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2013-12-04ALSA: hda - Fix silent output on MacBook Air 2,1Takashi Iwai1-8/+31
MacBook Air 2,1 has a fairly different pin assignment from its brother MBA 1,1, and yet another quirks are needed for pin 0x18 and 0x19, similarly like what iMac 9,1 requires, in order to make the sound working on it. Reported-and-tested-by: Bruno Prémont <bonbons@linux-vserver.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-12-04Merge tag 'asoc-v3.13-rc2' of ↵Takashi Iwai11-49/+40
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v3.13 A smattering of fixes here, some core ones for the rate combination issues for things other than simple bitmasks, for readback of byte controls and for updating the power of value muxes plus a bunch of driver fixes of varying severity. The warning fix in the i.MX FIQ driver is fixing a warning introduced by a previous fix.
2013-12-04ALSA: hda - Fix missing ELD info when using jackpoll_ms parameterDavid Henningsson1-12/+15
In the case of using jackpoll_ms instead of unsol events, the jack was correctly detected, but ELD info was not refreshed on plug-in. And without ELD info, no proper restriction of pcm, which can in turn break sound output on some devices. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-12-04ALSA: hda/realtek - remove hp_automute_hook from alc283_fixup_chromebookKailang Yang1-1/+0
I forgot to remove the hp_automute_hook from alc283_fixup_chromebook. It doesn't need this for other chrome os machine. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-12-03Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/atmel', ↵Mark Brown8-39/+24
'asoc/fix/fsl', 'asoc/fix/kirkwood', 'asoc/fix/omap', 'asoc/fix/rcar', 'asoc/fix/wm8731' and 'asoc/fix/wm8990' into asoc-linus
2013-12-03Merge remote-tracking branch 'asoc/fix/core' into asoc-linusMark Brown3-10/+16
2013-12-03ASoC: wm8731: fix dsp mode configurationBo Shen1-2/+2
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> Cc: stable@vger.kernel.org
2013-12-03ALSA: hda/realtek - Independent of model for HPKailang Yang1-10/+28
Create single model for HP. The headset jack module was difference between other chrome book. It need to manual control Mic jack detect. Chrome OS loaded driver by models. Remove old assigned fixup table from ALC269 fixup list entry. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-12-03ALSA: hda - Fix headset mic input after muted internal mic (Dell/Realtek)David Henningsson1-0/+2
By trial and error, I found this patch could work around an issue where the headset mic would stop working if you switch between the internal mic and the headset mic, and the internal mic was muted. It still takes a second or two before the headset mic actually starts working, but still better than nothing. Information update from Kailang: The verb was ADC digital mute(bit 6 default 1). Switch internal mic and headset mic will run alc_headset_mode_default. The coef index 0x11 will set to 0x0041. Because headset mode was fixed type. It doesn't need to run alc_determine_headset_type. So, the value still keep 0x0041. ADC was muted. BugLink: https://bugs.launchpad.net/bugs/1256840 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-12-02ALSA: hda - Use always amps for auto-mute on AD1986A codecTakashi Iwai1-0/+3
It seems that AD1986A cannot manage the dynamic pin on/off for auto-muting, but rather gets confused. Since each output has own amp, let's use it instead. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971 Cc: <stable@vger.kernel.org> [v3.11+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-12-02ALSA: hda/analog - Handle inverted EAPD properly in vmaster hookTakashi Iwai1-0/+2
ad_vmaster_eapd_hook() needs to handle the inverted EAPD case properly, too. Otherwise the output gets broken on Lenovo N100 with AD1986A codec. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971 Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-12-02ALSA: hda - Another fixup for ASUS laptop with ALC660 codecTakashi Iwai1-0/+1
