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* sound: oss: sb_common: Used kmemdup instead of kmalloc and memcpyAlexandru Gheorghiu2013-04-021-2/+1
| | | | | | | | | Used kmemdup instead of replicating it's behaviour with kmalloc followed by memcpy. Patch found using coccinelle. Signed-off-by: Alexandru Gheorghiu <gheorghiuandru@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* sound: oss: uart401: Used kmemdup instead of kmalloc and memcpyAlexandru Gheorghiu2013-04-021-4/+7
| | | | | | | | | Used kmemdup instead of replicating it's behaviour with kmalloc followed by memcpy. Patch found using coccinelle. Signed-off-by: Alexandru Gheorghiu <gheorghiuandru@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - VIA prefers side surrounds over HPTakashi Iwai2013-03-221-0/+20
| | | | | | | | | | | | The recent fix for the independent HP reduced the availability of the side surround output, because there are only 4 DACs for 7.1 and a HP outputs. Adjust the badness tables for VIA so that 7.1 outputs are activated for the cost of missing independent HP. Once when we implement the dynamic DAC switching to multiple outputs, this conflicts will be eased in future... Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Lower the badness for independent HP penaltyTakashi Iwai2013-03-221-1/+1
| | | | | | | The lack of independent HP mode shouldn't be too bad, but currently its badness is set a bit too high. Let's lower it. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Allow codec drivers to give own badness tablesTakashi Iwai2013-03-222-14/+29
| | | | | | | | | | | The standard badness values don't seem to fit to all preferences. Some configuration prefer the side output over the headphone, some want the speaker over the surround, etc. This patch moves the badness table pointers into hda_gen_spec, so that the codec driver can override them. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'for-linus' into for-nextTakashi Iwai2013-03-225-34/+171
|\ | | | | | | | | | | | | | | | | | | | | | | | | Merge back for-linus branch for the badness table adjustment for VIA codecs * for-linus: ALSA: hda - Fix DAC assignment for independent HP ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader ALSA: hda - Fix typo in checking IEC958 emphasis bit ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls() ALSA: snd-usb: mixer: propagate errors up the call chain ALSA: usb: Parse UAC2 extension unit like for UAC1 ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver
| * ALSA: hda - Fix DAC assignment for independent HPTakashi Iwai2013-03-211-0/+46
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The generic parser should evaluate the availability of the independent HP when specified. Otherwise a DAC without the direct connection to the corresponding pin may be assigned for the HP, but the driver doesn't check it at all. The problem was actually seen on some machines with VT1708s or equivalent codec, where DAC0 is assigned to HP although it can be connected only via aamix. This patch adds the badness evaluation for the independent HP to make it working properly. Reported-by: Lydia Wang <LydiaWang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loaderTakashi Iwai2013-03-201-23/+109
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The current DSP loader code abuses snd_hda_lock_devices() for ensuring the DSP loader not conflicting with the other normal operations. But this trick obviously doesn't work for the PM resume since the streams are kept opened there where snd_hda_lock_devices() returns -EBUSY. That means we need another lock mechanism instead of abuse. This patch provides the new lock state to azx_dev. Theoretically it's possible that the DSP loader conflicts with the stream that has been already assigned for another PCM. If it's running, the DSP loader should simply fail. If not -- it's the case for PM resume --, we should assign this stream temporarily to the DSP loader, and take it back to the PCM after finishing DSP loading. If the PCM is operated during the DSP loading, it should get an error, too. Reported-and-tested-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix typo in checking IEC958 emphasis bitTakashi Iwai2013-03-201-1/+1
