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2012-01-16ALSA: Au88x0 - Support 4 channels playback when AC97 codecs has SDAC bitRaymond Yau
- Check SDAC bit of AC97 codec for supporting 4 channels playback. Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-16ALSA: HDA: Fix internal microphone on Dell Studio 16 XPS 1645David Henningsson
More than one user reports that changing the model from "both" to "dmic" makes their Internal Mic work. Cc: stable@kernel.org Tested-by: Martin Ling <martin-launchpad@earth.li> BugLink: https://bugs.launchpad.net/bugs/795823 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-13ALSA: Don't prompt for CONFIG_SND_COMPRESS_OFFLOADTakashi Iwai
CONFIG_SND_COMPRESS_OFFLOAD is an item to be selected by the dirver just like CONFIG_SND_PCM, and no need to prompt for explicit selection. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-13ALSA: HDA: Use LPIB position fix for Macbook Pro 7,1David Henningsson
Several users have reported "choppy" audio under the 3.2 kernel, and that changing position_fix to 1 has resolved their problem. The chip is an nVidia Corporation MCP89 High Definition Audio, [10de:0d94] (rev a2). Cc: stable@kernel.org (v3.2+) BugLink: https://bugs.launchpad.net/bugs/909419 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-12Merge branch 'topic/hda' into for-linusTakashi Iwai
2012-01-12Merge branch 'topic/misc' into for-linusTakashi Iwai
2012-01-12Merge branch 'for-3.3' of ↵Takashi Iwai
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into topic/asoc
2012-01-11Merge branch 'for-3.3' of ↵Takashi Iwai
git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc into topic/asoc
2012-01-11ASoC: twl6040 - Add method to query optimum PDM_DL1 gainLiam Girdwood
The DL1 PDM interface adds a little gain depending on the output device. Add a method to retrieve the gain value for machine driver usage. Signed-off-by: Liam Girdwood <lrg@ti.com>
2012-01-11ALSA: hda - Fix the lost power-setup of seconary pins after PM resumeTakashi Iwai
When multiple headphone or other detectable output pins are present, the power-map has to be updated after resume appropriately, but the current driver doesn't check all pins but only the first pin (since it's enough to check it for the mute-behavior). This resulted in the silent output from the secondary outputs after PM resume. This patch fixes the problem by checking all pins at (re-)init time. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=740347 Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-11ALSA: usb-audio: add Yamaha MOX6/MOX8 supportClemens Ladisch
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-11ALSA: virtuoso: add S/PDIF input support for all XonarsClemens Ladisch
All Xonar cards support S/PDIF input, but the cards without optical or coaxial plugs have only undocumented pin connectors. Support for the ST/STX was already added in a previous patch; this adds support for the D1/DX (JP2), DG (J5), DS (J5), and HDAV Slim (J12). Many thanks to Zoltan Miklos for testing the DS and DX. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-11ALSA: ice1724 - Support for ooAoo SQ210aPavel Hofman
This card shares PCI ids with Chaintec AV710. Therefore, it will not be detected automatically, it can only be activated by the module parameter model=sq210a. Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-11ALSA: ice1724 - Allow card info based on model onlyPavel Hofman
When two different cards share the same PCI vendor/subvendor identification, allow card info based on model only. Do not require subvendor ID. Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-11ALSA: ice1724 - Create capture pcm only for ADC-enabled configurationsPavel Hofman
Add the capture pcm only if there is at least one ADC configured in the SYSCONF register. Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-11ALSA: hdspm - Provide unique driver id based on card serialAdrian Knoth
Before, /proc/asound looked like this: 2 [Default ]: HDSPM - RME RayDAT_f1cd85 RME RayDAT S/N 0xf1cd85 at 0xf7300000, irq 18 In case of a second HDSPM card, its name would be Default_1. This is cumbersome, because the order of the cards isn't stable across reboots. To help userspace tools referring to the correct card, this commit provides a unique id for each card: 2 [HDSPMxf1cd85 ]: HDSPM - RME RayDAT_f1cd85 RME RayDAT S/N 0xf1cd85 at 0xf7300000, irq 18 In this example, userspace (configuration files) would then use hw:HDSPMxf1cd85 to choose the right card. The serial is masked to 24bits, so this string is always shorter than sixteen chars. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-10ASoC: Dynamically allocate the rtd device for a non-empty release()Mark Brown