ASUS Z35HL laptop also needs the very same fix as the previous one that was applied to ASUS W7J. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66231 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-12-02ALSA: atmel: Fix possible array overflowTakashi Iwai1-1/+2
The static checker found a possible array overflow in atmel/abdac.c: static checker warning: "sound/atmel/abdac.c:373 set_sample_rates() error: buffer overflow 'dac->rates' 6 <= 6" This patch papers over the buggy point, by ensuring that dac->rates[] update not overflowing the actual array size. Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-12-02ALSA: hda - Fix complete_all() timing in deferred probesTakashi Iwai1-4/+5
When the probe of snd-hda-intel driver is deferred due to f/w loading or the nested module loading, complete_all() should be also delayed until the initialization really finished. Otherwise, vga-switcheroo client would start switching before the actual init is done. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-12-02ALSA: hda - Fix bad EAPD setup for HP machines with AD1984ATakashi Iwai1-0/+1
It seems that EAPD on NID 0x16 is the only control over all outputs on HP machines with AD1984A while turning EAPD on NID 0x12 breaks the output. Thus we need to avoid fiddling EAPD on NID. As a quick workaround, just set own_eapd_ctrl flag for the wrong EAPD, then implement finer EAPD controls. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66321 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-12-02ASoC: core: fix devres parameter in devm_snd_soc_register_card()Shawn Guo1-2/+2
Since devm_card_release() expects parameter 'res' to be a pointer to struct snd_soc_card, devm_snd_soc_register_card() should really pass such a pointer rather than the one to struct device. This bug causes the kernel Oops below with imx-sgtl500 driver when we remove the module. It happens because with 'card' pointing to the wrong structure, card->num_rtd becomes 0 in function soc_remove_dai_links(). Consequently, soc_remove_link_components() and in turn soc_cleanup_codec[platform]_debugfs() will not be called on card removal. It results in that debugfs_card_root is being removed while its child entries debugfs_codec_root and debugfs_platform_root are still there, and thus the kernel Oops. Fix the bug by correcting the parameter 'res' to be the pointer to struct snd_soc_card. $ lsmod Module Size Used by snd_soc_imx_sgtl5000 3506 0 snd_soc_sgtl5000 13677 2 snd_soc_imx_audmux 5324 1 snd_soc_imx_sgtl5000 snd_soc_fsl_ssi 8139 2 imx_pcm_dma 1380 1 snd_soc_fsl_ssi $ rmmod snd_soc_imx_sgtl5000 Unable to handle kernel paging request at virtual address e594025c pgd = be134000 [e594025c] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: snd_soc_imx_sgtl5000(-) snd_soc_sgtl5000 snd_soc_imx_audmux snd_soc_fsl_ssi imx_pcm_dma CPU: 0 PID: 1793 Comm: rmmod Not tainted 3.13.0-rc1 #1570 task: bee28900 ti: bfbec000 task.ti: bfbec000 PC is at debugfs_remove_recursive+0x28/0x154 LR is at snd_soc_unregister_card+0xa0/0xcc pc : [<80252b38>] lr : [<80496ac4>] psr: a0000013 sp : bfbede00 ip : bfbede28 fp : bfbede24 r10: 803281d4 r9 : bfbec000 r8 : 803271ac r7 : bef54440 r6 : 00000004 r5 : bf9a4010 r4 : bf9a4010 r3 : e5940224 r2 : 00000000 r1 : bef54450 r0 : 803271ac Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c53c7d Table: 4e13404a DAC: 00000015 Process rmmod (pid: 1793, stack limit = 0xbfbec240) Stack: (0xbfbede00 to 0xbfbee000) de00: 00000000 bf9a4010 bf9a4010 00000004 bef54440 bec89000 bfbede44 bfbede28 de20: 80496ac4 80252b1c 804a4b60 bfbede60 bf9a4010 00000004 bfbede54 bfbede48 de40: 804a4b74 80496a30 bfbede94 bfbede58 80328728 804a4b6c bfbede94 a0000013 de60: bf1b5800 bef54440 00000002 bf9a4010 7f0169f8 bf9a4044 00000081 8000e9c4 de80: bfbec000 00000000 bfbedeac bfbede98 80328cb0 80328618 7f016000 bf9a4010 dea0: bfbedec4 bfbedeb0 