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2012-12-04ALSA: usb-audio: sync ep init fix for audioformat mismatchEldad Zack1-7/+99
Commit 947d299686aa9cc8aecf749d54e8475c6e498956 , "ALSA: snd-usb: properly initialize the sync endpoint", while correcting the initialization of the sync endpoint when opening just the data endpoint, prevents devices that has a sync endpoint, with a channel number different than that of the data endpoint, from functioning. Due to a different channel and period bytes count, attempting to initialize the sync endpoint will fail at the usb host driver. For example, when using xhci: cannot submit urb 0, error -90: internal error With this patch, if a sync endpoint has multiple audioformats, a matching audioformat is preferred. An audioformat must be found with at least one channel and support the requested sample rate and PCM format, otherwise the stream will not be opened. If the number of channels differ between the selected audioformat and the requested format, adjust the period bytes count accordingly. It is safe to perform the calculation on the basis of the channel count, since the requested PCM audio format and the rate must be supported by the selected audioformat. Cc: Jeffrey Barish <jeff_barish@earthlink.net> Cc: Daniel Mack <zonque@gmail.com> Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-04ALSA: usb-audio: Fix missing autopm for MIDI inputTakashi Iwai1-42/+46
The commit [88a8516a: ALSA: usbaudio: implement USB autosuspend] added the support of autopm for USB MIDI output, but it didn't take the MIDI input into account. This patch adds the following for fixing the autopm: - Manage the URB start at the first MIDI input stream open, instead of the time of instance creation - Move autopm code to the common substream_open() - Make snd_usbmidi_input_start/_stop() more robust and add the running state check Reviewd-by: Clemens Ladisch <clemens@ladisch.de> Tested-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-04ALSA: usb-audio: Avoid autopm calls after disconnectionTakashi Iwai1-1/+22
Add a similar protection against the disconnection race and the invalid use of usb instance after disconnection, as well as we've done for the USB audio PCM. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=51201 Reviewd-by: Clemens Ladisch <clemens@ladisch.de> Tested-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@vger.kernel.org> 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>
2012-11-26ALSA: 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>
2012-11-22Merge branch 'for-linus' into for-nextTakashi Iwai1-1/+1
2012-11-22ALSA: 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>
2012-11-21ALSA: 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>
2012-11-19Merge branch 'for-linus' into for-nextTakashi Iwai2-5/+9
Merge the recent HD-audio codec change for fixing recursive suspend calls. Conflicts: sound/pci/hda/hda_codec.c
2012-11-19Merge branch 'usb-midi-fix-3.7' of git://git.alsa-project.org/alsa-kprivate into for-linusTakashi Iwai1-2/+6
Merge a regression fix for USB MIDI on non-standard usb-audio drivers by Clemens.
2012-11-18ALSA: ua101, usx2y: fix broken MIDI outputClemens Ladisch1-2/+6
Commit 88a8516a2128 (ALSA: usbaudio: implement USB autosuspend) added autosuspend code to all files making up the snd-usb-audio driver. However, midi.c is part of snd-usb-lib and is also used by other drivers, not all of which support autosuspend. Thus, calls to usb_autopm_get_interface() could fail, and this unexpected error would result in the MIDI output being completely unusable. Make it work by ignoring the error that is expected with drivers that do not support autosuspend. Reported-by: Colin Fletcher <colin.m.fletcher@googlemail.com> Reported-by: Devin Venable <venable.devin@gmail.com> Reported-by: Dr Nick Bailey <nicholas.bailey@glasgow.ac.uk> Reported-by: Jannis Achstetter <jannis_achstetter@web.de> Reported-by: Rui Nuno Capela <rncbc@rncbc.org> Cc: Oliver Neukum <oliver@neukum.org> Cc: 2.6.39+ <stable@vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2012-11-17ALSA: usb-audio: use bitmap_weightJoe Perches1-6/+1
Use bitmap_weight to count the total number of bits set in bitmap. Signed-off-by: Joe Perches <joe@perches.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-14ALSA: usb-audio: Fix mutex deadlock at disconnectionTakashi Iwai1-3/+3