The device model needs a release() function so it can free devices when they become dereferenced. Do that for rtds. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-10ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSCAxel Lin
commit 739be96 "ASoC: Fix build dependency for SND_ATMEL_SOC_SSC" introduces below build warnings: drivers/misc/Kconfig:212:error: recursive dependency detected! drivers/misc/Kconfig:212: symbol ATMEL_SSC is selected by SND_ATMEL_SOC_SSC sound/soc/atmel/Kconfig:9: symbol SND_ATMEL_SOC_SSC is selected by SND_AT91_SOC_SAM9G20_WM8731 sound/soc/atmel/Kconfig:18: symbol SND_AT91_SOC_SAM9G20_WM8731 depends on ATMEL_SSC SND_ATMEL_SOC_SSC needs ATMEL_SSC to pass compilation. This patch remove the "select ATMEL_SSC" from SND_ATMEL_SOC_SSC to avoid above warnings. And then ensures all the machine drivers that select SND_ATMEL_SOC_SSC need to depend on ATMEL_SSC. Reported-by: Stephen Rothwell <sfr@canb.auug.org.au> Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-10ALSA: hda - Fix the detection of "Loopback Mixing" control for VIA codecsTakashi Iwai
Currently the driver checks only the out_mix_path[] for the primary output route for judging whether to create the loopback-mixing control or not. But, there are cases where aamix-routing is available only on headphone or speaker paths but not on the primary output path. So, the driver ignores such cases inappropriately. This patch fixes the check of the loopback-mixing control by testing all mix-routing paths. Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-10ALSA: hda - Return the error from get_wcaps_type() for invalid NIDsTakashi Iwai
When an invalid NID is given, get_wcaps() returns zero as the error, but get_wcaps_type() takes it as the normal value and returns a bogus AC_WID_AUD_OUT value. This confuses the parser. With this patch, get_wcaps_type() returns -1 when value 0 is given, i.e. an invalid NID is passed to get_wcaps(). Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=740118 Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-10ALSA: hda - Use auto-parser for HP laptops with cx20459 codecTakashi Iwai
These laptops can work well with the auto-parser and their BIOS setups, and in addition, the auto-parser fixes the problem with S3/S4 where the unsol event handling is killed after resume due to fallback to the single-cmd mode. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=740115 Cc: <stable@kernel.org> [v3.1+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-09ALSA: asihpi - Fix potential Oops in snd_asihpi_cmode_info()Takashi Iwai
Dan Carpenter reported that setting 0 to uinfo->value.enumerated.items in snd_asihpi_cmode_info() may lead to Oops. This function should return an error immediately in such a case instead. Cc: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-09ALSA: hdsp - Fix potential Oops in snd_hdsp_info_pref_sync_ref()Takashi Iwai
Dan Carpenter reported that setting 0 to uinfo->value.enumerated.items in snd_hdsp_info_pref_sync_ref() may lead to Oops. This function should return an error immediately in such a case instead. Cc: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-09ALSA: hda/cirrus - support for iMac12,2 modelJérémy Lal
This early 2011 model just need to have headphones on GPI02 instead of GPI01, and use BIOS pincfgs. It is detected by codec SSID. The iMac12,1 model is known to work the same way, although maybe not with the same codec SSID. Signed-off-by: Jérémy Lal <kapouer@melix.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-09ASoC: cx20442: add bias control over a platform provided regulatorJanusz Krzysztofik
Now that a regulator device for controlling the codec chip reset state over a platform agnostic regulator API is available on the only board using this driver so far, extend the driver with a bias control function which will request virtual power to the codec chip from that virtual regulator, and will supersede the present implementation existing at the sound card level. Thanks to the regulator sharing mechanism, both the old (the sound card) and the new (the codec) implementations should coexist smoothly until the sound card file is updated. For this to work as expected, update the sound card .set_bias_level callback to not touch codec->dapm.bias_level. While extending the cx20442 structure, drop unused control_type member. Created against linxu-3.2-rc6, tested on top of patch 1/4 "ARM: OMAP1: ams-delta: set up a regulator over the modem reset GPIO pin". Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@ti.com>
