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* sound/oss-msnd-pinnacle: ioctl needs the inodeArnd Bergmann2010-07-141-1/+1
| | | | | | | | This broke in sound/oss: convert to unlocked_ioctl, when I missed one of the ioctl functions still using the inode pointer. Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* sound/oss: convert to unlocked_ioctlArnd Bergmann2010-07-126-51/+119
| | | | | | | | These are the final conversions for the ioctl file operation so we can remove it in the next merge window. Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* sound: push BKL into open functionsArnd Bergmann2010-07-128-35/+89
| | | | | | | | | | | | | | | This moves the lock_kernel() call from soundcore_open to the individual OSS device drivers, where we can deal with it one driver at a time if needed, or just kill off the drivers. All core components in ALSA already provide adequate locking in their open()-functions and do not require the big kernel lock, so there is no need to add the BKL there. Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: via82xx: allow changing the initial DXS volumeClemens Ladisch2010-07-121-2/+7
| | | | | | | | | | | | | As per-stream volume controls, the DXS controls are not intended to adjust the overall sound level and so are initialized every time a stream is opened. However, there are special situations where one wants to reduce the overall volume in the digital domain, i.e., before the AC'97 codec's PCM volume control. To allow this, add a module parameter that sets the initial DXS volume. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Tested-by: Soeren D. Schulze <soeren.d.schulze@gmx.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: silence a superfluous warningClemens Ladisch2010-07-091-4/+1
| | | | | | | | | | It is not advisable to print a warning when a device does not support setting the sample rate because this is perfectly valid for devices with a single rate or where rates are implicitly changed by selecting another alternate setting. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Remove unneeded ;Eliot Blennerhassett2010-07-061-1/+1
| | | | | Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Minor HPI error handling fixesEliot Blennerhassett2010-07-061-2/+8
| | | | | | | | Handle errors in tuner level caching, Ccorrect error code for aesebu rx status. Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Change compander API and tidyEliot Blennerhassett2010-07-062-143/+211
| | | | | | | | | Compander API changed to one function per parameter. Factor out some common code for stereo log value reading. Make some more entity functions static. Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Add ASI5200 familyEliot Blennerhassett2010-07-061-0/+7
| | | | | Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - Use version string instead of printf formattingEliot Blennerhassett2010-07-061-3/+1
| | | | | Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: asihpi - HPI API updatesEliot Blennerhassett2010-07-063-27/+55
| | | | | | | | | Remove some deprecated items. Change compander api to one function per parameter. Add a version string define. Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* soundcore_open: Reduce the area BKL coverageJohn Kacur2010-07-051-5/+8
| | | | | | | | | | | | | | Most of this function is protected by the sound_loader_lock. We can push down the BKL to this call out err = file->f_op->open(inode,file); In order to build the sound core without the BKL, we will need to push the lock_kernel() call into the ~20 device drivers that register their file operations. Signed-off-by: John Kacur <jkacur@redhat.com> Signed-off-by: Arnd Bergmann <arnd@arndb.de> Acked-by: Alan Cox <alan@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'devel' of git://git.alsa-project.org/alsa-kernel into topic/miscTakashi Iwai2010-07-052-15/+24
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| * sis7019: increase reset delaysDavid Dillow2010-06-281-3/+3
| | | | | | | | | | | | | | | | | | A few boards using this controller are reported to need a little extra time during their reset cycle. Reported-by: Michael Goeke <michael.goeke@icachip.de> Signed-off-by: Dave Dillow <dave@thedillows.