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2010-12-06sound: Fixed line limit issue in sound/ac97_bus.cJeffrin Jose
This is a patch to the sound/ac97_bus.c file that fixes up a 80 character line limit issue found by the checkpatch.pl tool. Signed-off-by: Jeffrin Jose <ahiliation@yahoo.co.in> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-12-06ALSA: oxygen: update hardware commentsClemens Ladisch
Reformat and update the comments that describe the hardware connections on the various models. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-12-06ALSA: oxygen: show correct package IDClemens Ladisch
Instead of the hardcoded "CMI8788", show the actual chip name. Note: This is neither what the chip is (it's always the same), nor what the chip is actually called. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-12-06ALSA: oxygen: allow to dump codec registersClemens Ladisch
To help with debugging, add the registers of the model-specific codecs to the controller and AC97 register dump in the proc file. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-12-06ALSA: virtuoso: fix front panel routing for D1/DX/ST(X)Clemens Ladisch
The "Front Panel" switch on the Xonar D1/DX actually switches only the output direction, so mark it appropriately. The front panel microphone is controlled by the FMIC2MIC bit of the CM9780. It was unconditionally enabled on the D1/DX and never set on the ST(X); add a control for it. Selecting the front panel microphone as source does not actually disable the microphone jack, but this is bug-compatible with the Windows driver, and users rely on it. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-12-06ALSA: virtuoso: add HDMI enable switch for HDAV1.3Clemens Ladisch
The GPIO bit that enables analog output on the Xonar HDAV1.3 also disables the HDMI audio output, so we better add a switch for it. Hopefully, this is sufficient to make the HDMI output work. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-12-06ALSA: virtuoso: initialize unknown GPIO bitsClemens Ladisch
Initialize the configuration of some unknown GPIO output bits (that might not be used at all) to be the same as in the Windows driver, just to be sure. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-12-01ALSA: hdsp - Add support for RPM io boxFlorian Faber
Add support for the RME HDSP RPM IO box. Changes have been made in the identification of the IO box and the neccessary controls have been added. Signed-off-by: Florian Faber <faberman@linuxproaudio.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-24ALSA: support module on-demand loading for seq and timerKay Sievers
If CONFIG_SND_DYNAMIC_MINORS is used, assign /dev/snd/seq and /dev/snd/timer the usual static minors, and export specific module aliases to generate udev module on-demand loading instructions: $ cat /lib/modules/2.6.33.4-smp/modules.devname # Device nodes to trigger on-demand module loading. microcode cpu/microcode c10:184 fuse fuse c10:229 ppp_generic ppp c108:0 tun net/tun c10:200 uinput uinput c10:223 dm_mod mapper/control c10:236 snd_timer snd/timer c116:33 snd_seq snd/seq c116:1 The last two lines instruct udev to create device nodes, even when the modules are not loaded at that time. As soon as userspace accesses any of these nodes, the in-kernel module-loader will load the module, and the device can be used. The header file minor calculation needed to be simplified to make __stringify() (supports only two indirections) in the MODULE_ALIAS macro work. This is part of systemd's effort to get rid of unconditional module load instructions and needless init scripts. Cc: Lennart Poettering <lennart@poettering.net> Signed-off-by: Kay Sievers <kay.sievers@vrfy.org> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22ALSA: timer: automatically load the high-resolution timerClemens Ladisch
Increase the default timer limit so that snd-hrtimer.ko can be automatically loaded when needed, e.g., when used as the default sequencer timer. This replaces the check for the obsolete CONFIG_SND_HPET. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22ALSA: pcm: optimize xrun detection in no-period-wakeup modeClemens Ladisch
Add a lightweight condition on top of the xrun checking so that we can avoid the division when the application is calling the update function often enough. Suggested-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22ALSA: pcm: detect xruns in no-period-wakeup modeClemens Ladisch
When period wakeups are disabled, successive calls to the pointer update function do not have a maximum allowed distance, so xruns cannot be detected with the pointer value only. To detect xruns, compare the actually elapsed time with the time that should have theoretically elapsed since the last update. When the hardware pointer has wrapped around due to an xrun, the actually elapsed time will be too big by about hw_ptr_buffer_jiffies. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22ALSA: oxygen: support for period wakeup disablingClemens Ladisch