| | | | | | | | | | | | | | | | | | | | There is a typo in convert_to_spdif_status() about checking the emphasis IEC958 status bit. It should check the given value instead of the resultant value. Reported-by: Martin Weishart <martin.weishart@telosalliance.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()Daniel Mack2013-03-201-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Creation of individual mixer controls may fail, but that shouldn't cause the entire mixer creation to fail. Even worse, if the mixer creation fails, that will error out the entire device probing. All the functions called by parse_audio_unit() should return -EINVAL if they find descriptors that are unsupported or believed to be malformed, so we can safely handle this error code as a non-fatal condition in snd_usb_mixer_controls(). That fixes a long standing bug which is commonly worked around by adding quirks which make the driver ignore entire interfaces. Some of them might now be unnecessary. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: Rodolfo Thomazelli <pe.soberbo@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: snd-usb: mixer: propagate errors up the call chainDaniel Mack2013-03-201-4/+6
| | | | | | | | | | | | | | | | | | | | In check_input_term() and parse_audio_feature_unit(), propagate the error value that has been returned by a failing function instead of -EINVAL. That helps cleaning up the error pathes in the mixer. Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb: Parse UAC2 extension unit like for UAC1Torstein Hegge2013-03-201-1/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | UAC2_EXTENSION_UNIT_V2 differs from UAC1_EXTENSION_UNIT, but can be handled in the same way when parsing the unit. Otherwise parse_audio_unit() fails when it sees an extension unit on a UAC2 device. UAC2_EXTENSION_UNIT_V2 is outside the range allocated by UAC1. Signed-off-by: Torstein Hegge <hegge@resisty.net> Acked-by: Daniel Mack <zonque@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driverTakashi Iwai2013-03-181-2/+2
| | | | | | | | | | | | | | | | I forgot to update spec->gpio_data in the automute hook, so it will be overridden at the init sequence, thus the machine is still silent when no headphone jack is plugged at boot time. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Enable "Headset Mic" name for some Dell Latitude devicesDavid Henningsson2013-03-211-1/+5
| | | | | | | | | | | | | | | | Now that we have a "Headset Mic" name, let's use it for some devices we know for sure has a headset mic jack. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Introduce "Headset Mic" nameDavid Henningsson2013-03-212-2/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Headset mic jacks, i e TRRS style jacks with Headphone Left, Headphone Right, Mic and GND signals, are becoming increasingly common and are now being shipped by several manufacturers. Unfortunately, the HDA specification does not give us any hint of whether a Mic pin belongs to such a jack or not, but it would still be helpful for the user to know (especially if there is one TRS Mic jack and one TRRS headset jack). This new fixup causes the first (non-dock, non-internal) mic to be a headset mic jack. The algorithm can be later refined if needed. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Add GPIO-based LED support on HP desktop machinesTakashi Iwai2013-03-181-0/+63
| | | | | | | | | | | | | | | | The new HP desktop machines have Realtek codecs and their LEDs are controlled via GPIO as for many laptop models. Add similar hooks as well as in patch_sigmatel.c for controlling LEDs. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Make the resume of digital beep setup properTakashi Iwai2013-03-181-1/+1
| | | | | | | | | | | | | | | | The verb to set up the digital beep via AC_VERB_SET_DIGI_CONVERT_2 should be executed at resume as well. Use the cached write for it being performed automatically at resume. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Fix power-saving during playing beep soundTakashi Iwai2013-03-182-11/+27
| | | | | | | | | | | | | | | | | | | | | | While playing the digital beep tone, the codec shouldn't be turned off. This patch adds proper snd_hda_power_up()/down() calls at each time when the beep is played or off. Also, this fixes automatically an unnecessary codec power-up at detaching the beep device when the beep isn't being played. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Move beep attach/detach calls in hda_generic.cTakashi Iwai2013-03-186-128/+72
| | | | | | | | | | | | | | | | | | Instead of calling snd_hda_attach_beep_device() and snd_hda_detach_beep_device() in each codec driver, move them to the generic parser. The codec driver just needs to set spec->beep_nid for activating the digital beep. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge branch 'for-linus' into for-nextTakashi Iwai2013-03-1821-60/+166
|\| | | | | | | Back-merged for refactoring beep stuff.