8032561c 80328c84 bf9a4010 7f0169f8 bfbedee4 bfbedec8 dec0: 80325e84 803255a8 bee28900 7f0169f8 00000000 78208d30 bfbedefc bfbedee8 dee0: 80325410 80325dd4 beca8100 7f0169f8 bfbedf14 bfbedf00 803264f8 803253c8 df00: 7f01635c 7f016a3c bfbedf24 bfbedf18 80327098 803264d4 bfbedf34 bfbedf28 df20: 7f016370 80327090 bfbedfa4 bfbedf38 80085ef0 7f016368 bfbedf54 5f646e73 df40: 5f636f73 5f786d69 6c746773 30303035 00000000 78208008 bfbedf84 bfbedf68 df60: 800613b0 80061194 fffffffe 78208d00 7efc2f07 00000081 7f016a3c 00000800 df80: bfbedf84 00000000 00000000 fffffffe 78208d00 7efc2f07 00000000 bfbedfa8 dfa0: 8000e800 80085dcc fffffffe 78208d00 78208d30 00000800 a8c82400 a8c82400 dfc0: fffffffe 78208d00 7efc2f07 00000081 00000002 00000000 78208008 00000800 dfe0: 7efc2e1c 7efc2ba8 76f5ca47 76edec7c 80000010 78208d30 00000000 00000000 Backtrace: [<80252b10>] (debugfs_remove_recursive+0x0/0x154) from [<80496ac4>] (snd_soc_unregister_card+0xa0/0xcc) r8:bec89000 r7:bef54440 r6:00000004 r5:bf9a4010 r4:bf9a4010 r3:00000000 [<80496a24>] (snd_soc_unregister_card+0x0/0xcc) from [<804a4b74>] (devm_card_release+0x14/0x18) r6:00000004 r5:bf9a4010 r4:bfbede60 r3:804a4b60 [<804a4b60>] (devm_card_release+0x0/0x18) from [<80328728>] (release_nodes+0x11c/0x1dc) [<8032860c>] (release_nodes+0x0/0x1dc) from [<80328cb0>] (devres_release_all+0x38/0x54) [<80328c78>] (devres_release_all+0x0/0x54) from [<8032561c>] (__device_release_driver+0x80/0xd4) r4:bf9a4010 r3:7f016000 [<8032559c>] (__device_release_driver+0x0/0xd4) from [<80325e84>] (driver_detach+0xbc/0xc0) r5:7f0169f8 r4:bf9a4010 [<80325dc8>] (driver_detach+0x0/0xc0) from [<80325410>] (bus_remove_driver+0x54/0x98) r6:78208d30 r5:00000000 r4:7f0169f8 r3:bee28900 [<803253bc>] (bus_remove_driver+0x0/0x98) from [<803264f8>] (driver_unregister+0x30/0x50) r4:7f0169f8 r3:beca8100 [<803264c8>] (driver_unregister+0x0/0x50) from [<80327098>] (platform_driver_unregister+0x14/0x18) r4:7f016a3c r3:7f01635c [<80327084>] (platform_driver_unregister+0x0/0x18) from [<7f016370>] (imx_sgtl5000_driver_exit+0x14/0x1c [snd_soc_imx_sgtl5000]) [<7f01635c>] (imx_sgtl5000_driver_exit+0x0/0x1c [snd_soc_imx_sgtl5000]) from [<80085ef0>] (SyS_delete_module+0x130/0x18c) [<80085dc0>] (SyS_delete_module+0x0/0x18c) from [<8000e800>] (ret_fast_syscall+0x0/0x48) r6:7efc2f07 r5:78208d00 r4:fffffffe Code: 889da9f8 e5983020 e3530000 089da9f8 (e5933038) ---[ end trace 825e7e125251a225 ]--- Signed-off-by: Shawn Guo <shawn.guo@linaro.org> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-12-02ASoC: omap: n810: Convert to clk_prepare_enable/clk_disable_unprepareJarkko Nikula1-2/+2
N810 audio driver has stopped working at some point. Probably when OMAP2 was converted to common clock framework since now call to clk_enable dumps the stack trace in drivers/clk/clk.c: __clk_enable() due clk->prepare_count is zero. Fix this by converting clk_enable/_disable calls to those that take care of clock prepare/unprepare. I'm not queueing this to linux-stable since OMAP2 common clock framework conversion in commit ed1ebc4948fd ("ARM: OMAP2: clock: Convert to common clk") happened before N810 was really usable in mainline and user base for N810 is anyway small. Potential linux-stable candidates are only those after commit 3d3a6d18abc6 ("watchdog: introduce retu_wdt driver"). Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-12-02ASoC: fsl: set correct platform drvdata in pcm030_fabric_probe()Wei Yongjun1-1/+2
platform_set_drvdata(op, pdata) in pcm030_fabric_probe() will be overwrited when calling snd_soc_register_card(card), but cm030_fabric_remove() use drvdata as a type of struct pcm030_audio_data, so we should move platform_set_drvdata() below snd_soc_register_card() call. Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-12-02ASoC: fsl: imx-pcm-fiq: Remove unused 'runtime' variableFabio Estevam1-1/+0