Allow disabling period wakeup interrupts for all PCM streams. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22ALSA: hda-intel: support for period wakeup disablingClemens Ladisch
Allow disabling period wakeup interrupts for HDA PCM streams. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22ALSA: pcm: support for period wakeup disablingClemens Ladisch
This patch allows to disable period interrupts which are not needed when the application relies on a system timer to wake-up and refill the ring buffer. The behavior of the driver is left unchanged, and interrupts are only disabled if the application requests this configuration. The behavior in case of underruns is slightly different, instead of being detected during the period interrupts the underruns are detected when the application calls snd_pcm_update_avail, which in turns forces a refresh of the hw pointer and shows the buffer is empty. More specifically this patch makes a lot of sense when PulseAudio relies on timer-based scheduling to access audio devices such as HDAudio or Intel SST. Disabling interrupts removes two unwanted wake-ups due to period elapsed events in low-power playback modes. It also simplifies PulseAudio voice modules used for speech calls. To quote Lennart "This patch looks very interesting and desirable. This is something have long been waiting for." Support for this in hardware drivers is optional. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22Merge branch 'fix/misc' into topic/miscTakashi Iwai
2010-11-22ALSA: hda: Use hp-laptop quirk to enable headphones automute for Asus A52JDaniel T Chen
BugLink: https://launchpad.net/bugs/677652 The original reporter states that, in 2.6.35, headphones do not appear to work, nor does inserting them mute the A52J's onboard speakers. Upon inspecting the codec dump, it appears that the newly committed hp-laptop quirk will suffice to enable this basic functionality. Testing was done with an alsa-driver build from 2010-11-21. Reported-and-tested-by: Joan Creus Cc: <stable@kernel.org> [2.6.35+] Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22ALSA: snd-atmel-abdac: test wrong variableVasiliy Kulikov
After clk_get() pclk is checked second time instead of sample_clk check. Signed-off-by: Vasiliy Kulikov <segoon@openwall.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22ALSA: azt3328: period bug fix (for PA), add missing ACK on stop timerAndreas Mohr
. Fix PulseAudio "ALSA driver bug" issue (if we have two alternated areas within a 64k DMA buffer, then max period size should obviously be 32k only). Back references: http://pulseaudio.org/wiki/AlsaIssues http://fedoraproject.org/wiki/Features/GlitchFreeAudio . In stop timer function, need to supply ACK in the timer control byte. . Minor log output correction When I did my first PA testing recently, the period size bug resulted in quite precisely observeable half-period-based playback distortion. PA-based operation is quite a bit more underrun-prone (despite its zero-copy optimizations etc.) than raw ALSA with this rather spartan sound hardware implementation on my puny Athlon. Note that even with this patch, azt3328 still doesn't work for both cases yet, PA tsched=0 and tsched (on tsched=0 it will playback tiny fragments of periods, leading to tiny stuttering sounds with some pauses in between, whereas with timer-scheduled operation playback works fine - minus some quite increased underrun trouble on PA vs. ALSA, that is). Signed-off-by: Andreas Mohr <andi@lisas.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22ALSA: hda: Add Samsung R720 SSID for subwoofer pin fixupDaniel T Chen
BugLink: https://launchpad.net/bugs/677830 The original reporter states that the subwoofer does not mute when inserting headphones. We need an entry for his machine's SSID in the subwoofer pin fixup list, so add it there (verified using hda_analyzer). Reported-and-tested-by: i-NoD Cc: <stable@kernel.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22ALSA: HDA: Create mixers on ALC887David Henningsson
BugLink: http://launchpad.net/bugs/669092 ALC887 does not have any volume control ability on the mixer NIDs, so put the volume controls on the dac NIDs instead. Without this patch, ALC887 users cannot use alsamixer at all. Cc: stable@kernel.org Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22ALSA: sound/pci/asihpi/hpioctl.c: Remove unnecessary casts of pci_get_drvdataJoe Perches
Signed-off-by: Joe Perches <joe@perches.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22ALSA: sound/core/pcm_lib.c: Remove unnecessary semicolonsJoe Perches
Signed-off-by: Joe Perches <joe@perches.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22ALSA: sound/ppc: Use printf extension %pR for struct resourceJoe Perches
