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2010-02-17Merge branch 'fix/misc' into topic/miscTakashi Iwai16-57/+191
Conflicts: sound/pci/hda/patch_realtek.c
2010-02-17Merge remote branch 'alsa/fixes' into fix/miscTakashi Iwai2-7/+19
2010-02-17ALSA: Echoaudio, fix Guru Meditation #00000005.48454C50Giuliano Pochini1-1/+3
This patch fixes a division by zero error in the irq handler. There is a small window between the hw_params() callback and when runtime->frame_bits is set by ALSA middle layer. When another substream is already running, if an interrupt is delivered during that window the irq handler calls pcm_pointer() which does a division by zero. The patch below makes the irq handler skip substreams that are initialized but not started yet. Cc to Clemens Ladisch because he proposed an alternate fix. For more information, please read the original thread in the linux-kernel mailing list: http://lkml.org/lkml/2010/2/2/187 Signed-off-by: Giuliano Pochini <pochini@shiny.it> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-16ALSA: usbmixer - use MAX_ID_ELEMS where possibleJaroslav Kysela1-2/+3
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-16ALSA: usbmixer - add usb_id value to usbmixer proc fileJaroslav Kysela1-2/+3
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-16ALSA: pcm core - fix fifo_size channels interval checkJaroslav Kysela1-4/+4
Signed-off-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@kernel.org>
2010-02-16ALSA: usbmixer - introduce /proc/asound/card#/usbmixer fileJaroslav Kysela1-14/+61
The usbmixer proc file contains mapping between ALSA control API and USB mixer control units. The purpose of this file is for debugging and a problem diagnostics. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-16Merge branch 'topic/misc' of ↵Jaroslav Kysela119-1321/+3364
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into devel
2010-02-16Merge branch 'fixes' into develJaroslav Kysela24-132/+155
2010-02-16ALSA: USB MIDI support for Access Music VirusTISebastien Alaiwan3-0/+65
Here's a patch that adds MIDI support through USB for one of the Access Music synths, the VirusTI. The synth uses standard USBMIDI protocol on its USB interface 3, although it does signal "vendor specific" class. A magic string has to be sent on interface 3 to enable the sending of MIDI from the synth (this string was found by sniffing usb communication of the Windows driver). This is all my patch does, and it works on my computer. Please note that the synth can also do standard usb audio I/O on its interfaces 2&3, which already works with the current snd-usb-audio driver, except for the audio input from the synth. I'm going to work on it when I have some time. Signed-off-by: Sebastien Alaiwan <sebastien.alaiwan@gmail.com> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> (cosmetics, list terminator) Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-16ALSA: usb-audio: reduce MIDI packet size to work around broken firmwareClemens Ladisch1-3/+15
Extend the list of devices whose firmware does not expect more than one USB MIDI packet in one USB packet. bug report: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3752 Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@kernel.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-15ALSA: Echoaudio - Add suspend support #2Giuliano Pochini3-78/+222
This patch adds rearranges parts of the initialization code and adds suspend and resume callbacks. This patch adds suspend and resume callbacks. It also rearranges parts of the initialization code so it can be used in both the first initialization (when the module is loaded we also have to load default settings) and the resume callback (where we have to restore the previous settings). Signed-off-by: Giuliano Pochini <pochini@shiny.it> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-15ALSA: Echoaudio - Add suspend support #1Giuliano Pochini15-80/+115
Move the controls init code outside the init_hw() function because is must not be called during resume. This patch moves the code that initializes the card's controls with default valued from the init_hw() function into a separated set_mixer_defaults() function (one for each of the 16 supported cards). This change is necessary because during resume we must resurrect the hardware without losing the previous settings. set_mixer_defaults() must be called only once when the module is loaded. Signed-off-by: Giuliano Pochini <pochini@shiny.it> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-15ALSA: Echoaudio - Add firmware cache #2Giuliano Pochini2-4/+41
This patch implements a simple cache for the firmware files when CONFIG_PM is defined. This patch changes get_firmware(), free_firmware() and adds free_firmware_cache(). The first two functions implement a very simple cache and the latter is used to actually release all the stored firmwares when the module is unloaded. When CONFIG_PM is not enabled those functions act as before, that is free_firmware() releases the firmware immediately and free_firmware_cache() does nothing. Signed-off-by: Giuliano Pochini <pochini@shiny.it> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-15ALSA: Echoaudio - Add firmware cache #1Giuliano Pochini18-65/+69
