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* ALSA: snd-usb: properly initialize the sync endpointDaniel Mack2012-11-221-1/+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>
* Merge branch 'usb-midi-fix-3.7' of git://git.alsa-project.org/alsa-kprivate ↵Takashi Iwai2012-11-191-2/+6
|\ | | | | | | | | | | | | into for-linus Merge a regression fix for USB MIDI on non-standard usb-audio drivers by Clemens.
| * ALSA: ua101, usx2y: fix broken MIDI outputClemens Ladisch2012-11-181-2/+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>
* | ALSA: usb-audio: Fix mutex deadlock at disconnectionTakashi Iwai2012-11-141-3/+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>
* | ALSA: usb-audio: Fix crash at re-preparing the PCM streamTakashi Iwai2012-11-083-0/+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>
* | ALSA: usb-audio: Fix races at disconnection in mixer_quirks.cTakashi Iwai2012-10-301-7/+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>
* | ALSA: usb-audio: Use rwsem for disconnect protectionTakashi Iwai2012-10-304-17/+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>
* | ALSA: usb-audio: Fix races at disconnectionTakashi Iwai2012-10-305-41/+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>
* | Merge tag 'sound-3.7' of ↵Linus Torvalds2012-10-121-0/+46
|\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound 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
| * | ALSA: usb-audio: Add TASCAM US122 MKII playbackOto Petřík2012-10-081-0/+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>
* | | mm: kill vma flag VM_RESERVED and mm->reserved_vm counterKonstantin Khlebnikov2012-10-093-3/+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>
* | | Merge tag 'sound-3.7' of ↵Linus Torvalds2012-10-0911-67/+216
|\| | | |/ |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound 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 ...
| * ALSA: usb - disable broken hw volume for Tenx TP6911David Henningsson2012-09-201-0/+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>
| * ALSA: usb-audio: Avoid unnecessary EP setups in prepareTakashi Iwai2012-09-193-3/+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>
| * ALSA: usb-audio: Move configuration to prepare.Dylan Reid2012-09-192-60/+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>
| * ALSA: usb-audio: Don't require hw_params in endpoint.Dylan Reid2012-09-193-17/+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>
| * ALSA: usb-audio: set period_bytes in substream.Dylan Reid2012-09-191-0/+2
| | | | | | | | | | | | | | | | Set the peiod_bytes member of snd_usb_substream. It was no longer being set, but will be needed to resume properly in a future commit. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * Merge branch 'for-linus' into for-nextTakashi Iwai2012-09-114-29/+72
| |\ | | | | | | | | | To merge HD-audio fixes back to 3.7 development line
| * | ALSA: snd-usb: Add quirks for Playback Designs devicesDaniel Mack2012-09-045-0/+49
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Playback Designs' USB devices have some hardware limitations on their USB interface. In particular: - They need a 20ms delay after each class compliant request as the hardware ACKs the USB packets before the device is actually ready for the next command. Sending data immediately will result in buffer overflows in the hardware. - The devices send bogus feedback data at the start of each stream which confuse the feedback format auto-detection. This patch introduces a new quirks hook that is called after each control packet and which adds a delay for all devices that match Playback Designs' USB VID for now. In addition, it adds a counter to snd_usb_endpoint to drop received packets on the floor. Another new quirks function that is called once an endpoint is started initializes that counter for these devices on their sync endpoint. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: Andreas Koch <andreas@akdesigninc.com> Supported-by: Demian Martin <demianm_1@yahoo.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: USB: Support for (original) Xbox CommunicatorMarko Friedemann2012-09-031-0/+53
| | | | | | | | | | | | | | | | | | | | | Added support for Xbox Communicator to USB quirks. Signed-off-by: Marko Friedemann <mfr@bmx-chemnitz.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usb-audio: Remove obsoleted fields in struct snd_usb_substreamTakashi Iwai2012-08-291-2/+0
| | | | | | | | | | | | | | | | | | | | | The two entries are duplicated in struct snd_usb_endpoint. Seems forgotten in the last clean-up. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge branch 'for-linus' into for-nextTakashi Iwai2012-08-202-4/+3