Using %pR standardizes the struct resource output. Signed-off-by: Joe Perches <joe@perches.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22ALSA: ac97: Apply quirk for Dell Latitude D610 binding Master and Headphone ↵Daniel T Chen
controls BugLink: https://launchpad.net/bugs/669279 The original reporter states: "The Master mixer does not change the volume from the headphone output (which is affected by the headphone mixer). Instead it only seems to control the on-board speaker volume. This confuses PulseAudio greatly as the Master channel is merged into the volume mix." Fix this symptom by applying the hp_only quirk for the reporter's SSID. The fix is applicable to all stable kernels. Reported-and-tested-by: Ben Gamari <bgamari@gmail.com> Cc: <stable@kernel.org> [2.6.32+] Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-11ALSA: AT73C213: Rectify misleading comment.Peter Rosin
The Atmel SSC can divide by even numbers, not only powers of two. Signed-off-by: Peter Rosin <peda@axentia.se> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-11ALSA: sound/pci/ctxfi/ctpcm.c: Remove potential for use after freeJulia Lawall
In each function, the value apcm is stored in the private_data field of runtime. At the same time the function ct_atc_pcm_free_substream is stored in the private_free field of the same structure. ct_atc_pcm_free_substream dereferences and ultimately frees the value in the private_data field. But each function can exit in an error case with apcm having been freed, in which case a subsequent call to the private_free function would perform a dereference after free. On the other hand, if the private_free field is not initialized, it is NULL, and not invoked (see snd_pcm_detach_substream in sound/core/pcm.c). To avoid the introduction of a dangling pointer, the initializations of the private_data and private_free fields are moved to the end of the function, past any possible free of apcm. This is safe because the previous calls to snd_pcm_hw_constraint_integer and snd_pcm_hw_constraint_minmax, which take runtime as an argument, do not refer to either of these fields. In each function, there is one error case where apcm needs to be freed, and a call to kfree is added. The sematic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e,e1,e2,e3; identifier f,free1,free2; expression a; @@ *e->f = a ... when != e->f = e1 when any if (...) { ... when != free1(...,e,...) when != e->f = e2 * kfree(a) ... when != free2(...,e,...) when != e->f = e3 } // </smpl> Signed-off-by: Julia Lawall <julia@diku.dk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-11ALSA: sound/mixart: avoid redefining {readl,write}_{le,be} accessorsFlorian Fainelli
If the platform already provides a definition for these accessors do not redefine them. The warning was caught on MIPS. Signed-off-by: Florian Fainelli <florian@openwrt.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-11ALSA: HDA: Enable digital mic on IDT 92HD87BDavid Henningsson
BugLink: http://launchpad.net/bugs/673075 According to the datasheet of 92HD87B, there is a digital mic at nid 0x11, so enable it in order to be able to use the mic. Cc: stable@kernel.org Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-11sound/oss: Remove unnecessary casts of void ptrJesper Juhl
The [vk][cmz]alloc(_node) family of functions return void pointers which it's completely unnecessary/pointless to cast to other pointer types since that happens implicitly. This patch removes such casts from sound/oss/ Signed-off-by: Jesper Juhl <jj@chaosbits.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-11sound/oss/dev_table.c: Use vzallocJoe Perches
Signed-off-by: Joe Perches <joe@perches.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-03Merge branch 'fix/asoc' into for-linusTakashi Iwai
2010-11-03ASoC: tpa6130a2: Get rid of compile warning from tpa6130a2_powerJarkko Nikula
Patch "ASoC: tpa6130a2: Fix unbalanced regulator disables" introduced a compiler warning "‘ret’ may be used uninitialized in this function". Initialize ret to zero to get rid of it and making sure that the function does not return any random error code when the code is falling through. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-03ALSA: oxygen: add HiFier Serenade supportClemens Ladisch
Add support for the TempoTec/MediaTek HiFier Serenade sound card. The PCI ID was already there, but the driver handled it like the Fantasia model, which resulted in a dummy recording device. As a stereo output-only card, this model is to be handled exactly like the HG2PCI. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-03ALSA: oxygen: reorganize PCI IDsClemens Ladisch
Sort the PCI IDs so that they make logical sense. Also move the card name comments into this list because the model symbols should be (more) self-explanationary. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-03Merge branch 'for-2.6.37' of ↵Takashi Iwai
git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into fix/asoc
2010-11-03ALSA: oxygen: add Kuroutoshikou CMI8787-HG2PCI supportClemens Ladisch