Commit 68f9672b (ASoC: fsl: imx-pcm-fiq: remove bogus period delta calculation) introduced the following build warning: sound/soc/fsl/imx-pcm-fiq.c:53:26: warning: unused variable 'runtime' [-Wunused-variable] Remove the unused 'runtime' variable. Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Acked-by: Oskar Schirmer <oskar@scara.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-12-02ASoC: fsl: imx-pcm-fiq: remove bogus period delta calculationOskar Schirmer1-19/+2
Originally snd_hrtimer_callback() used iprtd->period_time for some jiffies based estimation to determine the right moment to call snd_pcm_period_elapsed(). As timer drifts may well be a problem, this was changed in commit b4e82b5b785670b6 to be based on buffer transmission progress, using iprtd->offset and runtime->buffer_size to calculate the amount of data since last period had elapsed. Unfortunately, iprtd->offset counts in bytes, while runtime->buffer_size counts frames, so adding these to find some delta is like comparing apples and oranges, and eventually results in negative delta values every now and then. This is no big harm, because it simply causes snd_pcm_period_elapsed() being called more often than necessary, as negative delta is taken for a large unsigned value by implicit conversion rule. Nonetheless, the calculation is broken, so one would replace the runtime->buffer_size by its equivalent in bytes. But then, there are chances snd_pcm_period_elapsed() is called late, because calculating the moment for the elapsed period into delta is based against the iprtd->last_offset, which is not necessarily the first byte of the period in question, but some random byte which the FIQ handler left us with in r8/r9 by accident. Again, negative impact is low, as there are plenty of periods already prefilled with data, and snd_pcm_period_elapsed() will probably be called latest when the following period is reached. However, the calculation is conceptually broken, and we are best off removing the clever stuff altogether. snd_pcm_period_elapsed() is now simply called once everytime snd_hrtimer_callback() is run, which may not be most accurate, but at least this way we are quite sure we dont miss an end of period. There is not much extra effort wasted by superfluous calls to snd_pcm_period_elapsed(), as the timer frequency closely matches the period size anyway. Signed-off-by: Oskar Schirmer <oskar@scara.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-29ALSA: hda - Fix silent output on ASUS W7J laptopTakashi Iwai1-0/+12
The recent kernels got regressions on ASUS W7J with ALC660 codec where no sound comes out. After a long debugging session, we found out that setting the pin control on the unused NID 0x10 is mandatory for the outputs. And, it was found out that another magic of NID 0x0f that is required for other ASUS laptops isn't needed on this machine. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66081 Reported-and-tested-by: Andrey Lipaev <lipaev@mail.ru> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-29ASoC: core: Use consistent byte ordering in snd_soc_bytes_getCharles Keepax1-2/+2
snd_soc_bytes_put treats the data in the binary control as big endian words, however snd_soc_bytes_get uses the endian of the host machine. This causes the two functions to be inconsistant with how the mask is applied on little endian machines. This patch applies the big_endian format used in snd_soc_bytes_put to snd_soc_bytes_get. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-29ALSA: dice: fix array limits in dice_proc_read()Dan Carpenter1-2/+2
The array limits are supposed to be in units of u32 instead of in bytes. The current code has a potential array overflow. Fixes: c614475b0ea9 ('ALSA: dice: add a proc file to show device information') Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-29ALSA: hda - Add mono speaker quirk for Dell Inspiron 5439David Henningsson1-0/+1
