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2012-11-30ALSA: hda - Add a helper function for simple enum kcontrolsTakashi Iwai5-48/+44
The same type of code is being used in multiple places in various codec drivers, so put it as a core library. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-30ALSA: hda - Pass errors properly in alc_auto_check_switches()Takashi Iwai1-22/+37
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-30ALSA: hda - Refactor alc_kcontrol_new() usagesTakashi Iwai1-28/+18
Allocate the name string and assign the structure in alc_kcontrol_new() itself to reduce the code. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-30ALSA: hda - Use standard sort function in hda_auto_parser.cTakashi Iwai1-59/+47
Just refactoring, no functional changes. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29ALSA: usb - Don't create "Speaker" mixer controls on headphones and headsetsDavid Henningsson1-0/+30
A lot of headsets/headphones have a "Speaker" mixer control. This confuses PulseAudio to think it is a speaker instead of a headphone/headset. Therfore, we rename it to "Headphone". We determine if something is a headphone similar to how udev determines form factor (see 78-sound-card.rules). BugLink: https://bugs.launchpad.net/bugs/1082357 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29ALSA: usb-audio: FT C400 sync playback EP to capture EPEldad Zack1-0/+13
The playback endpoint uses implicit feedback mode, similar to the M-Audio FTU. Like with the FTU, we need to associate the sync pipe ourselves. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29ALSA: usb-audio: Fast Track C400 mixer controlsEldad Zack1-0/+176
Add a mixer quirks for the M-Audio Fast Track C400 and create the following: * Volume controls * Effect Type (reusing FTU controls) * Effect Volume * Effect Send/Return * Effect Program * Effect Feedback Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29ALSA: usb-audio: Fast Track C400 mixer rangesEldad Zack1-0/+27
Add ranges for various Fast Track C400 controls, as observed while using the vendor's mixer control software (res values are an estimation). Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29ALSA: usb-audio: M-Audio Fast Track C400 quirks tableEldad Zack1-0/+71
Adds a quirks table for the M-Audio Fast Track C400. Thanks to Clemens Ladisch <clemens@ladisch.de> for pointing out that the table must be sorted. Based on the following patch from the alsa-devel list: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-May/051676.html See also: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-April/051219.html Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29ALSA: usb-audio: parameterize FTU effect unit controlEldad Zack1-8/+16
Adds the unit ID and the control as parameters to the creation of the effect unit control for the M-Audio Fast Track Ultra. This allows the code to be shared with other devices that use different unit ID and control, such as the M-Audio Fast Track C400. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29ALSA: usb-audio: skip UAC2 EFFECT_UNITEldad Zack1-1/+12
Current code mishandles the case where the device is a UAC2 and the bDescriptorSubtype is a UAC2 Effect Unit (0x07). It tries to parse it as a Processing Unit (which is similar to two other UAC1 units with overlapping subtypes), but since the structure is different (See: 4.7.2.10, 4.7.2.11 in UAC2 standard), the parsing is done incorrectly and prevents the device from initializing. For now, just ignore the unit. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29ALSA: usb-audio: add control index offsetEldad Zack3-1/+20
Currently, channel IDs exceeding 31 (0x1f) cannot be used. The channel ID is derived from the cmask. Extending cmask to a 64-bit type would only allow it to go up to 63 (0x3f). Some devices have channel IDs exceeding that as well. To address that, add an offset to the mixer element which is then accounted for in the UAC set/get functions. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29ALSA: usb-audio: use sender stride for implicit feedbackEldad Zack1-3/+6
For implicit feedback endpoints, the number of bytes for each packet is matched by the corresponding synchronizing endpoint. The size is calculated by taking the actual size and dividing it by the stride - currently by the endpoint's stride, but we should use the synchronization source's stride. This is evident when the number of channels differ between the synchronization source and the implicitly fed-back endpoint, as with M-Audio Fast Track C400 - the synchronization source (capture) has 4 channels, while the implicit feedback mode endpoint has 6. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29ALSA: usb-audio: replace hardcoded value with constEldad Zack1-1/+1
In this context, 0x01 is USB_ENDPOINT_XFER_ISOC. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-28ALSA: hda - Call snd_array_init() early and only onceTakashi Iwai6-103/+105
This is a preliminary patch for introducing a protection to access races of snd_array instances. Call snd_array_init() appropriately at the initialization time and don't call it twice. Also the allocations of codec-spec structs are cleaned up by helper functions in patch_sigmatel.c and patch_analog.c. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-26ALSA: usb-audio: add channel map supportTakashi Iwai2-5/+227
