 00:1b.0 Audio device [0403]: Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller [8086:1e20] (rev 04) Subsystem: ASUSTeK Computer Inc. Device [1043:115d] Signed-off-by: W. Trevor King <wking@tremily.us> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: oxygen: Xonar DG(X): modify DAC routingClemens Ladisch2014-04-021-0/+10
| | | | | | | | | | | | commit 1f91ecc14deea9461aca93273d78871ec4d98fcd upstream. When selecting the audio output destinations (headphones, FP headphones, multichannel output), unnecessary I2S channels are digitally muted to avoid invalid signal levels on the other outputs. Signed-off-by: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: oxygen: Xonar DG(X): capture from I2S channel 1, not 2Roman Volkov2014-04-021-1/+1
| | | | | | | | | | | | | commit 3dd77654fb1d7f68b9739f3039bad8dbbc0739f8 upstream. Actually CS4245 connected to the I2S channel 1 for capture, not channel 2. Otherwise capturing and playback does not work for CS4245. Signed-off-by: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> [bwh: Backported to 3.2: adjust context] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: usb-audio: Add quirk for Logitech Webcam C500Takashi Iwai2014-04-021-0/+1
| | | | | | | | | | commit e805ca8b0a9b6c91099c0eaa4b160a1196a4ae25 upstream. Logitech C500 (046d:0807) needs the same workaround like other Logitech Webcams. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ASoC: sta32x: Fix wrong enum for limiter2 release rateTakashi Iwai2014-04-021-1/+1
| | | | | | | | | | | | | | | | | | | | | | | commit b3619b288b621e63f66908045f48495869a996a6 upstream. There is a typo in the Limiter2 Release Rate control, a wrong enum for Limiter1 is assigned. It must point to Limiter2. Spotted by a compile warning: In file included from sound/soc/codecs/sta32x.c:34:0: sound/soc/codecs/sta32x.c:223:29: warning: ‘sta32x_limiter2_release_rate_enum’ defined but not used [-Wunused-variable] static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum, ^ include/sound/soc.h:275:18: note: in definition of macro ‘SOC_ENUM_DOUBLE_DECL’ struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \ ^ sound/soc/codecs/sta32x.c:223:8: note: in expansion of macro ‘SOC_ENUM_SINGLE_DECL’ static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum, ^ Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@linaro.org> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ASoC: wm8770: Fix wrong number of enum itemsTakashi Iwai2014-04-021-2/+2
| | | | | | | | | | | | | | | commit 7a6c0a58dc824523966f212c76322d47c5b0e6fe upstream. wm8770 codec driver defines ain_enum with a wrong number of items. Use SOC_ENUM_DOUBLE_DECL() macro and it's automatically fixed. Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ASoC: sta32x: Fix array access overflowTakashi Iwai2014-04-021-36/+36
| | | | | | | | | | | | | | | commit 025c3fa9256d4c54506b7a29dc3befac54f5c68d upstream. Preset EQ enum of sta32x codec driver declares too many number of items and it may lead to the access over the actual array size. Use SOC_ENUM_SINGLE_DECL() helper and it's automatically fixed. Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: usb-audio: work around KEF X300A firmware bugClemens Ladisch2014-04-021-0/+9
| | | | | | | | | | | | commit 624aef494f86ed0c58056361c06347ad62b26806 upstream. When the driver tries to access Function Unit 10, the KEF X300A speakers' firmware apparently locks up, making even PCM streaming impossible. Work around this by ignoring this FU. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: hda/realtek - Avoid invalid COEFs for ALC271XTakashi Iwai2014-04-021-0/+3
| | | | | | | | | | | | | | | | | | commit d3c56568f43807135f2c2a09582a69f809f0d8b7 upstream. We've seen often problems after suspend/resume on Acer Aspire One AO725 with ALC271X codec as reported in kernel bugzilla, and it turned out that some COEFs doesn't work and triggers the codec communication stall. Since these magic COEF setups are specific to ALC269VB for some PLL configurations, the machine works even without these manual adjustment. So, let's simply avoid applying them for ALC271X. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181 Signed-off-by: Takashi Iwai <tiwai@suse.de> [bwh: Backported to 3.2: return 0] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: Enable CONFIG_ZONE_DMA for smaller PCI DMA masksTakashi Iwai2014-04-021-0/+12