Add support for the Kuroutoshikou CMI8787-HG2PCI sound card. [replaced non-latin letters in the patch by tiwai] Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-03ALSA: oxygen: merge HiFier driver into snd-oxygenClemens Ladisch
The snd-hifier driver contains more duplicated code than model-specific code, so it does not make sense for it to be a separate driver. Handling the two-channel output restriction can be easily done in the generic driver. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-03Merge branch 'fix/misc' into topic/miscTakashi Iwai
2010-11-03ALSA: hda - MacBookAir3,1(3,2) alsa supportEdgar (gimli) Hucek
This patch add support for the MacBookAir3,1 and MacBookAir3,2 to the alsa sound system. Signed-off-by: Edgar (gimli) Hucek <gimli@dark-green.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-02Merge commit 'v2.6.37-rc1' into for-2.6.37Mark Brown
2010-11-02ASoC: fix the building issue of missing codec field in 'struct snd_soc_card'Eric Miao
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-02ALSA: usb-audio - Support for Power/Status LED on Creative USB X-Fi S51Mandar Joshi
This patch adds support for Power/Status LED on Creative USB X-Fi S51. There is just one LED on the device. The LED can either be On or it can be set to Blink. There doesn't seem to be a way to switch it off. The control message to change LED status is similar to that of audigy2nx except that the index is to be set to 0 and value is 1 for Blink and 0 for On. The 'Power LED' control in alsamixer when muted will cause the LED to Blink continuously. When unmuted the LED will stay On. The Creative driver under Windows sets the LED to blink whenever audio is muted. This LED can be treated as the CMSS LED but I figured since there is just one LED, it should be treated as the Power LED. Is that alright? I've also changed the comment "Usb X-Fi" to "Usb X-Fi S51" as there are other external X-Fi devices from Creative like Usb X-Fi Go and Xmod. The volume knob and LED support patch doesn't apply to them. Signed-off-by: Mandar Joshi <emailmandar@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-02ALSA: asihpi - Unsafe memory management when allocating control cacheJesper Juhl
I noticed that sound/pci/asihpi/hpicmn.c::hpi_alloc_control_cache() does not check the return value from kmalloc(), which may fail. If kmalloc() fails we'll dereference a null pointer and things will go bad fast. There are two memory allocations in that function and there's also the problem that the first may succeed and the second may fail and nothing is done about that either which will also go wrong down the line. Signed-off-by: Jesper Juhl <jj@chaosbits.net> Acked-by: Eliot Blennerhassett <linux@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-01ASoC: Update WARN uses in wm_hubsJoe Perches
Add missing newlines. Signed-off-by: Joe Perches <joe@perches.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-01ASoC: Include cx20442 to SND_SOC_ALL_CODECSJarkko Nikula
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-01ASoC: Fix SND_SOC_ALL_CODECS typo for jz4740Jarkko Nikula
Include jz4740.c to SND_SOC_ALL_CODECS when the dependencies are met. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-01ASoC: Remove volatility from WM8900 POWER1 registerMark Brown
Not all bits can be read back from POWER1 so avoid corruption when using a read/modify/write cycle by marking it non-volatile - the only thing we read back from it is the chip revision which has diagnostic value only. We can re-add later but that's a more invasive change than is suitable for a bugfix. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Cc: stable@kernel.org
2010-11-01ALSA: lx6464es - make 1 bit signed bitfield unsignedTim Blechmann
converts a 1 bit signed bitfield to an unsigned. Reported-by: Dr. David Alan Gilbert <linux@treblig.org> Signed-off-by: Tim Blechmann <tim@klingt.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-01ALSA: cs46xx memory management fixes for cs46xx_dsp_spos_create()Jesper Juhl
When reading through sound/pci/cs46xx/dsp_spos.c I noticed a couple of things in cs46xx_dsp_spos_create(). It seems to me that we don't always free the various memory buffers we allocate and we also do some work (structure member assignment) early, that is completely pointless if some of the memory allocations fail and we end up just aborting the whole thing. I don't have hardware to test, so the patch below is compile tested only, but it makes the following changes: - Make sure we always free all allocated memory on failures. - Don't do pointless work assigning to structure members before we know all memory allocations, that may abort progress, have completed successfully. - Remove some trailing whitespace. Signed-off-by: Jesper Juhl <jj@chaosbits.net> Tested-by: Ondrej Zary <linux@rainbow-software.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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