2012-01-09ALSA: usb-audio - Avoid flood of frame-active debug messagesTakashi Iwai
With some buggy devices, the usb-audio driver may give "frame xxx active" kernel messages too often. Better to keep it as debug-only using snd_printdd(), and also add the rate-limit for avoiding floods. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=738681 Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-09ALSA: snd-usb-us122l: Delete calls to preempt_disableKarsten Wiese
They are not needed here. Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de> Cc: stable@kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-09mfd: Put WM8994 into cache only mode when suspendingMark Brown
This is required by the ASoC driver for very low power modes where the device is fully idle but we want to update controls. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-08ASoC: Fix idma build after update for channel count checkMark Brown
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-08ALSA: hdspm - Refactor serial number to avoid code duplicationAdrian Knoth
The serial number is used multiple times in hdspm.c. Since it belongs to the card, let's store it in struct hdspm and refer to it whenever necessary. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08ALSA: usb-audio: fix possible hang and overflow in ↵Xi Wang
parse_uac2_sample_rate_range() A malicious USB device may feed in carefully crafted min/max/res values, so that the inner loop in parse_uac2_sample_rate_range() could run for a long time or even never terminate, e.g., given max = INT_MAX. Also nr_rates could be a large integer, which causes an integer overflow in the subsequent call to kmalloc() in parse_audio_format_rates_v2(). Thus, kmalloc() would allocate a smaller buffer than expected, leading to a memory corruption. To exploit the two vulnerabilities, an attacker needs physical access to the machine to plug in a malicious USB device. This patch makes two changes. 1) The type of "rate" is changed to unsigned int, so that the loop could stop once "rate" is larger than INT_MAX. 2) Limit nr_rates to 1024. Suggested-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Xi Wang <xi.wang@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08ALSA: Au88x0 - Fix channels swapping of 4 channels playbackRaymond Yau
Fix channels swapping of 4 channels playback by using vortex_adbdma_stopfifo instead of vortex_adbdma_pausefifo for SNDRV_PCM_TRIGGER_STOP event Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08ALSA: Au88x0 - Fix IRQ fifo error and channels swap of 4 channels playbackRaymond Yau
Fix IRQ fifo error when playing stereo by set stereo flag of fifo control. This also fix the swap of front and rear channels on au8830. Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08ALSA: Au88x0 - Fix Xtalk's constantsRaymond Yau
- Fix XtalkGainsDefault, XtalkGains1Chn - Fix XtalkWideCoefsLeftEQ, XtalkWideCoefsRightEQ - Fix XtlakWideCoefsLeftXT, XtalkWideCoefsRightXT Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08ALSA: Au88x0 - Xtalk - fix write/read of eq and xt instatesRaymond Yau
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08ALSA: ice1724 - External clock item only for cards with SPDIF_INPavel Hofman
Append the external clock item to the clock list only if the SPDIF_IN capability is defined in the SPDIF register. Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08ALSA: ice1724 - Check for ac97 to avoid kernel oopsPavel Hofman
Cards with identical PCI ids but no AC97 config in EEPROM do not have the ac97 field initialized. We must check for this case to avoid kernel oops. Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08ALSA: HDA: Remove Poulsbo position fix quirksDavid Henningsson
Now that we have changed the poulsbo chip to use LPIB position fix, we can remove the individual machine quirks that do the same thing. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08ALSA: HDA: Fix typo for ALC269VB_FIXUP_DMICDavid Henningsson
This fixup is not actually used, so in practice this is just a cosmetic fix. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08ALSA: HDA: Add support for Cirrus Logic 4213David Henningsson
