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2008-01-31[ALSA] S3C2412: suspend and resume supportBen Dooks
Support for suspend/resume for the S3C2412 ASoC IIS core driver. Signed-off-by: Ben Dooks <ben-linux@fluff.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] ASoC: S3C2412 IIS driverBen Dooks
S3C2412 SoC IIS support for ALSA/ASoC Signed-off-by: Ben Dooks <ben-linux@fluff.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] soc - Support suspend and resume of the I2S interface on s3c24xxGraeme Gregory
Signed-off-by: Graeme Gregory <graeme@openmoko.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] soc - Reinitialise DMA on every resumeGraeme Gregory
This one changes the DMA initialisation as it turns out the DMA driver in s3c24xx doesnt store registers between suspend/resume so you have to re-initialise the channels on every resume. Signed-off-by: Graeme Gregory <graeme@openmoko.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] soc - Initial WM8753 TLV support for capture mixerLiam Girdwood
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] soc - Don't lock the codec list in snd_soc_dapm_new_widgets()Mark Brown
snd_soc_dapm_new_widgets() takes the codec lock when adding new widgets, causing lockdep warnings when applications later call down through ALSA to adjust controls. Since widgets are only added during probe this lock should be unneeded so don't take it. Thanks to Dmitry Baryshkov <dbaryshkov@gmail.com> for reporting this issue. Cc: Dmitry Baryshkov <dbaryshkov@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] soc - Add support for passing kcontrols with eventsLaim Girdwood
Signed-off-by: Laim Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] soc - Clean up tabsLiam Girdwood
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] soc - Fix power switching support for DAPM_SWITCH widgetsMilan plzik
Signed-off-by: Milan plzik <milan.plzik@gmail.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] soc - Ensure PCMs are suspendedLiam Girdwood
This fixes a bug whereby PCMs were not being suspended when the rest of the audio subsystem was suspended. Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] soc - Add D1 power event to power down event sequenceLiam Girdwood
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] ASoC TLV supportPhilipp Zabel
Add TLV support to ASoC. Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] soc - Add device level DAPM eventLiam Girdwood
Added a device level dapm event so that both the machine and codec are informed when dapm events occur. Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] Fix inverted Phone volume WM9712 mixer controlJoe Sauer
Signed-off-by: Joe Sauer <jsauer@vernier.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda: STAC9228 VT fixesMatthew Ranostay
Moved 2 systems PCI_QUIRK values to STAC_DELL_BIOS. Also the second front HP jack is incorrect defined in the BIOS VT's for some laptops, this patch corrects this. Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda-codec - Device ID for Macbook sound cardJiang zhe
Please refer to the [0003680] on ALSA bugtracking system. The user found that 'model=mbp3' works and provided the ID. From: Jiang zhe <zhe.jiang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda-codec - Update realtek codec supportKailang Yang
1. Support HP rp5700 2. Fixed alc_subsystem_id function (Bug fixed and support Desktop) 3. Support ASUS EP20 Signed-off-by: Kailang Yang <kailang@realtek.com.tw> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] rawmidi: let sparse know what is going on _for real_Marcin Ślusarz
snd_rawmidi_kernel_read1/write1 weren't annotated but used copy_to_user/copy_from_user when one of parameters (kernel) was equal to 0 remove it and add properly annotated parameter Signed-off-by: Marcin Ślusarz <marcin.slusarz@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda: 92HD71BXX Mono Mute SupportMatthew Ranostay
Added a mono output mute mixer for the 92hd71bxx family of codecs, this also removes the need for the mono out node to explicitly unmuted in the core init. Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] Remove sound/driver.hTakashi Iwai
This header file exists only for some hacks to adapt alsa-driver tree. It's useless for building in the kernel. Let's move a few lines in it to sound/core.h and remove it. With this patch, sound/driver.h isn't removed but has just a single compile warning to include it. This should be really killed in future. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] Remove PCM sleep_min and tickTakashi Iwai
The 'tick' in PCM is set (again) via sw_params. And, nobody uses this feature at all except for a command line option of aplay. (This is literally 'nobody', as I checked alsa-lib API calls in all programs in major distros.) Above all, if we need finer wake-ups for the position update, it's basically an issue that the driver should solve, not tuned by each application. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] PCM - clean up snd_pcm_lib_read/writeTakashi Iwai