Changes the way the firmware is passed through functions. When CONFIG_PM is enabled the firmware cannot be released because the driver will need it again to resume the card. With this patch the firmware is passed as an index of the struct firmware card_fw[] in place of a pointer. That same index is then used to locate the firmware in the firmware cache. Signed-off-by: Giuliano Pochini <pochini@shiny.it> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-12Merge branch 'fix/hda' of ↵Linus Torvalds1-6/+3
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: ALSA: hda - use WARN_ON_ONCE() for zero-division detection
2010-02-12ALSA: hda - use WARN_ON_ONCE() for zero-division detectionTakashi Iwai1-6/+3
Replace the zero-division warning message with WARN_ON_ONCE() per the advice by Linus. This shouldn't happen, but if it happens, it's possible that the bug happens often due to buggy IRQs. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-12Merge branch 'fix/hda' of ↵Linus Torvalds1-0/+6
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: ALSA: hda-intel: Avoid divide by zero crash
2010-02-12Merge remote branch 'alsa/devel' into topic/miscTakashi Iwai3-165/+273
2010-02-11ALSA: usbmixer - add possibility to remap dB valuesJaroslav Kysela2-55/+93
USB devices tends to represent dB ranges in different way than ALSA expects. Add possibility to override these values and add guessed values for SoundBlaster MP3+. Also rename 'Capture Input Source' control to 'Capture Source' for SoundBlaster MP3+ and Extigy. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-09ALSA: hda-intel: Avoid divide by zero crashJody Bruchon1-0/+6
On my AMD780V chipset, hda_intel.c can crash the kernel with a divide by zero for as-yet unknown reasons. A simple check for zero prevents it, though the problem that causes it remains. Since the workaround is harmless and won't affect anyone except victims of this bug, it should be safe; moreover, because this crash can be triggered by a user-mode application, there are denial of service implications on the systems affected by the bug without the patch. Signed-off-by: Jody Bruchon <jody@nctritech.com> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-09sound: use DEFINE_PCI_DEVICE_TABLEAlexey Dobriyan67-70/+70
Use DEFINE_PCI_DEVICE_TABLE() to make PCI device ids go to .devinit.rodata section, so they can be discarded in some cases, and make them const. Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-05Merge remote branch 'alsa/fixes' into for-linusTakashi Iwai4-41/+32
2010-02-05Merge branch 'fix/asoc' into for-linusTakashi Iwai1-0/+1
2010-02-05Merge branch 'fix/hda' into for-linusTakashi Iwai1-3/+18
2010-02-05ASoC: pandora: Add APLL supply to fix audio outputGrazvydas Ignotas1-0/+1
Pandora's external DAC is using 256*Fs output from the TWL4030 codec, and TWL4030 needs to have APLL enabled for it's 256*Fs output to function. Signed-off-by: Grazvydas Ignotas <notasas@gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-05ALSA: ice1724 - aureon - fix wm8770 volume offsetJaroslav Kysela1-5/+7
The volume register is from 0..0x7f and 0..0x1a range is mute. Also, fix mute combining in wm_vol_put(). The wrong behaviour was noticed by Peter Christensen. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-05ALSA: cosmetic: make hda intel interrupt name consistent with othersMaxim Levitsky1-1/+1
This renames the interrupt name in /proc/interrupt. HDA Intel -> hda_intel This also eliminates space from the name, probably helping some parsers. Don't think anybody depends on this name in userspace Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-05ALSA: hda - Delay switching to polling mode if an interrupt was missingMaxim Levitsky1-2/+17
My sound codec seems sometimes (very rarely) to omit interrupts (ALC268) However, interrupt mode still works. Thus if we get timeout, poll the codec once. If we get 3 such polls in a row, then switch to polling mode. This patch is maybe an bandaid, but this might be a workaround for hardware bug. Signed-off-by: Maxim Levitsky <maximlevitsky@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-05ALSA: ice1712: fix: lock samplerate when samplerate locking is enabledSebastien Alaiwan1-0/+9
I found that the sampling rate locking setting of the ice1712 sound driver was only half-respected : when the driver was locked to, let's say, 44100Hz, and a usermode app was requesting 48000Hz playback, the request was succesful although the soundcard would continue to run at 44100Hz. Here's a patch that will make those requests to fail. Signed-off-by: Sebastien Alaiwan <sebastien.alaiwan@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-04ALSA: ctxfi - fix PTP address initializationJaroslav Kysela3-36/+25
After hours of debugging, I finally found the reason why some source and runtime combination does not work. The PTP (page table pages) address must be aligned. I am not sure how much, but alignment to PAGE_SIZE is sufficient. Also, use ALSA's page allocation routines to ensure proper virtual -> physical address translation. Cc: <stable@kernel.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-02ALSA: dummy driver - add model parameterJaroslav Kysela1-110/+180