The CS4213 chip is similar to the CS4210, but it does not have SPDIF capabilities. Also, it has fewer pins, and the vendor specific nid is different. With this patch, we have working inputs and outputs (and automute/autoswitch). However, we don't know anything about the vendor specific processing coefficients, so we don't read or write to that node in this patch. BugLink: https://bugs.launchpad.net/bugs/910792 Tested-by: Hsin-Yi Chen <hychen@canonical.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08ALSA: HDA: Fix automute for Cirrus Logic 421xDavid Henningsson
There was a bug in the automute logic causing speakers not to mute when headphones were plugged in. Cc: stable@kernel.org Tested-by: Hsin-Yi Chen <hychen@canonical.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08ALSA: HDA: Fix master control for Cirrus Logic 421XDavid Henningsson
The control name "HP/Speakers" is non-standard, and since there is only one DAC on this chip there is no need for a virtual master anyway. Cc: stable@kernel.org Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08ALSA: HDA: Use LPIB position fix for OaktrailDavid Henningsson
According to the thread on alsa-devel, the LPIB method is to prefer for Oaktrail controller chip. Reference: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-January/047800.html Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-07ASoC: check for substream not channels_min in pcm enginesJoachim Eastwood
This is a follow up on 53dea36c70c1857 which fixes the other affected pcm engines. Description from 53dea36c70c1857: Don't rely on the codec's channels_min information to decide wheter or not allocate a substream's DMA buffer. Rather check if the substream itself was allocated previously. Without this patch I was seeing null-pointer dereferenc in atmel-pcm. Signed-off-by: Joachim Eastwood <joachim.eastwood@jotron.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-05ASoC: Fix build dependency for SND_ATMEL_SOC_SSCAxel Lin
Make SND_ATMEL_SOC_SSC select ATMEL_SSC to fix below build errors: LD .tmp_vmlinux1 sound/built-in.o: In function `atmel_ssc_remove': sound/soc/atmel/atmel_ssc_dai.c:713: undefined reference to `ssc_free' sound/built-in.o: In function `atmel_ssc_probe': sound/soc/atmel/atmel_ssc_dai.c:700: undefined reference to `ssc_request' sound/built-in.o: In function `atmel_ssc_set_audio': sound/soc/atmel/atmel_ssc_dai.c:845: undefined reference to `ssc_request' sound/soc/atmel/atmel_ssc_dai.c:851: undefined reference to `ssc_free' make: *** [.tmp_vmlinux1] Error 1 Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-03ASoC: sta32x: Optimize the array work to find rate_min and rate_maxAxel Lin
For a given ir and fs, there is at most one possible match for the case mclk_ratios[ir][j].ratio * fs == freq. Thus we can break from the inner loop once a match is found. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Johannes Stezenbach <js@sig21.net> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-03ASoC: soc-pcm: Allocate PCM operations dynamically to support multiple DAIsSangsu Park
The original code does not cover the case that two DAIs(CPU) have different ASoC core PCM operations(like mmap, pointer...). Currently we have only one global soc_pcm_ops for ASoC core PCM operation. When two DAIs have different pointer functions, second DAI's pointer function is set for both first DAI and second DAI in case of original code. This patch uses runtime's pcm_ops instead of global pcm_ops for each DAIs. So each DAIs can have different ASoC core PCM operations. This is needed to support multiple DAIs. Signed-off-by: Sangsu Park <sangsu4u.park@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-02ASoC: pxa: Convert corgi to use snd_soc_register_card()Axel Lin
Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Haojian Zhuang <haojian.zhuang@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-02ASoC: Fix return value of wm8580_set_sysclk()Axel Lin
We can't just pass back the return value of snd_soc_update_bits() as it will be 1 if a bit changed rather than zero. Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-02ASoC: Use dai_fmt in tavorevb3 machine driverAxel Lin
Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Haojian Zhuang <haojian.zhuang@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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