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * sis7019: fix capture issues with multiple periods per bufferDavid Dillow2010-06-281-4/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When using a timing voice to clock out periods during capture, the driver would slowly loose synchronization and never catch up, eventually reaching a point where it no longer generated interrupts. To avoid this situation, the virtual period clocking was changed to shorten the next timing period when our timing voice falls too far behind the capture voice. In addition, the first virtual period for the timing voice was slightly too short, causing the timing voice to initially be ahead of the capture voice. While tracking down this problem, I noticed that the expected sample offset was being incorrectly initialized, causing an overrun to be incorrectly reported when the timing voice happened to be perfectly synchronized. Reported-by: Hans Schou <linux@schou.dk> Signed-off-by: Dave Dillow <dave@thedillows.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: pcm_lib: avoid timing jitter in snd_pcm_read/write()David Dillow2010-06-281-8/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When using poll() to wait for the next period -- or avail_min samples -- one gets a consistent delay for each system call that is usually just a little short of the selected period time. However, When using snd_pcm_read/write(), one gets a jittery delay that alternates between less than a millisecond and approximately two period times. This is caused by snd_pcm_lib_{read,write}1() transferring any available samples to the user's buffer and adjusting the application pointer prior to sleeping to the end of the current period. When the next period interrupt occurs, there is then less than avail_min samples remaining to be transferred in the period, so we end up sleeping until a second period occurs. This is solved by using runtime->twake as the number of samples needed for a wakeup in addition to selecting the proper wait queue to wake in snd_pcm_update_state(). This requires twake to be non-zero when used by snd_pcm_lib_{read,write}1() even if avail_min is zero. Signed-off-by: Dave Dillow <dave@thedillows.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: hda-intel - fix wallclk variable update and conditionJaroslav Kysela2010-06-021-2/+2
| | | | | | | | | | | | | | | | | | | | | | This patch fixes thinko introduced in "last minutes" before commiting of the last wallclk patch. It also fixes the condition checking if the first period after last wallclk update is processed. There is a little rounding error in period_wallclk. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* | ALSA: usb - Fix compile error with CONFIG_SND_DEBUG_VERBOSE=yTakashi Iwai2010-06-241-1/+1
| | | | | | | | | | | | Replaced the forgotten cval->mixer->ctrlif. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: simplify control interface accessDaniel Mack2010-06-238-44/+34
| | | | | | | | | | | | | | | | | | | | | | As the control interface is now carried in struct snd_usb_audio, we can simplify the API a little and also drop the private ctrlif field from struct usb_mixer_interface. Also remove a left-over function prototype in pcm.h. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: move and add some commentsDaniel Mack2010-06-232-10/+30
| | | | | | | | | | | | | | Also add a list of open topics. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-midi: whitespace fixesDaniel Mack2010-06-231-7/+7
| | | | | | | | | | Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: unify UAC macros and struct namesDaniel Mack2010-06-233-10/+10
| | | | | | | | | | | | | | | | | | | | Get rid of the last occurances of _v1 suffixes, and move the version number right after the "uac" string. Now things are consitent again. Sorry for the forth and back, but it just looks much nicer this way. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: clean up includes in clock.cDaniel Mack2010-06-231-15/+1
| | | | | | | | | | Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge branch 'fix/misc' into topic/miscTakashi Iwai2010-06-2321-266/+670
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| * | ALSA: usb-audio - Add volume resolution quirk for some Logitech webcamsAlexey Fisher2010-06-231-0/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some programs like Skype trying to set capture volume automatically. Normally it will tray, carefully step by step lover or higher, set the volume. In real word it work not really well, because devises and vendors lie about real audio settings. For example most Logitech webcams have 6400 or 3500 steps for capture volume. They do not tell that actual resolution is 384. So we have only 7 or 18 real steps. In this patch I set real resolution only for tested devices. Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usb/endpoint, fix dangling pointer useJiri Slaby2010-06-211-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Stanse found that in snd_usb_parse_audio_endpoints, there is a dangling pointer dereference. When snd_usb_parse_audio_format fails, fp is freed, and continue invoked. On the next loop, there is "fp && fp->altsetting == 1 && fp->channels == 1" test, but fp is set from the last iteration (but is bogus) and thus ilegally dereferenced. Set fp to NULL before "continue". Signed-off-by: Jiri Slaby <jslaby@suse.cz> Acked-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: asihpi - Get rid of incorrect "long" types and casts.Eliot Blennerhassett2010-06-171-11/+11