| |\ \ | | | | | | | | | | | | | | | | | | | | | | | | Conflicts: sound/pci/hda/hda_codec.c Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: print small buffers via %*ph[C]Andy Shevchenko2012-08-061-3/+2
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: snd-usb: fix next_packet_size calls for pause caseDaniel Mack2012-09-281-1/+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>
* | | ALSA: usb-audio: Fix bogus error messages for delay accountingTakashi Iwai2012-09-061-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The recent fix for the missing fine delayed time adjustment gives strange error messages at each start of the playback stream, such as delay: estimated 0, actual 352 delay: estimated 353, actual 705 These come from the sanity check in retire_playback_urb(). Before the stream is activated via start_endpoints(), a few silent packets have been already sent. And at this point the delay account is still in the state as if the new packets are just queued, so the driver gets confused and spews the bogus error messages. For fixing the issue, we just need to check whether the received packet is valid, whether it's zero sized or not. Reported-by: Markus Trippelsdorf <markus@trippelsdorf.de> Cc: <stable@vger.kernel.org> [v3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: snd-usb: fix cross-interface streaming devicesDaniel Mack2012-08-311-0/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit 68e67f40b ("ALSA: snd-usb: move calls to usb_set_interface") saved us some unnecessary calls to snd_usb_set_interface() but ignored the fact that there is at least one device out there which operates on two endpoint in different interfaces simultaniously. Take care for this by catching the case where data and sync endpoints are located on different interfaces and calling snd_usb_set_interface() between the start of the two endpoints. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Robert M. Albrecht <linux@romal.de> Cc: stable@kernel.org [v3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: snd-usb: fix calls to next_packet_sizeDaniel Mack2012-08-313-13/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In order to support devices with implicit feedback streaming models, packet sizes are now stored with each individual urb, and the PCM handling code which fills the buffers purely relies on the size fields now. However, calling snd_usb_audio_next_packet_size() for all possible packets in an URB at once, prior to letting the PCM code do its job does in fact not lead to the same behaviour than what the old code did: The PCM code will break its loop once a period boundary is reached, consequently using up less packets that it really could. As snd_usb_audio_next_packet_size() implements a feedback mechanism to the endpoints phase accumulator, the number of calls to that function matters, and when called too often, the data rate runs out of bounds. Fix this by making the next_packet function public, and call it from the PCM code as before if the packet data sizes are not defined. Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: stable@kernel.org [v3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: snd-usb: restore delay informationDaniel Mack2012-08-311-3/+26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Parts of commit 294c4fb8 ("ALSA: usb: refine delay information with USB frame counter") were unfortunately lost during the refactoring of the snd-usb driver in 3.5. This patch adds them back, restoring the correct delay information behaviour. Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: stable@kernel.org [3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: snd-usb: use list_for_each_safe for endpoint resourcesPavel Roskin2012-08-311-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | snd_usb_endpoint_free() frees the structure that contains its argument. Signed-off-by: Pavel Roskin <proski@gnu.org> Cc: stable@vger.kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: snd-usb: Fix URB cancellation at stream startDaniel Mack2012-08-303-11/+15
| |/ |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream") fixed a scheduling-while-atomic bug that happened when snd_usb_endpoint_start was called from the trigger callback, which is an atmic context. However, the patch breaks the idea of the endpoints reference counting, which is the reason why the driver has been refactored lately. Revert that commit and let snd_usb_endpoint_start() take care of the URB cancellation again. As this function is called from both atomic and non-atomic context, add a flag to denote whether the function may sleep. Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: stable@kernel.org [3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture streamTakashi Iwai2012-08-162-4/+3