Add the support for channel maps of the PCM streams on USB audio devices. The channel map information is already found in ChannelConfig descriptor entries, which haven't been referred until now. Each chmap entry is added to audioformat list entry and copied to TLV dynamically instead of creating a whole chmap array. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-23ALSA: usb-audio: fix delay account during pauseTakashi Iwai1-1/+17
When a playback stream is paused, the stream isn't actually stopped, thus we still need to take care of the in-flight data amount for the delay calculation. Otherwise the value of subs->last_delay is no longer reliable and can give a bogus value after resuming from pause. This will result in "delay: estimated XX, actual YY" error messages. Also, during pause after all in flight data are processed (i.e. last_delay = 0), we don't have to calculate the actual delay from the current frame. Give a short path in such a case. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-23ALSA: usb-audio: ignore delay calculation for capture streamTakashi Iwai1-1/+2
It doesn't make sense to calculate the delay for capture streams in the current implementation. It's always zero, so we should skip the computation in snd_usb_pcm_pointer() in the case of capture. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-23ALSA: hda - Add a fixup for internal mic on MacBook Pro 8,1Takashi Iwai1-0/+14
The internal mic on MBP81 gives only the right channel, and the left channel is static. Add a verb to fix the ADC2 channel mode to expand mono right to stereo. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=50781 Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-23ALSA: hda - Add missing verb applications in patch_cirrus.cTakashi Iwai1-15/+2
We forgot to apply the fixup verbs in cs_init(). But adding the fixup verbs will break mbp101 fixup that has been fixed recently again, since the mbp101 fixup contains the wrong verbs to override. So these bogus verbs must be removed, too. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-23ALSA: hda - Move static ADC setup for CS4206 to init verbsTakashi Iwai1-3/+3
It's constant, so better to be put in the static init array. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-23Merge branch 'for-linus' into for-nextTakashi Iwai8-52/+48
2012-11-23ALSA: snd-usb: properly initialize the sync endpointDaniel Mack1-1/+1
Jeffrey Barish reported an obvious bug in the pcm part of the usb-audio driver which causes the code to not initialize the sync endpoint from configure_endpoint(). Reported-by: Jeffrey Barish <jeff_barish@earthlink.net> Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: stable@kernel.org [3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-23ALSA: emu10k1: Add missing ifdef for emu->suspend referenceTakashi Iwai1-0/+2
It's defined only for PM. Reported-by: Fengguang Wu <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-23ALSA: hdsp: Add missing inclusion of linux/vmalloc.hTakashi Iwai1-0/+1
Reported-by: Fengguang Wu <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-22ALSA: emu10k1: don't update firmware during suspend/resumeTakashi Iwai2-0/+6
Add a flag to suppress the update in emu1010_firmware_thread() during suspend/resume. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-22ALSA: emu10k1: cache emu1010 firmwareTakashi Iwai1-52/+36
Instead of calling request_firmware() at each time, keep the obtained firmware internally and reuse it. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-22ALSA: hdsp: improve firmware cachingTakashi Iwai2-28/+35
As the recent firmware code tries to reread and cache the firmware by itself, it's even better to keep the struct firmware data instead of keeping a local copy. Also, it makes little sense to disable the fw loader for this driver, so added the explicit dependency, too. Last, but not least, allocate the firmware data loaded via ioctl in vmalloc'ed buffer instead, as the firmware size isn't that small. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-22ALSA: ymfpci: Use kmalloc for register buffer for PMTakashi Iwai1-3/+3
Th buffer to save registers for PM is enough small for kmalloc(), not necessary to use vmalloc(). Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-22ALSA: hda - Don't release firmware when CONFIG_PM is setTakashi Iwai1-0/+2
The new firmware code tries to re-read the formerly read firmware files before suspend. Thus it's wiser to keep the "patch" firmware in the driver for avoiding this unnecessary re-reading. Of course, this will consume a bit of memory for unused stuff, but the patch fw is supposed to be fairly small, so it's more benefit in the end. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-22ALSA: vx: hard dependency on the standard fw loaderTakashi Iwai2-139/+1
Yet again like previous two commits, drop the old hwdep user-space firmware code from vx driver (snd-vxpocket and snd-vx222). Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-22ALSA: pcxhr: hard dependency on the standard fw loaderTakashi Iwai2-86/+1
Like the previous commit for mixart, drop the home-baked fw loader code. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-22ALSA: mixart: hard dependency on the standard fw loaderTakashi Iwai2-76/+1
It makes no longer sense to keep the old hwdep user-space firmware loading, which has been deprecated since ages ago. Just add a hard dependency on CONFIG_FW_LOADER and drop the useless code. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-22ALSA: hda - Don't overwrite the pin default configsTakashi Iwai1-42/+3