| | | | | | | | | | | | | | | | | | | | | These give incorrect results for index wrap on 64 bit. Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usb-audio: fix UAC2 control value queriesDaniel Mack2010-06-111-5/+14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | For RANGE requests, we should only query as much bytes as we're in fact interested in. For CUR requests, we shouldn't confuse the firmware with an overlong request but just ask for 2 bytes. This might need fixing in the future as it's not entirely clear when to dispatch 1-byte, 2-byte and 4-byte request blocks. For now, we assume everything is coded in 16bit - this works for all firmware implementations I've seen. Signed-off-by: Daniel Mack <daniel@caiaq.de> Reported-by: Alex Lee <alexlee188@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usb-audio: parse UAC2 sample rate ranges correctlyDaniel Mack2010-06-111-18/+74
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | A device may report its supported sample rates in ranges rather than in discrete triplets. The code used to only parse the MIN field instead of properly paying attention to the MAX and RES values. Also, handle RES values of 1 correctly and announce a continous sample rate range in this case. Signed-off-by: Daniel Mack <daniel@caiaq.de> Reported-by: Alex Lee <alexlee188@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usb-audio: fix control messages for USB_RECIP_INTERFACEDaniel Mack2010-06-113-6/+16
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Control messages directed to an interface must have the interface number set in the lower 8 bits of wIndex. This wasn't done correctly for some clock and mixer messages. Signed-off-by: Daniel Mack <daniel@caiaq.de> Reported-by: Alex Lee <alexlee188@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usb-audio: add check for faulty clock in parse_audio_format_rates_v2()Daniel Mack2010-06-111-0/+6
| | | | | | | | | | | | | | | Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: sound/spi: patch for the unuseful variable removalWan ZongShun2010-06-081-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | The '*bitclk' of structure 'snd_at73c213' seems no use, so I make a patch to remove the unnecessary variable. Signed-off-by: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: atmel: set "channel A event" output to debugYegor Yefremov2010-06-081-1/+1
| | | | | | | | | | | | | | | Signed-off-by: Yegor Yefremov <yegorslists@googlemail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge branch 'for-linus' of ↵Linus Torvalds2010-06-0416-227/+535
| |\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: ALSA: hda-intel - fix wallclk variable update and condition ALSA: asihpi - Fix uninitialized variable ALSA: hda: Use LPIB for ASUS M2V usb/gadget: Replace the old USB audio FU definitions in f_audio.c ASoC: MX31ads sound support should depend on MACH_MX31ADS_WM1133_EV1 ASoC: Add missing Kconfig entry for Phytec boards ALSA: usb-audio: export UAC2 clock selectors as mixer controls ALSA: usb-audio: clean up find_audio_control_unit() ALSA: usb-audio: add UAC2 sepecific Feature Unit controls ALSA: usb-audio: unify constants from specification ALSA: usb-audio: parse clock topology of UAC2 devices ALSA: usb-audio: fix selector unit string index accessor include/linux/usb/audio-v2.h: add more UAC2 details ALSA: usb-audio: support partially write-protected UAC2 controls ALSA: usb-audio: UAC2: clean up parsing of bmaControls ALSA: hda: Use LPIB for another mainboard ALSA: hda: Use mb31 quirk for an iMac model ALSA: hda: Use LPIB for an ASUS device
| | * \ Merge branch 'fix/asoc' into for-linusTakashi Iwai2010-06-021-1/+10
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| | | * \ Merge branch 'for-2.6.35' of ↵Takashi Iwai2010-06-021-1/+10
| | | |\ \ | | | | |/ | | | |/| | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc
| | | | * ASoC: MX31ads sound support should depend on MACH_MX31ADS_WM1133_EV1Sascha Hauer2010-05-311-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | | * ASoC: Add missing Kconfig entry for Phytec boardsSascha Hauer2010-05-311-0/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | Merge branch 'fix/hda' into for-linusTakashi Iwai2010-06-021-2/+2
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| | | * \ \ Merge branch 'master' of git.alsa-project.org:alsa-kernel into fix/hdaTakashi Iwai2010-06-021-2/+2