| | | | | | | | | | | | | | | commit 80ab8eae70e51d578ebbeb228e0f7a562471b8b7 upstream. The PCI devices with DMA masks smaller than 32bit should enable CONFIG_ZONE_DMA. Since the recent change of page allocator, page allocations via dma_alloc_coherent() with the limited DMA mask bits may fail more frequently, ended up with no available buffers, when CONFIG_ZONE_DMA isn't enabled. With CONFIG_ZONE_DMA, the system has much more chance to obtain such pages. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=68221 Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constantLars-Peter Clausen2014-04-021-1/+1
| | | | | | | | | | | | commit e20970ada3f699c113fe64b04492af083d11a7d8 upstream. The driver defines ADAU1701_SEROCTL_WORD_LEN_16 as 0x10 while it should be b10, so 0x2. This patch fixes it. Reported-by: Magnus Reftel <magnus.reftel@lockless.no> Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: rme9652: fix a missing comma in channel_map_9636_ds[]Takashi Iwai2014-04-021-1/+1
| | | | | | | | | | | | commit 770bd4bf2e664939a9dacd3d26ec9ff7a3933210 upstream. The lack of comma leads to the wrong channel for an SPDIF channel. Unfortunately this wasn't caught by compiler because it's still a valid expression. Reported-by: Alexander Aristov <aristov.alexander@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ASoC: wm8904: fix DSP mode B configurationBo Shen2014-02-151-1/+1
| | | | | | | | | | | | commit f0199bc5e3a3ec13f9bc938556517ec430b36437 upstream. When wm8904 work in DSP mode B, we still need to configure it to work in DSP mode. Or else, it will work in Right Justified mode. Signed-off-by: Bo Shen <voice.shen@atmel.com> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@linaro.org> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: Add SNDRV_PCM_STATE_PAUSED case in wait_for_avail functionJongHo Kim2014-02-151-0/+2
| | | | | | | | | | | | | commit ed697e1aaf7237b1a62af39f64463b05c262808d upstream. When the process is sleeping at the SNDRV_PCM_STATE_PAUSED state from the wait_for_avail function, the sleep process will be woken by timeout(10 seconds). Even if the sleep process wake up by timeout, by this patch, the process will continue with sleep and wait for the other state. Signed-off-by: JongHo Kim <furmuwon@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: hda - Add enable_msi=0 workaround for four HP machinesDavid Henningsson2014-02-151-0/+4
| | | | | | | | | | | | | | | | | | | commit 693e0cb052c607e2d41edf9e9f1fa99ff8c266c1 upstream. While enabling these machines, we found we would sometimes lose an interrupt if we change hardware volume during playback, and that disabling msi fixed this issue. (Losing the interrupt caused underruns and crackling audio, as the one second timeout is usually bigger than the period size.) The machines were all machines from HP, running AMD Hudson controller, and Realtek ALC282 codec. BugLink: https://bugs.launchpad.net/bugs/1260225 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> [bwh: Backported to 3.2: adjust context] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ASoC: wm8731: fix dsp mode configurationBo Shen2014-01-031-2/+2
| | | | | | | | | | | | | commit b4af6ef99a60c5b56df137d7accd81ba1ee1254e upstream. According to WM8731 "PD, Rev 4.9 October 2012" datasheet, when it works in DSP mode A, LRP = 1, while works in DSP mode B, LRP = 0. So, fix LRP for DSP mode as the datesheet specification. Signed-off-by: Bo Shen <voice.shen@atmel.com> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@linaro.org> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: hda/realtek - Set pcbeep amp for ALC668Kailang Yang2014-01-031-0/+1
| | | | | | | | | | commit 9ad54547cf6f4410eba83bb95dfd2a0966718d6d upstream. Set the missing pcbeep default amp for ALC668. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: hda/realtek - Add support of ALC231 codecKailang Yang2014-01-031-0/+1
| | | | | | | | | | | commit ba4c4d0a9021ab034554d532a98133d668b87599 upstream. It's compatible with ALC269. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> [bwh: Backported to 3.2: adjust context] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ASoC: wm8990: Mark the register map as dirty when powering downMark Brown2014-01-031-0/+2
| | | | | | | | | | commit 2ab2b74277a86afe0dd92976db695a2bb8b93366 upstream. Otherwise we'll skip sync on resume. Signed-off-by: Mark Brown <broonie@linaro.org> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: pcsp: Fix the order of input device unregistrationTakashi Iwai2014-01-031-1/+1