| * ALSA: hda/cirrus - Fix the digital beep registrationTakashi Iwai2013-03-181-4/+4
| | | | | | | | | | | | | | | | The argument passed to snd_hda_attach_beep_device() is a widget NID while spec->beep_amp holds the composed value for amp controls. Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix missing beep detach in patch_conexant.cTakashi Iwai2013-03-181-1/+7
| | | | | | | | | | | | | | | | This leaks the beep input device after module unload, which leads to Oops. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=55321 Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: documentation: Fix typo in Documentation/soundMasanari Iida2013-03-172-2/+2
| | | | | | | | | | | | | | Correct spelling typos in Documentation/sound/alsa Signed-off-by: Masanari Iida <standby24x7@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix missing EAPD/GPIO setup for Cirrus codecsTakashi Iwai2013-03-151-0/+4
| | | | | | | | | | | | | | | | During the transition to the generic parser, the hook to the codec specific automute function was forgotten. This resulted in the silent output on some MacBooks. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * sound: sequencer: cap array index in seq_chn_common_event()Dan Carpenter2013-03-151-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | "chn" here is a number between 0 and 255, but ->chn_info[] only has 16 elements so there is a potential write beyond the end of the array. If the seq_mode isn't SEQ_2 then we let the individual drivers (either opl3.c or midi_synth.c) handle it. Those functions all do a bounds check on "chn" so I haven't changed anything here. The opl3.c driver has up to 18 channels and not 16. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda/ca0132 - Remove extra setting of dsp_state.Dylan Reid2013-03-151-9/+6
| | | | | | | | | | | | | | | | spec->dsp_state is initialized to DSP_DOWNLOAD_INIT, no need to reset and check it in ca0132_download_dsp(). Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda/ca0132 - Check download state of DSP.Dylan Reid2013-03-151-2/+3
| | | | | | | | | | | | | | | | | | | | | | Instead of using the dspload_is_loaded() function, check the dsp_state that is kept in the spec. The dspload_is_loaded() function returns true if the DSP transfer was never started. This false-positive leads to multiple second delays when ca0132_setup_efaults() times out on each write. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda/ca0132 - Check if dspload_image succeeded.Dylan Reid2013-03-151-2/+6
| | | | | | | | | | | | | | | | | | | | If dspload_image() fails, it was ignored and dspload_wait_loaded() was still called. dsp_loaded should never be set to true in this case, skip it. The check in dspload_wait_loaded() return true if the DSP is loaded or if it never started. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Disable IDT eapd_switch if there are no internal speakersDavid Henningsson2013-03-141-0/+29
| | | | | | | | | | | | | | | | | | If there are no internal speakers, we should not turn the eapd switch off, because it might be necessary to keep high for Headphone. BugLink: https://bugs.launchpad.net/bugs/1155016 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix snd_hda_get_num_raw_conns() to return a correct valueTakashi Iwai2013-03-121-10/+14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the connection list expansion in hda_codec.c and hda_proc.c, the value returned from snd_hda_get_num_raw_conns() is used as the array size to store the connection list. However, the function returns simply a raw value of the AC_PAR_CONNLIST_LEN parameter, and the widget list with ranges isn't considered there. Thus it may return a smaller size than the actual list, which results in -ENOSPC in snd_hda_get_raw_conections(). This patch fixes the bug by parsing the connection list correctly also for snd_hda_get_num_raw_conns(). Reported-and-tested-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb-audio: add a workaround for the NuForce UDH-100Clemens Ladisch2013-03-121-0/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | The NuForce UDH-100 numbers its interfaces incorrectly, which makes the interface associations come out wrong, which results in the driver erroring out with the message "Audio class v2 interfaces need an interface association". Work around this by searching for the interface association descriptor also in some other place where it might have ended up. Reported-and-tested-by: Dave Helstroom <helstroom@google.com> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: asihpi - fix potential NULL pointer dereferenceWei Yongjun2013-03-121-1/+2