This is a cleanup for the dummy driver. The model kernel module parameter is introduced to select the soundcard emulation. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-02ALSA: trivial: sound seq ioctl dbg: print hexadecimal value padded with 0sThadeu Lima de Souza Cascardo1-1/+1
Instead of padding with blanks and printing "number=0x a", print "number=0x0a". Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-01Merge remote branch 'alsa/devel' into topic/miscTakashi Iwai2-6/+40
2010-02-01sound: control: fix minimum TLV lengthClemens Ladisch1-1/+1
Allow TLV blocks that do not have any values; the smallest possible TLV is an empty container or one where the information is only in the tag. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-01sound: control: actually allow TLV command accessClemens Ladisch1-2/+3
Creating a control with TLV_COMMAND access was not possible because snd_ctl_new1() forgot to include it in the mask of allowable access bits. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-01-31Merge branch 'fix/asoc' into for-linusTakashi Iwai2-2/+2
2010-01-31Merge branch 'fix/hda' into for-linusTakashi Iwai1-0/+1
2010-01-29ASoC: AM3517: ASoC driver not getting compiledAnuj Aggarwal1-1/+1
Commit 761c9d45 (ASoC: Fix build of OMAP sound drivers) changes CONFIG_MACH_OMAP3517EVM -> CONFIG_SND_OMAP_SOC_OMAP3517EVM in the Makefile. Whereas the config option defined in Kconfig is SND_OMAP_SOC_AM3517EVM. Because of this, ASoC driver for AM3517 was not getting compiled. Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-29ASoC: AIC23: Fixing writes to non-existing registers in resume functionAnuj Aggarwal1-1/+1
Commit e9ff5eb2 (Fixing infinite loop in resume path) uses wrong AIC23 register in resume function because of which register writes happen on some non-existing registers. Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-27ALSA: hda - Add an ASUS mobo to MSI blacklistTakashi Iwai1-0/+1
Sid Boyce reported that his machine locks up without enable_msi=0 option. This looks like another ASUS mobo with Nvidia combo. Reported-by: Sid Boyce <sboyce@blueyonder.co.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-27ALSA: pcm_native - fix runtime->boundary calculationJaroslav Kysela1-3/+36
The code in pcm_lib updating runtime->hw_ptr_interrupt expects that runtime->boundary is divisible with runtime->period_size. Thanks are going to Clemens Ladisch for the notice. Fix the runtime->boundary calculation using buffer_size * period_size as base and find a least common multiple for 32bit platforms when the expression might overflow. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-01-26Merge remote branch 'alsa/devel' into topic/miscTakashi Iwai4-5/+18
2010-01-26ALSA: pcm_lib - return back hw_ptr_interruptJaroslav Kysela3-4/+8
Clemens Ladisch noted for hw_ptr_removal in "cleanup & merge hw_ptr update functions" commit: "It is possible for the status/delay ioctls to be called when the sound card's pointer register alreay shows a position at the beginning of the new period, but immediately before the interrupt is actually executed. (This happens regularly on a SMP machine with mplayer.) When that happens, the code thinks that the position must be at least one period ahead of the current position and drops an entire buffer of data." Return back the hw_ptr_interrupt variable. The last interrupt pointer is always computed from the latest hw_ptr instead of tracking it separately (in this case all hw_ptr checks and modifications might influence also hw_ptr_interrupt and it is difficult to keep it consistent). Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-01-26ALSA: cs46xx: Fix cpu idling with resumeFlorian Zumbiehl1-0/+10
Make sure that capture DMA doesn't stay enabled after system resume as that potentially prevents the processor from entering deep sleep states. Signed-off-by: Florian Zumbiehl <florz@florz.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-25Merge branch 'fix/hda' into for-linusTakashi Iwai1-15/+22
2010-01-25ASoC: fix a memory-leak in wm8903Guennadi Liakhovetski1-1/+2
Remember to free the temporary register-cache. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
2010-01-25ALSA: hda - add possibility to choose speakers configuration for 4930gŁukasz Wojniłowicz1-0/+1
Now one can choose speaker configuration in e.g. PulseAudio mixer Signed-off-by: Łukasz Wojniłowicz <lukasz.wojnilowicz@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-21Merge branch 'topic/noncached-mmap' into topic/miscTakashi Iwai6-2/+19
2010-01-21ALSA: hda - AD1988 codec - fix SPDIF-input mixer initialization (unmute)Jaroslav Kysela1-1/+10
The SPDIF-input pin 0x1c is muted by default in hardware. Unmute appropriate pin to get captured samples instead zeros. Tested on Lenovo Thinkstation. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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