| | | | | | | | | | | | commit 6408eac2665955343cd0e4bcd7d6237ce39611ed upstream. The current code may access to the already freed object. The input device must be accessed and unregistered before freeing the top level sound object. Signed-off-by: Takashi Iwai <tiwai@suse.de> [bwh: Backported to 3.2: adjust context] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: msnd: Avoid duplicated driver nameTakashi Iwai2014-01-031-2/+2
| | | | | | | | | | | | | | | | | | | commit 092f9cd16aac7d054af1755c945f37c1b33399e6 upstream. msnd_pinnacle.c is used for both snd-msnd-pinnacle and snd-msnd-classic drivers, and both should have different driver names. Using the same driver name results in the sysfs warning for duplicated entries like kobject: 'msnd-pinnacle.7' (cec33408): kobject_release, parent (null) (delayed) kobject: 'msnd-pinnacle' (cecd4980): kobject_release, parent cf3ad9b0 (delayed) ------------[ cut here ]------------ WARNING: CPU: 0 PID: 1 at fs/sysfs/dir.c:486 sysfs_warn_dup+0x7d/0xa0() sysfs: cannot create duplicate filename '/bus/isa/drivers/msnd-pinnacle' ...... Reported-by: Fengguang Wu <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: hda - Add support for CX20952Takashi Iwai2014-01-031-0/+3
| | | | | | | | | | commit 8f42d7698751a45cd9f7134a5da49bc5b6206179 upstream. It's a superset of the existing CX2075x codecs, so we can reuse the existing parser code. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ASoC: ak4642: prevent un-necessary changes to SG_SL1Phil Edworthy2014-01-031-1/+1
| | | | | | | | | | | | | | commit 7b5bfb82882b9b1c8423ce0ed6852ca3762d967a upstream. If you record the sound during playback, the playback sound becomes silent. Modify so that the codec driver does not clear SG_SL1::DACL bit which is controlled under widget Signed-off-by: Phil Edworthy <phil.edworthy@renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: 6fire: Fix probe of multiple cardsTakashi Iwai2014-01-031-1/+1
| | | | | | | | | | | | | commit 9b389a8a022110b4bc055a19b888283544d9eba6 upstream. The probe code of snd-usb-6fire driver overrides the devices[] pointer wrongly without checking whether it's already occupied or not. This would screw up the device disconnection later. Spotted by coverity CID 141423. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ASoC: wm_hubs: Add missing break in hp_supply_event()Takashi Iwai2013-11-281-0/+1
| | | | | | | | | | commit 268ff14525edba31da29a12a9dd693cdd6a7872e upstream. Spotted by coverity CID 115170. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@linaro.org> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: hda - Add a fixup for ASUS N76VZTakashi Iwai2013-11-281-0/+1
| | | | | | | | | | | commit 6fc16e58adf50c0f1e4478538983fb5ff6f453d4 upstream. ASUS N76VZ needs the same fixup as N56VZ for supporting the boost speaker. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=846529 Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ASoC: dapm: Fix source list debugfs outputsTakashi Iwai2013-11-281-1/+1
| | | | | | | | | | | | commit ff18620c2157671a8ee21ebb8e6a3520ea209b1f upstream. ... due to a copy & paste error. Spotted by coverity CID 710923. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@linaro.org> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: hda - Add fixup for ASUS N56VZTakashi Iwai2013-11-281-0/+1
| | | | | | | | | | | commit c6cc3d58b4042f5cadae653ff8d3df26af1a0169 upstream. ASUS N56VZ needs a fixup for the bass speaker pin, which was already provided via model=asus-mode4. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=841645 Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
* ALSA: snd-usb-usx2y: remove bogus frame checksDaniel Mack2013-11-282-25/+4
| | | | | | | | | | | | | | | | | | | commit a9d14bc0b188a822e42787d01e56c06fe9750162 upstream. The frame check in i_usX2Y_urb_complete() and i_usX2Y_usbpcm_urb_complete() is bogus and produces false positives as described in this LAU thread: http://linuxaudio.org/mailarchive/lau/2013/5/20/200177 This patch removes the check code entirely. Cc: fzu@wemgehoertderstaat.de Reported-by: Dr Nicholas J Bailey <nicholas.bailey@glasgow.ac.uk> Suggested-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
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