Since we keep the pin default config values anyway internally, we don't have to set the values in the codec. This patch removes the code writing the pincfg values. As a gratis bonus, we can remove also the code restoring the original pincfg values at PM resume or module free. This will give us more benefit, as it can reduce the unnecessary power-up of codecs. This won't change the driver functionality. The only difference would be that the codec proc file will show the original pincfg values instead of the actually referred values. The actually referred values can be determined from sysfs *_pin_configs files. (Also hda-emu was updated to follow this change.) Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21ALSA: hda - Remove shutup calls in free callbacksTakashi Iwai3-4/+0
The free callback is called at the state where no extra verbs are executed, thus calling *_shutup() is useless, as it's checking the shutdown flag. Remove such superfluous calls. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21ALSA: hda - Use standard helper functions in patch_hdmi.cTakashi Iwai1-4/+3
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21ALSA: usb-audio: process pending stop at PCM hw_free and closeTakashi Iwai1-2/+2
PCM hw_free and close should wait until all the pending stop operations have been finished. Basically only PCM trigger callback should use non-wait calls. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21ALSA: usb-audio: stop both data and sync endpoints asynchronouslyTakashi Iwai3-9/+13
As we are stopping the endpoints asynchronously now, it's better to trigger the stop of both data and sync endpoints and wait for pending stopping operations, instead of the sequential trigger-and-wait procedure. So the wait argument in snd_usb_endpoint_stop() is dropped, and it's expected that the caller synchronizes explicitly by calling snd_usb_endpoint_sync_pending_stop(). (Actually there is only one place calling this, so it was safe to change.) Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21ALSA: usb-audio: simplify endpoint deactivation codeTakashi Iwai1-16/+7
For further code simplification, drop the conditional call for usb_kill_urb() with can_wait argument in deactivate_urbs(), and use only usb_unlink_urb() and wait_clear_urbs() pairs. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21ALSA: usb-audio: simplify snd_usb_endpoint_start/stop argumentsTakashi Iwai3-26/+21
Reduce the redundant arguments for snd_usb_endpoint_start() and snd_usb_endpoint_stop(). Also replaced from int to bool. No functional changes by this commit. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21ALSA: usb-audio: Deprecate async_unlink optionTakashi Iwai3-9/+1
The async unlink behavior has been working over years. The option was provided only as a workaround for 2.4.x kernel. Let's get rid of it. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21ALSA: usb-audio: Return meaningful error codes instead of -1 in format.cSachin Kamat1-5/+5
Also, silences the following smatch warning: sound/usb/format.c:170 parse_audio_format_rates_v1() warn: returning -1 instead of -ENOMEM is sloppy Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21ALSA: usb/6fire: Fix potential NULL pointer dereference in comm.cSachin Kamat1-1/+2
'rt' was dereferenced before the NULL check. Moved the code after the check. Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21ALSA: pcm: Fix return code in pcm_native.cSachin Kamat1-1/+1
Return the value obtained from snd_pcm_hw_constraint_minmax() instead of -EINVAL. Silences the following smatch warning: sound/core/pcm_native.c:2003 snd_pcm_hw_constraints_complete() info: why not propagate 'err' from snd_pcm_hw_constraint_minmax() instead of -22? Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21ALSA: PCM: Remove redundant null check before kfreeSachin Kamat1-2/+1
kfree on a null pointer is a no-op. Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21ALSA: hda - Cirrus: Correctly clear line_out_pins when moving to speakerDavid Henningsson1-0/+1
If this array is not cleared, the jack related code later might fail to create "Internal Speaker Phantom Jack" on Dell Inspiron 3420 and Dell Vostro 2420. BugLink: https://bugs.launchpad.net/bugs/1076840 Cc: stable@vger.kernel.org (3.6+) Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21ALSA: hda - Add support for Realtek ALC292David Henningsson1-0/+1
We found a new codec ID 292, and that just a simple quirk would enable sound output/input on this ALC292 chip. BugLink: https://bugs.launchpad.net/bugs/1081466 Cc: stable@vger.kernel.org Tested-by: Acelan Kao <acelan.kao@canonical.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21Merge tag 'asoc-3.7' of ↵Takashi Iwai5-51/+45
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v3.7 The biggest batch of fixes here is the Kirkwood DMA fixes, plus a couple of other small fixes.
2012-11-21Merge branches 'fix/arizona', 'fix/cs4271', 'fix/kirkwood' and 'fix/samsung' ↵Mark Brown47-199/+468
of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into tmp
2012-11-21ASoC: kirkwood-i2s: more pause-mode fixesRussell King1-9/+0
Don't even momentarily set the pause status when starting the channel; if we do, we should check the busy bit to ensure that we comply with the spec. In any case, it isn't necessary; we will not active on a START event so there is no need to pause the DMA. Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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