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* ALSA: asihpi - Increase debug response buffer size.Eliot Blennerhassett2011-12-221-1/+1
| | | | | | | Enables retrieving more debug info in fewer transactions. Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Add new function codes.Eliot Blennerhassett2011-12-221-3/+10
| | | | | Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Remove unused structs and defsEliot Blennerhassett2011-12-221-94/+2
| | | | | | | Structs related to network flash update are not required in kernel. Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Update node types.Eliot Blennerhassett2011-12-222-5/+3
| | | | | | | | Add "Internal" node type. Remove GPI and GPO node types. Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Only set sync if card supports hardware stream grouping.Eliot Blennerhassett2011-12-221-3/+3
| | | | | Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Relax drained check for more reliable playback startup.Eliot Blennerhassett2011-12-221-3/+2
| | | | | Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Correct stray capital letters in identifier.Eliot Blennerhassett2011-12-222-8/+8
| | | | | Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Use snd_pcm_debug_name to get substream name.Eliot Blennerhassett2011-12-221-2/+6
| | | | | Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Volumes and meters may have 1 or 2 channels.Eliot Blennerhassett2011-12-221-2/+14
| | | | | | | The channel count can be queried to determine which. Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Low latency mode stream has fixed channel count.Eliot Blennerhassett2011-12-221-4/+13
| | | | | | | Unlike other streams which support 1..max channels, Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Update copyright to 2011Eliot Blennerhassett2011-12-2214-14/+14
| | | | | Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Split hpi version info into separate header file.Eliot Blennerhassett2011-12-226-34/+47
| | | | | | | and update HPI version to 4.10 Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - fix pcm dma pointer trackingEliot Blennerhassett2011-12-221-2/+2
| | | | | | | | Elapsed counter should only count data committed to snd_pcm_period_elapsed, rather than all data available Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: Add missing module parameters for als300 and cs5530 driversTakashi Iwai2011-12-192-2/+16
| | | | | | | These drviers defined only variables but didn't declare as module parameters. Also fix the enable variable to bool type. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: module_param: make bool parameters really boolRusty Russell2011-12-19108-171/+171
| | | | | | | | | | | | module_param(bool) used to counter-intuitively take an int. In fddd5201 (mid-2009) we allowed bool or int/unsigned int using a messy trick. It's time to remove the int/unsigned int option. For this version it'll simply give a warning, but it'll break next kernel version. Signed-off-by: Rusty Russell <rusty@rustcorp.com.au> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: added VOX ToneLab ST midi handlingSergiusz Urbaniak2011-12-121-0/+10
| | | | | Signed-off-by: Sergiusz Urbaniak <sergiusz.urbaniak@googlemail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihp: Use kcalloc instead of kzalloc to allocate arrayThomas Meyer2011-12-061-1/+1
| | | | | | | | | | | | The advantage of kcalloc is, that will prevent integer overflows which could result from the multiplication of number of elements and size and it is also a bit nicer to read. The semantic patch that makes this change is available in https://lkml.org/lkml/2011/11/25/107 Signed-off-by: Thomas Meyer <thomas@m3y3r.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: ctxf: Use kcalloc instead of kzalloc to allocate arrayThomas Meyer2011-12-061-1/+1
| | | | | | | | | | | | The advantage of kcalloc is, that will prevent integer overflows which could result from the multiplication of number of elements and size and it is also a bit nicer to read. The semantic patch that makes this change is available in https://lkml.org/lkml/2011/11/25/107 Signed-off-by: Thomas Meyer <thomas@m3y3r.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: sis7019 - convert to dev_*() loggingDavid Dillow2011-12-021-15/+14
| | | | | Signed-off-by: David Dillow <dave@thedillows.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'fix/misc' into topic/miscTakashi Iwai2011-12-0247-828/+642
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| * ALSA: sis7019 - give slow codecs more time to resetDavid Dillow2011-12-021-11/+53
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There are some AC97 codec and board combinations that have been observed to take a very long time to respond after the cold reset has completed. In one case, more than 350 ms was required. To allow users to have sound on those platforms, we'll wait up to 500ms for the codec to become ready. As a board may have multiple codecs, with some faster than others to reset, we add a module parameter to inform the driver which codecs should be present. Reported-by: KotCzarny <tjosko@yahoo.com> Signed-off-by: David Dillow <dave@thedillows.org> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * Merge branch 'fix/asoc' into for-linusTakashi Iwai2011-12-0112-522/+40
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| | * ASoC: Supply dcs_codes for newer WM1811 revisionsMark Brown2011-11-291-0/+2