Introduce a common helper function for snd_pcm_lib_read and snd_pcm_lib_write for cleaning up the code. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] Remove PCM xfer_align sw paramsTakashi Iwai
The xfer_align sw_params parameter has never been used in a sane manner, and no one understands what this does exactly. The current implementation looks also buggy because it allows write of shorter size than xfer_align. So, if you do partial writes, the write isn't actually aligned at all. Removing this parameter will make some pcm_lib_* code more readable (and less buggy). Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] Fix PCM write blockingTakashi Iwai
The snd_pcm_lib_write1() may block in some weird condition: - the stream isn't started - avail_min is big (e.g. period size) - partial write up to buffer_size - avail_min The patch fixes this invalid blocking problem. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] Remove indirect control accessTakashi Iwai
This patch removes the indirect control access to the control elements. The indirect access has never been used and is even broken on 32bit ioctl wrapper. Let's clean it up. The pointers still remain in snd_ctl_elem_* structs just to make sure that the struct size won't change. Once after checking the size consistency, we can get rid of them, too. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda-codec - Add model=laptop for HP 350 laptopTakashi Iwai
Added the proper model=laptop for HP 350 laptop with Cxt5045 codec. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda-codec - Add test model for ALC268Jonathan Woithe
This implements a test model for the ALC268. It depends on the feature added by alc260-test-eapd-0.2.diff. This patch also adds a mention of the ALC260 test model to ALSA-Configuration.txt since this seems to have been missed. Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda-codec - Add EAPD controls for ALC260 test modelJonathan Woithe
This implements a switch control for the EAPD signal output by the ALC26x chips. Since some laptops may utilise this to activate useful things it is handy to have a control for this in the ALC26x test models. The patch includes the control in the ALC260 test model. Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] PCM core - remove SNDRV_PCM_TSTAMP_MMAP condition in snd_pcm_status()Jaroslav Kysela
The condition caused that the returned ring buffer position does not match with timestamp when SNDRV_PCM_TSTAMP_MMAP mode was enabled. Removing condition makes unified behaviour and interrupt based timestamp can be accessed via PCM_IOCTL_SYNC_PTR or mmaped status area. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda: Dynamically create digital gain mixersMatthew Ranostay
Dynamically create digital gain mixers for dmics that have out-amp support. Also some 92HD73xx's codecs don't have DMIC gains, so this also prevents creating dead mixers. Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda-intel - Support multiple devicesTakashi Iwai
It turned out that there can be multiple HD-audio devices on a single machine (e.g. on-board audio and HDMI on graphic cards), so we need to support multiple devices with snd-hda-intel driver. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] sound: Use time_before, time_before_eq, etc.Julia Lawall
The functions time_before, time_before_eq, time_after, and time_after_eq are more robust for comparing jiffies against other values. A simplified version of the semantic patch making this change is as follows: (http://www.emn.fr/x-info/coccinelle/) // <smpl> @ change_compare_np @ expression E; @@ ( - jiffies <= E + time_before_eq(jiffies,E) | - jiffies >= E + time_after_eq(jiffies,E) | - jiffies < E + time_before(jiffies,E) | - jiffies > E + time_after(jiffies,E) ) @ include depends on change_compare_np @ @@ #include <linux/jiffies.h> @ no_include depends on !include && change_compare_np @ @@ #include <linux/...> + #include <linux/jiffies.h> // </smpl> Signed-off-by: Julia Lawall <julia@diku.dk> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] es18xx: Enable wavetable input from ESS chipsKrzysztof Helt
This patch enables wavetable chips ES689/ES69X connected to ESS ES18xx chips. The wavetable chip uses FM DAC if the clock signal from the wavetable is active. It has no effect if there is no ESS wavetable chip present. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda-codec - Add IEC958 digital out support for Lenovo Thinkpads T61/X61Jerone Young
This patch adds IEC958 digital out support for the AD1984 sound card. This card can be found in Lenovo Thinkapds T61/X61. The digital out is not located on the Thinkpad, but optional docking station (it's coxial digital out). I've add this support as it is done the exact same way for the AD1983 & AD1884. I have tested this patch with my Lenovo Thinkpad T61 hooked up to a docking station (that has the digital coxial) and then run to my home theater reciever. Works like a charm :-) Signed-off-by: Jerone Young <jerone@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda: STAC927x VREF fixMatthew Ranostay