|/ | | | | | | | | | | | | | | | A PCM capture stream on usb-audio causes a scheduling-while-atomic BUG, as reported in the bugzilla entry below. It's because snd_usb_endpoint_start() is called at first at trigger START for a capture stream, and this function contains the left-over EP deactivation codes. The problem doesn't happen for a playback stream because the function is called at PCM prepare time, which can sleep. This patch fixes the BUG by moving the EP deactivation code into the PCM prepare callback. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46011 Cc: <stable@vger.kernel.org> [v3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: fix clock source validity indexDaniel Mack2012-08-011-1/+2
| | | | | | | | | | | | | | | uac_clock_source_is_valid() uses the control selector value to access the bmControls bitmap of the clock source unit. This is wrong, as control selector values start from 1, while the bitmap uses all available bits. In other words, "Clock Validity Control" is stored in D3..2, not D5..4 of the clock selector unit's bmControls. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Andreas Koch <andreas@akdesigninc.com> Cc: stable@kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'topic/misc' into for-nextTakashi Iwai2012-07-182-57/+104
|\ | | | | | | Generic updates for sound 3.6
| * ALSA: snd-usb-caiaq: initialize card pointerDaniel Mack2012-06-271-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | Fixes the following warning: CC [M] sound/usb/caiaq/device.o sound/usb/caiaq/device.c: In function ‘snd_probe’: sound/usb/caiaq/device.c:500:16: warning: ‘card’ may be used uninitialized in this function [-Wmaybe-uninitialized] Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb-audio: Convert table to preferred C99 formatMark Hills2012-06-111-10/+64
| | | | | | | | | | Signed-off-by: Mark Hills <mark@pogo.org.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb-audio: Use a table of mixer controlsMark Hills2012-06-111-56/+49
| | | | | | | | | | | | | | | | Allow mixer controls to be provided clearly in a table, to avoid quantity of error checking at each use. Signed-off-by: Mark Hills <mark@pogo.org.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: snd-usb: move calls to usb_set_interfaceDaniel Mack2012-07-132-89/+41
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The rework of the snd-usb endpoint logic moved the calls to snd_usb_set_interface() into the snd_usb_endpoint implemenation. This changed the order in which these calls are issued to the device, and thereby caused regressions for some webcams. Fix this by moving the calls back to pcm.c for now to make it work again and use snd_usb_endpoint_activate() to really tear down all remaining URBs in the flight, consequently fixing another regression caused by USB packets on the wire after altsetting 0 has been selected. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: Philipp Dreimann <philipp@dreimann.net> Reported-by: Joseph Salisbury <joseph.salisbury@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Fix the first PCM interface assignmentTakashi Iwai2012-07-061-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | In the new PCM streaming logic, the interface number is assigned to usb stream instance (subs->interface) after the format and rate setups are succeeded, but some codes are still passing subs->interface as the reference to helper functions. This leads to initializing with an invalid iface number (-1). This patch replaces the wrong references with the ones from the target fmt correctly. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: add BOSS GT-100 supportClemens Ladisch2012-06-251-0/+30