| | | | | | | | | | | | | | | | | | Based on initial data. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Error out if we can't generate a LRCLK at all for WM8994Mark Brown2011-11-281-0/+5
| | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Correct name of Speyside Main Speaker widgetMark Brown2011-11-271-1/+1
| | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: skip resume of soc-audio devices without codecsEric Miao2011-11-231-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There are cases where there is no working codec on the soc-audio devices, and snd_soc_suspend() will skip such device when suspending. Yet its counterpart snd_soc_resume() does not check this, causing complaints about spinlock lockup: [ 176.726087] BUG: spinlock lockup on CPU#0, kworker/0:2/1067, d8ab82a8 [ 176.732539] [<80014a14>] (unwind_backtrace+0x0/0xec) from [<805b3fc8>] (dump_stack+0x20/0x24) [ 176.741082] [<805b3fc8>] (dump_stack+0x20/0x24) from [<80322208>] (do_raw_spin_lock+0x118/0x158) [ 176.749882] [<80322208>] (do_raw_spin_lock+0x118/0x158) from [<805b7874>] (_raw_spin_lock_irqsave+0x5c/0x68) [ 176.759723] [<805b7874>] (_raw_spin_lock_irqsave+0x5c/0x68) from [<8002a020>] (__wake_up+0x2c/0x5c) [ 176.768781] [<8002a020>] (__wake_up+0x2c/0x5c) from [<804a6de8>] (soc_resume_deferred+0x3c/0x2b0) [ 176.777666] [<804a6de8>] (soc_resume_deferred+0x3c/0x2b0) from [<8004ee20>] (process_one_work+0x2e8/0x50c) [ 176.787334] [<8004ee20>] (process_one_work+0x2e8/0x50c) from [<8004fd08>] (worker_thread+0x1c8/0x2e0) [ 176.796566] [<8004fd08>] (worker_thread+0x1c8/0x2e0) from [<80053ec8>] (kthread+0xa4/0xb0) [ 176.804843] [<80053ec8>] (kthread+0xa4/0xb0) from [<8000ea70>] (kernel_thread_exit+0x0/0x8) Signed-off-by: Eric Miao <eric.miao@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: cs42l51: Fix off-by-one for reg_cache_sizeAxel Lin2011-11-231-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Just checking the code in cs42l51_fill_cache(): The cache pointer points to codec->reg_cache + 1. I think it is because CS42L51_FIRSTREG is 0x01, so codec->reg_cache[0] is not used here. Then we read CS42L51_NUMREGS bytes to cache. So we need reg_cache_size to be CS42L51_NUMREGS + 1. Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: drop support for PlayPaq with WM8510Paul Bolle2011-11-233-497/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | SoC Audio support for PlayPaq with WM8510 got added in commit 9aaca9683b ("[ALSA] Revised AT32 ASoC Patch"). That support depends on BOARD_PLAYPAQ. That Kconfig symbol didn't exist when that support got added in v2.6.27. It still doesn't. It has never been possible to even build this driver. Drop it. Signed-off-by: Paul Bolle <pebolle@tiscali.nl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: mpc8610: tell the CS4270 codec that it's the masterTimur Tabi2011-11-231-8/+16
| | | | | | | | | | | | | | | | | | | | | | | | | | | Commit ac601555 ("ASoC: Return early with -EINVAL if invalid dai format is detected") requires the machine driver to tell the CS4270 codec driver whether the CS4270 should be configured for master or slave operation. Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: cs4720: use snd_soc_cache_sync()Daniel Mack2011-11-231-9/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Replace the manual register restore mechanism in cs4270.c and call the generic snd_soc_cache_sync() handler instead. This factors code out in favour of core facilities and also fixes a bus confusion that is most probably caused by intermixing i2c-regmap functions and i2c_smbus_* accessors. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: Sven Neumann <s.neumann@raumfeld.com> Acked-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: SAMSUNG: Fix build errorBoojin Kim2011-11-221-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds <linux/modules.h> to fix following build errors. sound/soc/codecs/wm8994.c: In function 'wm8994_readable': sound/soc/codecs/wm8994.c:58: warning: unused variable 'wm8994' sound/soc/samsung/smdk_wm8994.c:176: error: expected declaration specifiers or '...' before string constant sound/soc/samsung/smdk_wm8994.c:176: warning: data definition has no type or storage class sound/soc/samsung/smdk_wm8994.c:176: warning: type defaults to 'int' in declaration of 'MODULE_DESCRIPTION' sound/soc/samsung/smdk_wm8994.c:176: warning: function declaration isn't a prototype sound/soc/samsung/smdk_wm8994.c:177: error: expected declaration specifiers or '...' before string constant Signed-off-by: Boojin Kim <boojin.kim@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: max9877: Update register if either val or val2 is changedAxel Lin2011-11-221-5/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the case of ((max9877_regs[reg] >> shift) & mask) != val but ((max9877_regs[reg2] >> shift) & mask) == val2, current code does not update the registers. Fix the logic to update registers if either val or val2 is changed. Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Fix wrong define for AD1836_ADC_WORD_OFFSETAxel Lin2011-11-221-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | According to the datasheet: The BIT[5:4] of ADC Control Register 2 is to control the word width. 00 = 25 Bits 01 = 20 Bits 10 = 16 Bits 11 = Invalid Thus, the AD1836_ADC_WORD_OFFSET should be defined as 4. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * | ALSA: hda - Fix S3/S4 problem on machines with VREF-pin mute-LEDCharles Chin2011-12-011-18/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The verb command in stac92xx_post_suspend caused the audio to stop working after resuming from S3 mode on HP laptops with the VREF-pin mute-LED control. Removing relevant post_suspend registering. Although removing D3 on AFG is no optimal solution, the impact should be small in comparison with the broken S3/S4. Signed-off-by: Charles Chin <Charles.Chin@idt.com> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda_intel - revert a quirk that affect VIA chipsetsMarc Vertes2011-11-291-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This quirk sould be reverted. It has the following probems: 1) The quirk was intended to "ASUS MV2-MX SE" motherboards only, but the ID used matches a much broader range, potentially all boards containing a VIA chipset model in the family of vendor VIA 0x1106 and audio device ID 0x3288, which encompasses VIA-VT82xx, VIA-VT1xx and VIA-VT20xx chipsets. 2) VIA chipsets rely on azx_via_get_position() to handle correctly dma transfers during capture. Using POS_FIX_LPIB instead of POS_FIX_VIACOMBO leads to partially corrupted input buffers during capture. The effects of this bug are not immediately visible, it took strong DSP expertise, some expensive signal generator and a spectrum analyzer to identify it and verify correct behaviour using original default. 3) It's almost certain that the quirk did not fix the real problem, if there was one. Refer to original submission: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-February/025109.html Signed-of-by: Marc Vertes <mvertes@sigfox.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - Avoid touching mute-VREF pin for IDT codecsTakashi Iwai2011-11-291-1/+3
| | | | | | | | | | | | | | | | | | | | | Some HP laptops use a pin VREF for controlling the mute LED, and such a pin shouldn't be powered off. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usb-audio - Support for Roland GAIA SH-01 SynthesizerJohn F Leach2011-11-291-0/+31
| | | | | | | | | | | | | | | | | | | | | | | | | | | Added table quirks entry for Roland GAIA SH-01 Synthesizer based upon Roland SH-201 table entry as template. USB MIDI and audio was tested with Muse and Audacity. Signed-off-by: John F Leach <jfleach@jfleach.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge branch 'fix/hda' into for-linusTakashi Iwai2011-11-273-46/+56
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| | * | ALSA: hda - Fix jack-detection control of VT1708Takashi Iwai2011-11-271-33/+43
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | VT1708 has no support for unsolicited events per jack-plug, the driver implements the workq for polling the jack-detection. The mixer element "Jack Detect" was supposed to control this behavior on/off, but this doesn't work properly as is now. The workq is always started and the HP automute is always enabled. This patch fixes the jack-detect control behavior by triggering / stopping the work appropriately at the state change. Also the work checks the internal state to continue scheduling or not. Cc: <stable@kernel.org> [v3.1] Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | ALSA: hda - cut and paste typo in cs420x_models[]Dan Carpenter2011-11-271-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The CS420X_IMAC27 was copied from the line before but CS420X_APPLE was clearly intented. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | ALSA: hda/realtek - Minor cleanupTakashi Iwai2011-11-231-12/+12
| | | | | | | | | | | | | | | | | | | | | | | | Use an inline function for the common pattern for assigning a capsrc. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | Merge branch 'fix/asoc' into for-linusTakashi Iwai2011-11-2314-19/+88
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| | * ASoC: Ensure WM8731 register cache is synced when resuming from disabledMark Brown2011-11-211-0/+1
| | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| | * Merge branch 'DB_RANGE-size-fixes' of ↵Mark Brown2011-11-217-10/+10
| | |\ | | | | | | | | | | | | git://git.alsa-project.org/alsa-kprivate into for-3.2
| | | * ASoC: wm_hubs: fix DB_RANGE sizeClemens Ladisch2011-11-201-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent reading more data than actually is in the array. Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| | | * ASoC: wm9090: fix DB_RANGE sizeClemens Ladisch2011-11-201-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent reading more data than actually is in the arrays. Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| | | * ASoC: wm8993: fix DB_RANGE sizeClemens Ladisch2011-11-201-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent reading more data than actually is in the array. Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| | | * ASoC: wm8962: fix DB_RANGE sizeClemens Ladisch2011-11-201-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent reading more data than actually is in the arrays. Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| | | * ASoC: sgtl5000: fix DB_RANGE sizeClemens Ladisch2011-11-201-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent reading more data than actually is in the array. Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| | | * ASoC: rt5631: fix DB_RANGE sizeClemens Ladisch2011-11-201-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent the last entry from being omitted. Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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