This machine also has mono output if run through DAC node 0x03. Cc: stable@vger.kernel.org (v3.10+) BugLink: https://bugs.launchpad.net/bugs/1256212 Tested-by: David Chen <david.chen@canonical.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-28ALSA: hda - Add LFE chmap to ASUS ET2700Takashi Iwai1-3/+13
As the previous commit 1f0bbf03cb82 added the pin config for the bass speaker, this patch adds the corresponding LFE-only channel map on ASUS ET2700. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961 Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-28ALSA: hda - Initialize missing bass speaker pin for ASUS AIO ET2700Takashi Iwai1-0/+9
Add a fixup entry for the missing bass speaker pin 0x16 on ASUS ET2700 AiO desktop. The channel map will be added in the next patch, so that this can be backported easily to stable kernels. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-28ALSA: hda - limit mic boost on Asus UX31[A,E]Oleksij Rempel1-2/+9
This both devices need limit for internal dmic. [cosmetic change; renamed fixup name by tiwai] Signed-off-by: Oleksij Rempel <linux@rempel-privat.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-28ALSA: hda - Check leaf nodes to find aamix ampsTakashi Iwai1-12/+45
The current generic parser assumes blindly that the volume and mute amps are found in the aamix node itself. But on some codecs, typically Analog Devices ones, the aamix amps are separately implemented in each leaf node of the aamix node, and the current driver can't establish the correct amp controls. This is a regression compared with the previous static quirks. This patch extends the search for the amps to the leaf nodes for allowing the aamix controls again on such codecs. In this implementation, I didn't code to loop through the whole paths, since usually one depth should suffice, and we can't search too deeply, as it may result in the conflicting control assignments. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65641 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-27ASoC: kirkwood: Fix erroneous double output while playingJean-Francois Moine1-5/+5
This patch fixes the setting of the register KIRKWOOD_PLAYCTL which did always streaming on both I2S and SPDIF, ignoring the DAI ID. The bug was introduced by the commit 75b9b65ee5a "ASoC: kirkwood: add S/PDIF support" Signed-off-by: Jean-Francois Moine <moinejf@free.fr> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-27ASoC: kirkwood: Fix invalid S/PDIF formatJean-Francois Moine1-4/+8
This patch removes the 32 bits format which is not supported by S/PDIF output. Signed-off-by: Jean-Francois Moine <moinejf@free.fr> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-27ASoC: pcm: Always honor DAI min and max sample rate constraintsLars-Peter Clausen1-6/+12
snd_pcm_limit_hw_rates() will initialize the minimum and maximum sample rate for the PCM stream based on the rates specified in the rates field. Since we call snd_pcm_limit_hw_rates() after soc_pcm_init_runtime_hw() it will essentially overwrite the min and max rate set in soc_pcm_init_runtime_hw(). This may cause the minimum or maximum rate to be set to a value outside the range of one of the components if one of the components sets either SNDRV_PCM_RATE_CONTINUOUS or SNDRV_PCM_RATE_KNOT and the other component specified a discrete rate via SNDRV_PCM_RATE_[0-9]* that is outside of the first component's rate range. To fix this first calculate the minimum and maximum rates using snd_pcm_limit_hw_rates() and then on top of that apply the contraints specified in the snd_soc_pcm_stream structs. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Takashi iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-27ASoC: pcm: Fix rate_max calculationLars-Peter Clausen1-1/+1
In order to make sure that the sample rate is in the supported range of both components the maximum rate of the card should be the minimum of the maximum rate of each components. There is one special case to consider though, if max_rate is set to 0 this means there is no maximum specified, so use min_not_zero() macro which will give use the desired result. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Takashi iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-27ASoC: wm5110: Remove output OSR and PGA volume controlsCharles Keepax1-25/+0