| | | | | | | | | | | | | | Reported-by: John McFarland <mcfarljm@gmail.com> Tested-by: John McFarland <mcfarljm@gmail.com> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: snd_usb_audio: ignore ctrl errors on QuickCam Pro for NotebooksOleksij Rempel2012-06-201-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This webcam works mostly ok, exept with skype. Skype sends lots of ctrl messages to dynamically ajust record level. If for some reasons it pokes some error every thing goes broken: - first pulseaudio blocks sound for all apps - then video is reseted - then skype freez dmesg has lots of messages like: cannot set freq 16000 to ep 0x86" Setting ignore_ctl_error=1 fixes this problem. Signed-off-by: Oleksij Rempel <bug-track@fisher-privat.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: snd_usb_audio: ignore ctrl errors on QuickCam E3500Oleksij Rempel2012-06-201-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | if this cam is pluged in, pulse audio can't initiate capture device. dmesg has lots of messages like: "cannot set freq 16000 to ep 0x86" Setting ignore_ctl_error=1 fixes this problem. Signed-off-by: Oleksij Rempel <bug-track@fisher-privat.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: 6fire: use NULL instead of 0 for pointer assignmentDaniel Mack2012-06-181-1/+1
| | | | | | | | | | | | Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: Torsten Schenk <torsten.schenk@zoho.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: snd-usb: make snd_usb_substream_capture_trigger staticDaniel Mack2012-06-181-1/+2
| | | | | | | | | | Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: snd-usb: fix sync pipe checkDaniel Mack2012-06-181-7/+11
| | | | | | | | | | | | | | | | | | | | | | Fix a bogus sanity check for sync pipe in pcm.c. This flaw was introduced during the streaming logic refactorization. While at it, improve the error messages that are generated in such cases. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: <ben@b1c1l1.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Fix substream assignmentsTakashi Iwai2012-06-082-4/+4
|/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In 3.5 kernel, the endpoint is assigned dynamically for the substreams, but the PCM assignment still checks the presence of the endpoint pointer. This ended up in duplicated PCM substream creations at probing time, resulting in kernel warnings like: WARNING: at fs/proc/generic.c:586 proc_register+0x169/0x1a6() Pid: 1152, comm: modprobe Not tainted 3.5.0-rc1-00110-g71fae7e #2 Call Trace: [<ffffffff8102a400>] warn_slowpath_common+0x83/0x9c [<ffffffff8102a4bc>] warn_slowpath_fmt+0x46/0x48 [<ffffffff813829ad>] ? add_preempt_count+0x39/0x3b [<ffffffff811292f0>] proc_register+0x169/0x1a6 [<ffffffff8112962e>] create_proc_entry+0x74/0x8c [<ffffffffa018eb63>] snd_info_register+0x3e/0xc3 [snd] [<ffffffffa01fde2e>] snd_pcm_new_stream+0xb1/0x404 [snd_pcm] [<ffffffffa024861f>] snd_usb_add_audio_stream+0xd2/0x230 [snd_usb_audio] [<ffffffffa0241d33>] ? snd_usb_parse_audio_format+0x252/0x34f [snd_usb_audio] [<ffffffff810d6b17>] ? kmem_cache_alloc_trace+0xab/0xbb [<ffffffffa0248c29>] snd_usb_parse_audio_interface+0x4ac/0x567 [snd_usb_audio] [<ffffffffa023f0ff>] snd_usb_create_stream+0xe9/0x125 [snd_usb_audio] [<ffffffffa023f9b1>] usb_audio_probe+0x62a/0x72c [snd_usb_audio] ..... This patch fixes the regression by checking the fixed endpoint number for each substream instead of the endpoint pointer. Reported-and-tested-by: Jamie Heilman <jamie@audible.transient.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: fix rate_list memory leakClemens Ladisch2012-05-311-0/+3
| | | | | | | | | | The array of sample rates is reallocated every time when opening the PCM device, but was freed only once when unplugging the device. Reported-by: "Alexander E. Patrakov" <patrakov@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: fix stream info output in /procDaniel Mack2012-05-211-0/+12
| | | | | | | | | Set some substream struct members to make the proc interface code work again. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Felix Homann <linuxaudio@showlabor.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio - Call get_min_max_*() after determining the name stringTakashi Iwai2012-05-151-13/+13
| | | | | | | | | get_min_max_with_quirks() must be called after the control id name string is determined, but the current code changes the id name string after calling the function. Reported-by: Christian Melki <christian.melki@ericsson.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Fix commentMark Hills2012-05-111-10/+2
| | | | | | | | | | | Explained by Takashi in <s5hfwbtmz0q.wl%tiwai@suse.de> > The reason is because get_min_max*() isn't called in the place you > created these controls, and get_min_max() would be called only for > integer volumes later even if uninitialized. A short cut for booleans. Signed-off-by: Mark Hills <mark@pogo.org.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: fix some typos in endpoint.c documentationDaniel Mack2012-04-241-25/+33
| | | | | | | Also be more specific about some details while at it. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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