The recent change for USB-audio disconnection race fixes introduced a mutex deadlock again. There is a circular dependency between chip->shutdown_rwsem and pcm->open_mutex, depicted like below, when a device is opened during the disconnection operation: A. snd_usb_audio_disconnect() -> card.c::register_mutex -> chip->shutdown_rwsem (write) -> snd_card_disconnect() -> pcm.c::register_mutex -> pcm->open_mutex B. snd_pcm_open() -> pcm->open_mutex -> snd_usb_pcm_open() -> chip->shutdown_rwsem (read) Since the chip->shutdown_rwsem protection in the case A is required only for turning on the chip->shutdown flag and it doesn't have to be taken for the whole operation, we can reduce its window in snd_usb_audio_disconnect(). Reported-by: Jiri Slaby <jslaby@suse.cz> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-13ALSA: usb-audio: Add quirk for Focusrite Scarlett 18i6Martin Schwenke1-0/+52
Probing this device currently fails in snd_usb_audio_probe() because the call to snd_usb_create_mixer() fails. This is due to unknown or non-standard interface descriptor subtypes in parse_audio_unit(): usbaudio: unit 51: unexpected type 0x09 snd-usb-audio: probe of 1-8:1.0 failed with error -5 Some people are working around this by recompiling usb-audio with the call to snd_usb_create_mixer() commented out. It would be nice to avoid that. While the best idea would be to look into the mixer creation failure, a reasonable short-term solution is to use quirks to only probe the trouble-free interfaces. This allows audio and MIDI interfaces to be used without any obvious issues. Interface 0 is the main one to ignore. It contains lots of control-fu, including the unexpected interface descriptor subtypes. Interface 5 is for firmware updates and I'm not sure how to get support for this. Interface 3 is some sort of control interface that I don't understand: Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 9 bInCollection 1 baInterfaceNr( 0) 1 Signed-off-by: Martin Schwenke <martin@meltin.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-08Merge branch 'for-linus' into for-nextTakashi Iwai3-0/+17
2012-11-08ALSA: usb-audio: Fix crash at re-preparing the PCM streamTakashi Iwai3-0/+17
There are bug reports of a crash with USB-audio devices when PCM prepare is performed immediately after the stream is stopped via trigger callback. It turned out that the problem is that we don't wait until all URBs are killed. This patch adds a new function to synchronize the pending stop operation on an endpoint, and calls in the prepare callback for avoiding the crash above. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181 Reported-and-tested-by: Artem S. Tashkinov <t.artem@lycos.com> Cc: <stable@vger.kernel.org> [v3.6] Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30Merge branch 'for-linus' into for-nextTakashi Iwai8-55/+141
... for migrating the core changes for USB-audio disconnection fixes
2012-10-30ALSA: usb-audio: Fix races at disconnection in mixer_quirks.cTakashi Iwai1-7/+51
Similar like the previous commit, cover with chip->shutdown_rwsem and chip->shutdown checks. Reported-by: Matthieu CASTET <matthieu.castet@parrot.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30ALSA: usb-audio: Use rwsem for disconnect protectionTakashi Iwai4-17/+21
Replace mutex with rwsem for codec->shutdown protection so that concurrent accesses are allowed. Also add the protection to snd_usb_autosuspend() and snd_usb_autoresume(), too. Reported-by: Matthieu CASTET <matthieu.castet@parrot.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30ALSA: usb-audio: Fix races at disconnectionTakashi Iwai5-41/+79
Close some races at disconnection of a USB audio device by adding the chip->shutdown_mutex and chip->shutdown check at appropriate places. The spots to put bandaids are: - PCM prepare, hw_params and hw_free - where the usb device is accessed for communication or get speed, in mixer.c and others; the device speed is now cached in subs->speed instead of accessing to chip->dev The accesses in PCM open and close don't need the mutex protection because these are already handled in the core PCM disconnection code. The autosuspend/autoresume codes are still uncovered by this patch because of possible mutex deadlocks. They'll be covered by the upcoming change to rwsem. Also the mixer codes are untouched, too. These will be fixed in another patch, too. Reported-by: Matthieu CASTET <matthieu.castet@parrot.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-25ALSA: sound/usb: remove CONFIG_EXPERIMENTALKees Cook1-1/+1
This config item has not carried much meaning for a while now and is almost always enabled by default. As agreed during the Linux kernel summit, remove it. Signed-off-by: Kees Cook <keescook@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-23ALSA: usb-audio: Add quirk for Reloop PlayDidier Villevalois1-0/+41
