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* ALSA: usb-audio: selector map for M-Audio FT C400Eldad Zack2013-01-141-0/+13
| | | | | | | | Add names of the clock sources for the M-Audio Fast Track C400. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: M-Audio FT C400 skip packet quirkEldad Zack2013-01-141-0/+11
| | | | | | | | Attain constant real-world latency by skipping 16 data packets. The number of packets to be skipped was found by trial and error. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: correct M-Audio C400 clock source quirkEldad Zack2013-01-141-2/+2
| | | | | | | | | | | | | | | Taking another look at the C400 descriptors, I see now that there is a clock selector (0x80) for this device. Right now, the clock source points to the internal clock (0x81), which is also valid. When the external clock source (0x82) is selected in the mixer, and the rates mismatch (if it's free-running it is fixed to 48KHz), xruns will occur. Set the clock ID to the clock selector unit (0x81), which then allows the validation code to function correctly. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb - fix race in creation of M-Audio Fast track pro driverDavid Henningsson2013-01-141-3/+5
| | | | | | | | | | | | | | | | | | | A patch in the 3.2 kernel caused regression with hotplugging the M-Audio Fast track pro, or sound after suspend. I don't have the device so I haven't done a full analysis, but it seems userspace (both udev and pulseaudio) got confused when a card was created, immediately destroyed, and then created again. However, at least one person in the bug report (martin djfun) reports that this patch resolves the issue for him. It also leaves a message in the log: "snd-usb-audio: probe of 1-1.1:1.1 failed with error -5" which is a bit misleading. It is better than non-working audio, but maybe there's a more elegant solution? BugLink: https://bugs.launchpad.net/bugs/1095315 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Fix NULL dereference by access to non-existing substreamTakashi Iwai2013-01-111-0/+10
| | | | | | | | | | | | | | The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for audioformat mismatch] introduced the correction of parameters to be set for sync EP. But since the new code assumes that the sync EP is always paired with the data EP of another direction, it triggers Oops when a device only with a single direction is used. This patch adds a proper check of sync EP type and the presence of the paired substream for avoiding the crash. Reported-and-tested-by: Jens Axboe <axboe@kernel.dk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Make ebox44_table staticSachin Kamat2013-01-101-1/+1
| | | | | | | | | Fixes the following sparse warning: sound/usb/mixer_quirks.c:1209:23: warning: symbol 'ebox44_table' was not declared. Should it be static? Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Fix kernel panic of Digidesign Mbox2 quirkDamien Zammit2013-01-044-15/+8
| | | | | | | | | | | | | | | This patch is based on 3.8-rc1. It fixes two things: 1) A kernel panic caused by incorrect allocation of a u8 variable "bootresponse". 2) A noisy dmesg (urb status -32) caused by broken pipe to an invalid midi endpoint. It is also a little cleaner because there is no need for a new QUIRK_MIDI type as suggested by kernel developers, since the device follows exactly the MIDIMAN protocol. Signed-off-by: Damien Zammit <damien@zamaudio.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Add support for Creative BT-D1 via usb sound quirksAlexander Schremmer2013-01-031-0/+22
| | | | | | | | | | | | | | | | | | | | | | Support the Creative BT-D1 Bluetooth USB audio device. Before this patch, Linux had trouble finding the correct USB descriptors and bailed out with these messages: no or invalid class specific endpoint descriptor Now it still prints these messages on hotplug: snd-usb-audio: probe of ...:1.0 failed with error -5 snd-usb-audio: probe of ...:1.2 failed with error -5 snd-usb-audio: probe of ...:1.3 failed with error -5 But the device works correctly, including the HID support. The patch is diff'ed against 3.8-rc1 but should apply to older kernels as well. Signed-off-by: Alexander Schremmer <alex@alexanderweb.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge tag 'sound-3.8' of ↵Linus Torvalds2012-12-204-0/+219