These are managed automatically in current revisions. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-27ASoC: atmel: sam9x5_wm8731: fix oops when unload moduleBo Shen1-0/+2
As the priv is not assigned to card->drvdata, it is NULL, so when unload module, it will cause NULL pointer oops. Assign priv to card->drvdata to fix this issue. Signed-off-by: Bo Shen <voice.shen@atmel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-27ALSA: hda - Fix hp-mic mode without VREF bitsTakashi Iwai1-1/+1
When the hp mic pin has no VREF bits, the driver forgot to set PIN_IN bit. Spotted during debugging old MacBook Airs. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65681 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-27ALSA: hda - Create Headhpone Mic Jack Mode when really neededTakashi Iwai1-8/+12
When a headphone jack is configurable as input, the generic parser tries to make it retaskable as Headphone Mic. The switching can be done smoothly if Capture Source control exists (i.e. there is another input source). Or when user explicitly enables the creation of jack mode controls, "Headhpone Mic Jack Mode" will be created accordingly. However, if the headphone mic is the only input source, we have to create "Headphone Mic Jack Mode" control because there is no capture source selection. Otherwise, the generic parser assumes that the input is constantly enabled, thus the headphone is permanently set as input. This situation happens on the old MacBook Airs where no input is supported properly, for example. This patch fixes the problem: now "Headphone Mic Jack Mode" is created when such an input selection isn't possible. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65681 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-27ALSA: usb: use multiple packets per urb for Wireless USB inbound audioThomas Pugliese1-1/+15
For Wireless USB audio devices, use multiple isoc packets per URB for inbound endpoints with a datainterval < 5. This allows the WUSB host controller to take advantage of bursting to service endpoints whose logical polling interval is less than the 4ms minimum polling interval limit in WUSB. Signed-off-by: Thomas Pugliese <thomas.pugliese@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-27ALSA: hda - Enable mute/mic-mute LEDs for more Thinkpads with Conexant codecHui Wang1-0/+23
Most Thinkpad Edge series laptops use conexant codec, so far although the codecs have different minor Vendor Id and minor Subsystem Id, they all belong to the cxt5066 family, this change can make the mute/mic-mute LEDs support more generic among cxt_5066 family. This design refers to the similar solution for the realtek codec ALC269 family in the patch_realtek.c. Cc: Alex Hung <alex.hung@canonical.com> Signed-off-by: Hui Wang <hui.wang@canonical.com> Acked-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-26Merge branch 'fix/firewire' into for-linusTakashi Iwai1-9/+6
2013-11-26ALSA: hda - Drop bus->avoid_link_reset flagTakashi Iwai3-4/+3
Use bus->power_keep_link_on instead. The controller shouldn't go to D3 when the link isn't reset, so essentially avoiding the link reset means avoiding the runtime PM. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-26ALSA: hda/realtek - Set pcbeep amp for ALC668Kailang Yang1-0/+1
Set the missing pcbeep default amp for ALC668. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-26ALSA: hda/realtek - Add support of ALC231 codecKailang Yang1-0/+1
It's compatible with ALC269. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-25ASoC: wm8990: Mark the register map as dirty when powering downMark Brown1-0/+2
Otherwise we'll skip sync on resume. Signed-off-by: Mark Brown <broonie@linaro.org> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