The Reloop Audio needs a fixed endpoint quirk with S24_3LE format and UAC_EP_CS_ATTR_SAMPLE_RATE attribute. Signed-off-by: Didier Villevalois <ptitjes@free.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-21ALSA: usb-audio: USB audio quirk for Roland VG-99 advanced modePete Leigh1-0/+34
Without this quirk the VG-99 will work in standard mode (set under USB on System menu page 2) giving 16 bits at 44.1 Khz audio in/out but no midi, and is not recognised when set to advanced mode. After applying this, I can also use the VG-99 in advanced mode: 24 24 bits audio in/out at 44.1 Khz, and midi in/out. Sysex is so far untested. In standard mode, the device appears with ID 0x00b3, so the behaviour isn't affected by this quirk. Thanks to Clemens Ladisch for simplifying and correcting my initial attempt! Signed-off-by: Pete Leigh <pete.leigh@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-21ALSA: snd-usb: remove unused variable in init_pitch_v2()Wei Yongjun1-3/+0
The variable ep is initialized but never used otherwise, so remove the unused variable. dpatch engine is used to auto generate this patch. (https://github.com/weiyj/dpatch) Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-12Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/soundLinus Torvalds1-0/+46
Pull sound updates #2 from Takashi Iwai: "This update contains a few cleanup works, regression/stable fixes gathered since the last pull request. - Clean up with generic hd-audio jack handling code by David Henningsson - A few regression fixes for standardized HD-audio auto-parser - Misc clean-up and small fixes" * tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: hda - do not detect jack on internal speakers for Realtek ALSA: hda - Fix missing beep on ASUS X43U notebook ALSA: hda - Remove AZX_DCAPS_POSFIX_COMBO ALSA: hda - Warn an allocation for an uninitialized array ALSA: hda/cirrus - Add missing init/free of hda_gen_spec ALSA: hda - Fix memory leaks at error path in patch_cirrus.c ALSA: hda - Add missing hda_gen_spec to struct via_spec ALSA: hda - remove "Mic Jack Mode" for headset jacks (Latitude Exx30) ALSA: hda - make Cirrus codec use generic unsol event handler ALSA: hda - make VIA codec use generic unsol event handler ALSA: hda - Remove dead GPIO code for VIA codec ALSA: usb-audio: Add TASCAM US122 MKII playback
2012-10-09mm: kill vma flag VM_RESERVED and mm->reserved_vm counterKonstantin Khlebnikov3-3/+3
A long time ago, in v2.4, VM_RESERVED kept swapout process off VMA, currently it lost original meaning but still has some effects: | effect | alternative flags -+------------------------+--------------------------------------------- 1| account as reserved_vm | VM_IO 2| skip in core dump | VM_IO, VM_DONTDUMP 3| do not merge or expand | VM_IO, VM_DONTEXPAND, VM_HUGETLB, VM_PFNMAP 4| do not mlock | VM_IO, VM_DONTEXPAND, VM_HUGETLB, VM_PFNMAP This patch removes reserved_vm counter from mm_struct. Seems like nobody cares about it, it does not exported into userspace directly, it only reduces total_vm showed in proc. Thus VM_RESERVED can be replaced with VM_IO or pair VM_DONTEXPAND | VM_DONTDUMP. remap_pfn_range() and io_remap_pfn_range() set VM_IO|VM_DONTEXPAND|VM_DONTDUMP. remap_vmalloc_range() set VM_DONTEXPAND | VM_DONTDUMP. [akpm@linux-foundation.org: drivers/vfio/pci/vfio_pci.c fixup] Signed-off-by: Konstantin Khlebnikov <khlebnikov@openvz.org> Cc: Alexander Viro <viro@zeniv.linux.org.uk> Cc: Carsten Otte <cotte@de.ibm.com> Cc: Chris Metcalf <cmetcalf@tilera.com> Cc: Cyrill Gorcunov <gorcunov@openvz.org> Cc: Eric Paris <eparis@redhat.com> Cc: H. Peter Anvin <hpa@zytor.com> Cc: Hugh Dickins <hughd@google.com> Cc: Ingo Molnar <mingo@redhat.com> Cc: James Morris <james.l.morris@oracle.com> Cc: Jason Baron <jbaron@redhat.com> Cc: Kentaro Takeda <takedakn@nttdata.co.jp> Cc: Matt Helsley <matthltc@us.ibm.com> Cc: Nick Piggin <npiggin@kernel.dk> Cc: Oleg Nesterov <oleg@redhat.com> Cc: Peter Zijlstra <a.p.zijlstra@chello.nl> Cc: Robert Richter <robert.richter@amd.com> Cc: Suresh Siddha <suresh.b.siddha@intel.com> Cc: Tetsuo Handa <penguin-kernel@I-love.SAKURA.ne.jp> Cc: Venkatesh Pallipadi <venki@google.com> Acked-by: Linus Torvalds <torvalds@linux-foundation.org> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2012-10-09Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/soundLinus Torvalds11-67/+216