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "This update contains overall only driver-specific fixes. Slightly large LOC are seen in usb-audio driver for a couple of new device quirks and cs42l71 ASoC driver for enhanced features. The others are a few small (regression) fixes HD-audio, and yet other small / trival ASoC fixes." * tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card: ALSA: HDA: Fix sound resume hang ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pins ALSA: hda - Fix the wrong pincaps set in ALC861VD dallas/hp fixup ALSA: hda - Set codec->single_adc_amp flag for Realtek codecs ASoC: atmel-ssc: change disable to disable in dts node ASoC: Prevent pop_wait overwrite ALSA: usb-audio: ignore-quirk for HP Wireless Audio ALSA: hda - Always turn on pins for HDMI/DP ALSA: hda - Fix pin configuration of HP Pavilion dv7 ASoC: core: Fix splitting of log messages ASoC: cs42l73: Change VSPIN/VSPOUT to VSPINOUT ASoC: cs42l73: Add DAPM events for power down. ASoC: cs42l73: Add DMIC's as DAPM inputs. ASoC: sigmadsp: Fix endianness conversion issue ASoC: tpa6130a2: Use devm_* APIs
| * ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card:Damien Zammit2012-12-194-0/+183
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch is the result of a lot of trial and error, since there are no specs available for the device. Full duplex support is provided, i.e. playback and recording in stereo. The format is hardcoded at 48000Hz @ 24 bit, which is the maximum that the device supports. Also, MIDI in and MIDI out both work. Users will notice that the S/PDIF light also flashes when playback or recording is active. I believe this means that S/PDIF input/output is simultaneously activated with the analogue i/o during use. But this particular functionality remains untested. Note that this particular version of the patch is so far untested on the physical hardware because I have not compiled a full kernel with the changes. However, extensive testing has been done by many users of the hardware who believe other versions of my patch have worked since circa 2009. [Modified to make a function static by tiwai] Signed-off-by: Damien Zammit <damien@zamaudio.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb-audio: ignore-quirk for HP Wireless AudioEldad Zack2012-12-151-0/+36
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As Joe Cooper <swelljoe@gmail.com> reported, "On most HP Envy laptops the snd-usb-audio module causes the system to become unresponsive and Gnome Shell 3 to crash.". See also: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-December/057729.html Add a quirk to ignore this device (for now) to solve the instability issue and allow other USB audio devices to be used. Reported-by: Joe Cooper <swelljoe@gmail.com> Tested-by: Isaac Smith <hunternet93@gmail.com> Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge branch 'for-linus' of ↵Linus Torvalds2012-12-131-1/+1
|\ \ | |/ |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial Pull trivial branch from Jiri Kosina: "Usual stuff -- comment/printk typo fixes, documentation updates, dead code elimination." * 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (39 commits) HOWTO: fix double words typo x86 mtrr: fix comment typo in mtrr_bp_init propagate name change to comments in kernel source doc: Update the name of profiling based on sysfs treewide: Fix typos in various drivers treewide: Fix typos in various Kconfig wireless: mwifiex: Fix typo in wireless/mwifiex driver messages: i2o: Fix typo in messages/i2o scripts/kernel-doc: check that non-void fcts describe their return value Kernel-doc: Convention: Use a "Return" section to describe return values radeon: Fix typo and copy/paste error in comments doc: Remove unnecessary declarations from Documentation/accounting/getdelays.c various: Fix spelling of "asynchronous" in comments. Fix misspellings of "whether" in comments. eisa: Fix spelling of "asynchronous". various: Fix spelling of "registered" in comments. doc: fix quite a few typos within Documentation target: iscsi: fix comment typos in target/iscsi drivers treewide: fix typo of "suport" in various comments and Kconfig treewide: fix typo of "suppport" in various comments ...