| | | | | | | | | | | | | | The dereference should be moved below the NULL test. Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: seq: Fix missing error handling in snd_seq_timer_open()Takashi Iwai2013-03-111-4/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_seq_timer_open() didn't catch the whole error path but let through if the timer id is a slave. This may lead to Oops by accessing the uninitialized pointer. BUG: unable to handle kernel NULL pointer dereference at 00000000000002ae IP: [<ffffffff819b3477>] snd_seq_timer_open+0xe7/0x130 PGD 785cd067 PUD 76964067 PMD 0 Oops: 0002 [#4] SMP CPU 0 Pid: 4288, comm: trinity-child7 Tainted: G D W 3.9.0-rc1+ #100 Bochs Bochs RIP: 0010:[<ffffffff819b3477>] [<ffffffff819b3477>] snd_seq_timer_open+0xe7/0x130 RSP: 0018:ffff88006ece7d38 EFLAGS: 00010246 RAX: 0000000000000286 RBX: ffff88007851b400 RCX: 0000000000000000 RDX: 000000000000ffff RSI: ffff88006ece7d58 RDI: ffff88006ece7d38 RBP: ffff88006ece7d98 R08: 000000000000000a R09: 000000000000fffe R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: ffff8800792c5400 R14: 0000000000e8f000 R15: 0000000000000007 FS: 00007f7aaa650700(0000) GS:ffff88007f800000(0000) GS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00000000000002ae CR3: 000000006efec000 CR4: 00000000000006f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 Process trinity-child7 (pid: 4288, threadinfo ffff88006ece6000, task ffff880076a8a290) Stack: 0000000000000286 ffffffff828f2be0 ffff88006ece7d58 ffffffff810f354d 65636e6575716573 2065756575712072 ffff8800792c0030 0000000000000000 ffff88006ece7d98 ffff8800792c5400 ffff88007851b400 ffff8800792c5520 Call Trace: [<ffffffff810f354d>] ? trace_hardirqs_on+0xd/0x10 [<ffffffff819b17e9>] snd_seq_queue_timer_open+0x29/0x70 [<ffffffff819ae01a>] snd_seq_ioctl_set_queue_timer+0xda/0x120 [<ffffffff819acb9b>] snd_seq_do_ioctl+0x9b/0xd0 [<ffffffff819acbe0>] snd_seq_ioctl+0x10/0x20 [<ffffffff811b9542>] do_vfs_ioctl+0x522/0x570 [<ffffffff8130a4b3>] ? file_has_perm+0x83/0xa0 [<ffffffff810f354d>] ? trace_hardirqs_on+0xd/0x10 [<ffffffff811b95ed>] sys_ioctl+0x5d/0xa0 [<ffffffff813663fe>] ? trace_hardirqs_on_thunk+0x3a/0x3f [<ffffffff81faed69>] system_call_fastpath+0x16/0x1b Reported-and-tested-by: Tommi Rantala <tt.rantala@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: ice1712: Initialize card->private_data properlySean Connor2013-03-071-0/+2
| | | | | | | | | | | | | | | | | | Set card->private_data in snd_ice1712_create for fixing NULL dereference in snd_ice1712_remove(). Signed-off-by: Sean Connor <sconnor004@allyinics.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Add support of new codec ALC233Kailang Yang2013-03-071-0/+2
| | | | | | | | | | | | | | It's compatible with ALC282. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda/ca0132 - Avoid division by zero in dspxfr_one_seg()Xi Wang2013-03-071-2/+6
| | | | | | | | | | | | | | | | | | | | Move the zero check `hda_frame_size_words == 0' before the modulus `buffer_size_words % hda_frame_size_words'. Also remove the redundant null check `buffer_addx == NULL'. Signed-off-by: Xi Wang <xi.wang@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - check NULL pointer when creating SPDIF PCM switchMengdong Lin2013-03-071-2/+7
| | | | | | | | | | | | | | | | If the new control cannot be created, this function will return to avoid snd_hda_ctl_add dereferencing a NULL control pointer. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - check NULL pointer when creating SPDIF controlsMengdong Lin2013-03-071-0/+2
| | | | | | | | | | | | | | | | If the SPDIF control array cannot be reallocated, the function will return to avoid dereferencing a NULL pointer. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * Merge tag 'asoc-v3.9-rc1' of ↵Takashi Iwai2013-03-076-18/+29
| |\ | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Updates for v3.9 A few driver fixes, none of them terribly dramatic.
| | * Merge remote-tracking branch 'asoc/fix/wm8960' into tmpMark Brown2013-03-071-4/+4
| | |\
| | | * ASoC: wm8960: Fix ADC power bitsMark Brown2013-03-011-2/+2
| | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | * ASoC: wm8960: Correct register 0 and 1 defaultsMark Brown2013-03-011-2/+2
| | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | Merge remote-tracking branch 'asoc/fix/wm8350' into tmpMark Brown2013-03-071-2/+2
| | |\ \
| | | * | ASoC: wm8350: Use jiffies rather than msecs in schedule_delayed_work()Axel Lin2013-03-031-2/+2
| | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The delay parameter of schedule_delayed_work() is number of jiffies to wait rather than miliseconds. Before commit 6d3c26bcb "ASoC: Use delayed work to debounce WM8350 jack IRQs", the debounce time is 200 miliseconds in wm8350_hp_jack_handler(). So I think this is a bug when convert to use delayed work. Signed-off-by: Axel Lin <axel.lin@ingics.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | Merge remote-tracking branch 'asoc/fix/tegra' into tmpMark Brown2013-03-072-2/+2
| | |\ \
| | | * | ASoC: tegra: fix I2S bit count maskStephen Warren2013-03-052-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This register field is 11 bits wide, not 15 bits wide. Given the way this value is currently, used, this patch has no practical effect. However, it's still best if the value is correct. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | Merge remote-tracking branch 'asoc/fix/arizona' into tmpMark Brown2013-03-072-10/+21
| | |\ \ \ | | | |/ / | | |/| |
| | | * | ASoC: wm5102: Apply a SYSCLK patch for later revsMark Brown2013-03-061-0/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Evaluation has identified some performance improvements to the device. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | * | ASoC: wm5110: Correct OUT2/3 volume and switch namesCharles Keepax2013-03-011-8/+8
| | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | * | ASoC: wm5102: Correct OUT2 volume and switch namesCharles Keepax2013-03-011-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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