Some laptops incorrectly assume the front input jack as a line in instead of a microphone in. Which in turn disables the voltage reference, in which non-amplified input is not possible. This patch enables VREF80 for the input jack. Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] oxygen: use uintptr_t in pointer castsClemens Ladisch
When we store the DMA channel number in the substream's private_data pointer, use uintptr_t as an intermediate step when casting from/to unsigned int to prevent the compiler from whining when the pointer size is different. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] sound/usb/usbaudio.c: fix build with CONFIG_PM=nAndrew Morton
sound/usb/usbaudio.c: In function 'usb_audio_suspend': sound/usb/usbaudio.c:3674: error: implicit declaration of function 'snd_pcm_sus\pend_all' Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda-codec - Sort ad1986a cfg tableTakashi Iwai
Sort the ad1986a config table by PCI SSID (the last toshiba entry was added wrongly). Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] add Asus Xonar driverClemens Ladisch
Add the snd-virtuoso driver for the Asus Virtuoso 200 chip used on the PCI and PCI-E models of the Xonar sound card. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] add CMI8788 driverClemens Ladisch
Add the snd-oxygen driver for the C-Media CMI8788 (Oxygen) chip, used on the Asound A-8788, AuzenTech X-Meridian, Bgears b-Enspirer, Club3D Theatron DTS, HT-Omega Claro, Razer Barracuda AC-1, Sondigo Inferno, and TempoTec HIFIER sound cards. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda-codec - alc268 input_mux should be a selector instead of mixerJiang Zhe
According to the [0003659], the node 0x23,0x24 is a selector. I checked the alc268 spec on the REALTEK website and it showed that they were selectors indeed. However, current code implement the alc268 input_mux in a mixer way. Signed-off-by: Jiang Zhe <zhe.jiang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda-codec - Device ID for Toshiba laptop which uses AD1986AJiang Zhe
The model laptop-eapd get rid of the high-pitched noise. (ALSA bug#3662) Signed-off-by: Jiang Zhe <zhe.jiang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda-codec - Fix capture mixers of ALC662 modelsHerton Ronaldo Krzesinski
The commit that added support for ASUS P701 eeepc also changed the mixers of other ALC662 models, duplicating entries for the Capture items, making them to not work anymore. This fixes it by removing duplicated entries using where possible the common alc662_capture_mixer. Also alc662_capture_mixer should use alc662* functions and not alc882 (I checked /proc/asound/card0/codec* on an eepc model and it's ok). Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] neo1973: ASoC include pathname fixHarald Welte
Fix s3c24xx include file path changes in asoc driver Signed-off-by: Harald Welte <laforge@openmoko.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] usb-audio: add UR-80 PCM quirkClemens Ladisch
Add a quirk entry to handle Edirol UR-80 audio I/O. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda-codec - Add missing #defines (and 1 rename) in hda_codec.hAndrew Paprocki
Added AC_VERB_GET_DIGI_CONVERT_2 and renamed AC_VERB_GET_DIGI_CONVERT to AC_VERB_GET_DIGI_CONVERT_1 to stay consistent with the SET variants. Added AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, and AC_PINCAP_LR_SWAP. The missing fields were listed in the ALC883 datasheet rev 1.3. Signed-off-by: Andrew Paprocki <andrew@ishiboo.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] hda-codec - Fix invalid access to non-existing dmux on STACTakashi Iwai
The digital mux on STAC codecs doesn't always exist although the driver builds dmux enum mixer elements unconditionally. Now the driver creates 'digital input source' mixer elements only when dmux is available. Also, the patch adds the missing dmux definition for STAC925x. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] cs4270: wrong sample rate when CONFIG_SND_SOC_CS4270_VD33_ERRATA is setTimur Tabi
When CONFIG_SND_SOC_CS4270_VD33_ERRATA is set, there was a mismatch between the mclk_ratios[] and cs4270_mode_ratios[] arrays. The two arrays have been merged and code has been shuffled. One side effect is that the cs4270_set_dai_sysclk() and cs4270_set_dai_fmt() functions are available only if I2C has been enabled. Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] sound/: Spelling fixesJoe Perches
Signed-off-by: Joe Perches <joe@perches.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-31[ALSA] at73c213: replace spinlock in mixer functions with a mutexHans-Christian Egtvedt
This patch fixes the locking bug in the at73c213 SPI sound driver. This bug was triggered because spinlocks were wrapped around the spi_sync call which might sleep. The fix was to add a mutex to the sound driver and replace the spinlocks in the mixer functions with mutex lock/unlock. Tested on STK1000/STK1002. Signed-off-by: Hans-Christian Egtvedt <hcegtvedt@atmel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>