2013-11-24ASoC: rcar: select REGMAPKuninori Morimoto1-0/+1
55e5b6fd5af04b6d8b0ac6635edf49476ff298ba (ASoC: rsnd: use regmap instead of original register mapping method) support regmap/regmap_field on Renesas sound driver. It needs CONFIG_REGMAP now. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-22Merge tag 'sound-fix2-3.13-rc1' of ↵Linus Torvalds9-47/+161
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull second set of sound fixes from Takashi Iwai: "A collection of small fixes in HD-audio quirks and runtime PM, ASoC rcar, abs8500 and other codecs. Most of commits are for stable kernels, too" * tag 'sound-fix2-3.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: hda - Set current_headset_type to ALC_HEADSET_TYPE_ENUM (janitorial) ALSA: hda - Provide missing pin configs for VAIO with ALC260 ALSA: hda - Add headset quirk for Dell Inspiron 3135 ALSA: hda - Fix the headphone jack detection on Sony VAIO TX ALSA: hda - Fix missing bass speaker on ASUS N550 ALSA: hda - Fix unbalanced runtime PM notification at resume ASoC: arizona: Set FLL to free-run before disabling ALSA: hda - A casual Dell Headset quirk ASoC: rcar: fixup dma_async_issue_pending() timing ASoC: rcar: off by one in rsnd_scu_set_route() ASoC: wm5110: Add post SYSCLK register patch for rev D chip ASoC: ab8500: Revert to using custom I/O functions ALSA: hda - Also enable mute/micmute LED control for "Lenovo dock" fixup ALSA: firewire-lib: include sound/asound.h to refer to snd_pcm_format_t ALSA: hda - Select FW_LOADER from CONFIG_SND_HDA_CODEC_CA0132_DSP ALSA: hda - Enable mute/mic-mute LEDs for more Thinkpads with Realtek codec ASoC: rcar: fixup mod access before checking
2013-11-22ALSA: firewire-lib: fix wrong value for FDF field as an empty packetTakashi Sakamoto1-9/+6
This commit fix out of specification about the value of FDF field in out packet with 'no data'. This affects blocking mode. According to IEC 61883-6, there is two way to generate AMDTP packets include no data in blocking mode. Way 1. an empty packet defined in IEC 61883-1 - Size of packet is 2 quadlets. - The value of FDF is sfc. - The packet includes only CIP headers Way 2. a special non-empty packet defined in IEC 61883-6 - Size of packet is following to blocking mode - The value of FDF is 0xff. This value is 'NO-DATA'. This means 'The receiver' must ignore all the data in a CIP with this FDF code'. - The packet includes dummy data. But current implementation is a combination of them. - Size of packet is 2 (way 1) - FDF = 0xff (way 2) This causes BeBoB chipset cannot sound. This patch applies Way 1. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Cc: Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-22ALSA: hda - Set current_headset_type to ALC_HEADSET_TYPE_ENUM (janitorial)David Henningsson1-1/+1
current_headset_type should be of the HEADSET_TYPE enum, not the HEADSET_MODE enum. Since ALC_HEADSET_TYPE_UNKNOWN and ALC_HEADSET_MODE_UNKNOWN are both 0, this patch is just janitorial. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-22ALSA: hda - Provide missing pin configs for VAIO with ALC260Takashi Iwai1-0/+20
Some models (or maybe depending on BIOS version) of Sony VAIO with ALC260 give no proper pin configurations as default, resulting in the non-working speaker, etc. Just provide the whole pin configurations via a fixup. Reported-by: Matthew Markus <mmarkus@hearit.co> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-21Merge tag 'asoc-v3.13-5' of ↵Takashi Iwai5-42/+86
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v3.13 A bunch of device specific fixes, nothing with a general impact here.
2013-11-21ALSA: hda - Add headset quirk for Dell Inspiron 3135David Henningsson1-0/+1
Cc: stable@vger.kernel.org (3.10+) BugLink: https://bugs.launchpad.net/bugs/1253636 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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