| * Fix misspellings of "whether" in comments.Adam Buchbinder2012-11-191-1/+1
| | | | | | | | | | | | | | | | "Whether" is misspelled in various comments across the tree; this fixes them. No code changes. Signed-off-by: Adam Buchbinder <adam.buchbinder@gmail.com> Signed-off-by: Jiri Kosina <jkosina@suse.cz>
* | ALSA: usb-audio: Enable S/PDIF on the ASUS Xonar U3Denis Washington2012-12-121-2/+5
| | | | | | | | | | | | | | | | | | The only required change is to extend the existing Xonar U1 mixer quirks to the U3, which seems to be controlled the same way. Signed-off-by: Denis Washington <denisw@online.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb6fire: prevent driver panic state when stoppingJurgen Kramer2012-12-071-0/+3
| | | | | | | | | | | | | | | | | | | | | | The patch below prevents the 6fire usb driver going into panic state when stopping playing. On some systems the urb in handler (usb6fire_pcm_in_urb_handler) is being called while urbs are being killed off, this causes the driver to set panic state and can result in the kernel warning 'URB %p submitted while active'. Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: snd-usb-caiaq: remove __dev* attributesBill Pemberton2012-12-072-7/+7
| | | | | | | | | | | | | | | | | | | | | | | | CONFIG_HOTPLUG is going away as an option. As result the __dev* markings will be going away. Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, and __devexit. Signed-off-by: Bill Pemberton <wfp5p@virginia.edu> Acked-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: snd-usb-6fire: remove __dev* attributesBill Pemberton2012-12-0710-17/+17
| | | | | | | | | | | | | | | | | | | | | | CONFIG_HOTPLUG is going away as an option. As result the __dev* markings will be going away. Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, and __devexit. Signed-off-by: Bill Pemberton <wfp5p@virginia.edu> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: sync ep init fix for audioformat mismatchEldad Zack2012-12-041-7/+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>
* | ALSA: usb-audio: Fix missing autopm for MIDI inputTakashi Iwai2012-12-041-42/+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>
* | ALSA: usb-audio: Avoid autopm calls after disconnectionTakashi Iwai2012-12-041-1/+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>
* | ALSA: usb - Don't create "Speaker" mixer controls on headphones and headsetsDavid Henningsson2012-11-291-0/+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>
* | ALSA: usb-audio: FT C400 sync playback EP to capture EPEldad Zack2012-11-291-0/+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>
* | ALSA: usb-audio: Fast Track C400 mixer controlsEldad Zack2012-11-291-0/+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>
* | ALSA: usb-audio: Fast Track C400 mixer rangesEldad Zack2012-11-291-0/+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>
* | ALSA: usb-audio: M-Audio Fast Track C400 quirks tableEldad Zack2012-11-291-0/+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>
* | ALSA: usb-audio: parameterize FTU effect unit controlEldad Zack2012-11-291-8/+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>
* | ALSA: usb-audio: skip UAC2 EFFECT_UNITEldad Zack2012-11-291-1/+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>
* | ALSA: usb-audio: add control index offsetEldad Zack2012-11-293-1/+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>
* | ALSA: usb-audio: use sender stride for implicit feedbackEldad Zack2012-11-291-3/+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>
* | ALSA: usb-audio: replace hardcoded value with constEldad Zack2012-11-291-1/+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>
* | ALSA: usb-audio: add channel map supportTakashi Iwai2012-11-262-5/+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>
* | ALSA: usb-audio: fix delay account during pauseTakashi Iwai2012-11-231-1/+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>
* | ALSA: usb-audio: ignore delay calculation for capture streamTakashi Iwai2012-11-231-1/+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>
* | Merge branch 'for-linus' into for-nextTakashi Iwai2012-11-221-1/+1
|\ \
| * | 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>
* | | ALSA: usb-audio: process pending stop at PCM hw_free and closeTakashi Iwai2012-11-211-2/+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>
* | | ALSA: usb-audio: stop both data and sync endpoints asynchronouslyTakashi Iwai2012-11-213-9/+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>
* | | ALSA: usb-audio: simplify endpoint deactivation codeTakashi Iwai2012-11-211-16/+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>
* | | ALSA: usb-audio: simplify snd_usb_endpoint_start/stop argumentsTakashi Iwai2012-11-213-26/+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>
* | | ALSA: usb-audio: Deprecate async_unlink optionTakashi Iwai2012-11-213-9/+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>
* | | ALSA: usb-audio: Return meaningful error codes instead of -1 in format.cSachin Kamat2012-11-211-5/+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>
* | | ALSA: usb/6fire: Fix potential NULL pointer dereference in comm.cSachin Kamat2012-11-211-1/+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>
* | | Merge branch 'for-linus' into for-nextTakashi Iwai2012-11-192-5/+9
|\| | | | | | | | | | | | | | | | | | | | | | | Merge the recent HD-audio codec change for fixing recursive suspend calls. Conflicts: sound/pci/hda/hda_codec.c
| * | 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: use bitmap_weightJoe Perches2012-11-171-6/+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>
* | | | ALSA: usb-audio: Add quirk for Focusrite Scarlett 18i6Martin Schwenke2012-11-131-0/+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>
* | | | Merge branch 'for-linus' into for-nextTakashi Iwai2012-11-083-0/+17
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| * | | 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>
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