Pull sound updates from Takashi Iwai: "This contains pretty many small commits covering fairly large range of files in sound/ directory. Partly because of additional API support and partly because of constantly developed ASoC and ARM stuff. Some highlights: - Introduced the helper function and documentation for exposing the channel map via control API, as discussed in Plumbers; most of PCI drivers are covered, will follow more drivers later - Most of drivers have been replaced with the new PM callbacks (if the bus is supported) - HD-audio controller got the support of runtime PM and the support of D3 clock-stop. Also changing the power_save option in sysfs kicks off immediately to enable / disable the power-save mode. - Another significant code change in HD-audio is the rewrite of firmware loading code. Other than that, most of changes in HD-audio are continued cleanups and standardization for the generic auto parser and bug fixes (HBR, device-specific fixups), in addition to the support of channel-map API. - Addition of ASoC bindings for the compressed API, used by the mid-x86 drivers. - Lots of cleanups and API refreshes for ASoC codec drivers and DaVinci. - Conversion of OMAP to dmaengine. - New machine driver for Wolfson Microelectronics Bells. - New CODEC driver for Wolfson Microelectronics WM0010. - Enhancements to the ux500 and wm2000 drivers - A new driver for DA9055 and the support for regulator bypass mode." Fix up various arm soc header file reorg conflicts. * tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits) ALSA: hda - Add new codec ALC283 ALC290 support ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls ALSA: hda - fix indices on boost volume on Conexant ALSA: aloop - add locking to timer access ALSA: hda - Fix hang caused by race during suspend. sound: Remove unnecessary semicolon ALSA: hda/realtek - Fix detection of ALC271X codec ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310 ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event ALSA: hda - make a generic unsol event handler ASoC: codecs: Add DA9055 codec driver ASoC: eukrea-tlv320: Convert it to platform driver ALSA: ASoC: add DT bindings for CS4271 ASoC: wm_hubs: Ensure volume updates are handled during class W startup ASoC: wm5110: Adding missing volume update bits ASoC: wm5110: Add OUT3R support ASoC: wm5110: Add AEC loopback support ASoC: wm5110: Rename EPOUT to HPOUT3 ASoC: arizona: Add more clock rates ASoC: arizona: Add more DSP options for mixer input muxes ...
2012-10-08ALSA: usb-audio: Add TASCAM US122 MKII playbackOto Petřík1-0/+46
Added quirk to provide at least playback-only support. Signed-off-by: Oto Petrik <oto.petrik@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-27ALSA: snd-usb: fix next_packet_size calls for pause caseDaniel Mack1-1/+7
Also fix the calls to next_packet_size() for the pause case. This was missed in 245baf983 ("ALSA: snd-usb: fix calls to next_packet_size"). Signed-off-by: Daniel Mack <zonque@gmail.com> Reviewed-by: Takashi Iwai <tiwai@suse.de> Reported-and-tested-by: Christian Tefzer <ctrefzer@gmx.de> Cc: stable@kernel.org [ Taking directly because Takashi is on vacation - Linus ] Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2012-09-20ALSA: usb - disable broken hw volume for Tenx TP6911David Henningsson1-0/+7
While going through Ubuntu bugs, I discovered this patch being posted and a confirmation that the patch works as expected. Finding out how the hw volume really works would be preferrable to just disabling the broken one, but this would be better than nothing. Credit: sndfnsdfin (qawsnews) BugLink: https://bugs.launchpad.net/bugs/559939 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-19ALSA: usb-audio: Avoid unnecessary EP setups in prepareTakashi Iwai3-3/+10
The recent fix for USB suspend breakage moved the code to set up EP from hw_params to prepare, but it means also the EP setup might be called multiple times unnecessarily because the prepare callback can be called multiple times without starting the stream (e.g. OSS emulation). This patch adds a new flag to struct snd_usb_substream indicating whether the setup of EP is required, and do it only when necessary, i.e. right after hw_params or suspend. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-19ALSA: usb-audio: Move configuration to prepare.Dylan Reid2-60/+76
Move interface and endpoint configuration from hw_params to prepare callback. During system suspend/resume when the USB device power isn't cycled the interface and endpoint configuration need to be set before audio playback can continue. Resume involves another call to prepare but not to hw_params, moving it here allows a playing stream to continue after resume. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-19ALSA: usb-audio: Don't require hw_params in endpoint.Dylan Reid3-17/+35
Change the interface to configure an endpoint so that it doesn't require a hw_params struct. This will allow it to be called from prepare instead of hw_params, configuring it after system resume. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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