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| | | | * | | ALSA: hda-intel - fix wallclk variable update and conditionJaroslav Kysela2010-06-021-2/+2
| | | | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch fixes thinko introduced in "last minutes" before commiting of the last wallclk patch. It also fixes the condition checking if the first period after last wallclk update is processed. There is a little rounding error in period_wallclk. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| | * | | | Merge branch 'fix/hda' into for-linusTakashi Iwai2010-06-022-0/+4
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| | | * | | ALSA: hda: Use LPIB for ASUS M2VDaniel T Chen2010-06-011-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | BugLink: https://launchpad.net/bugs/587546 Symptom: On the reporter's ASUS M2V, using PulseAudio in Ubuntu 10.04 LTS results in the PA daemon crashing shortly after attempting playback of an audio file. Test case: Using Ubuntu 10.04 LTS (Linux 2.6.32.12), Linux 2.6.33, or Linux 2.6.34, attempt playback of an audio file while PulseAudio is active. Resolution: add SSID for this machine to the position_fix quirk table, explicitly specifying the LPIB method. Reported-and-Tested-By: D Tangman Cc: <stable@kernel.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: hda: Use LPIB for another mainboardDaniel T Chen2010-05-311-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | BugLink: https://launchpad.net/bugs/580749 Symptom: on the original reporter's VIA VT1708-based board, the PulseAudio daemon dies shortly after the user attempts to play an audio file. Test case: boot from Ubuntu 10.04 LTS live cd; attempt to play an audio file. Resolution: add SSID for the original reporter's hardware to the position_fix quirk table, explicitly specifying the LPIB method. Reported-and-Tested-By: Harald Cc: <stable@kernel.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: hda: Use mb31 quirk for an iMac modelDaniel T Chen2010-05-311-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | BugLink: https://launchpad.net/bugs/542550 Symptom: On the reporter's iMac, in Ubuntu 10.04 LTS neither playback nor capture appear audible out-of-the-box. Test case: Boot from an Ubuntu 10.04 LTS live cd or from an installed configuration and attempt to play or capture audio. Resolution: Specify the mb31 quirk for this machine in the codec SSID table. Reported-and-Tested-By: f3a97 Cc: <stable@kernel.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: hda: Use LPIB for an ASUS deviceDaniel T Chen2010-05-311-0/+1
| | | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | BugLink: https://launchpad.net/bugs/465942 Symptom: On the reporter's ASUS device, using PulseAudio in Ubuntu 10.04 LTS results in the PA daemon crashing shortly after attempting to select capture or to configure the audio hardware profile. Test case: Using Ubuntu 10.04 LTS (Linux 2.6.32.12), Linux 2.6.33, or Linux 2.6.34, adjust the HDA device's capture volume with PulseAudio. Resolution: add SSID for this machine to the position_fix quirk table, explicitly specifying the LPIB method. Reported-and-Tested-By: Irihapeti Cc: <stable@kernel.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | ALSA: asihpi - Fix uninitialized variableTakashi Iwai2010-06-021-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Initialize prev_ctl properly before reference: sound/pci/asihpi/asihpi.c: In function ‘snd_card_asihpi_mixer_new’: sound/pci/asihpi/asihpi.c:2568:30: warning: ‘prev_ctl.dst_node_index’ may be used uninitialized in this function Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | ALSA: usb-audio: export UAC2 clock selectors as mixer controlsDaniel Mack2010-05-311-7/+35
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The UAC2 clock selectors are fortunately compatible with UAC1 audio selector units, so we can simply reuse the same approach to get all the linked units. Requests to this control need a different CS value though. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | ALSA: usb-audio: clean up find_audio_control_unit()Daniel Mack2010-05-311-8/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Use a struct to parse the audio units, and return usable descriptors for all types. There's no need to limit the result set, except for some kind of sanity check. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | ALSA: usb-audio: add UAC2 sepecific Feature Unit controlsDaniel Mack2010-05-311-6/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The bits to enable them are always 0 for UAC1 devices, so no additional checks are required. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
generated by cgit v1.1 at 2024-12-26 23:55:53 +0000