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* Merge tag 'sound-4.5-rc1' of ↵Linus Torvalds2016-01-177-21/+23
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound updates from Takashi Iwai: "We've had quite busy weeks in this cycle. Looking at ALSA core, the significant changes are a few fixes wrt timer and sequencer ioctls that have been revealed by fuzzer recently. Other than that, ASoC core got a few updates about DAI link handling, but these are rather straightforward refactoring. In drivers scene, ASoC received quite lots of new drivers in addition to bunch of updates for still ongoing Intel Skylake support and topology API. HD-audio gained a new HDMI/DP hotplug notification via component. FireWire got a pile of code refactoring/updates with SCS.1x driver integration. More highlights are shown below. [ NOTE: this contains also many commits for DRM. This is due to the pull of drm stable branch into sound tree, as the base of i915 audio component work for HD-audio. The highlights below don't contain these DRM changes, as these are supposed to be pulled via drm tree in anyway sooner or later. ] Core: - Handful fixes to harden ALSA timer and sequencer ioctls against races reported by syzkaller fuzzer - Irq description string can be unique to each card; only for HD-audio for now ASoC: - Conversion of the array of DAI links to a list for supporting dynamically adding and removing DAI links - Topology API enhancements to make everything more component based and being able to specify PCM links via topology - Some more fixes for the topology code, though it is still not final and ready for enabling in production; we really need to get to the point where that can be done - A pile of changes for Intel SkyLake drivers which hopefully deliver some useful initial functionality for systems with this chipset, though there is more work still to come - Lots of new features and cleanups for the Renesas drivers - ANC support for WM5110 - New drivers: Imagination Technologies IPs, Atmel class D speaker, Cirrus CS47L24 and WM1831, Dialog DA7128, Realtek RT5659 and RT56156, Rockchip RK3036, TI PC3168A, and AMD ACP - Rename PCM1792a driver to be generic pcm179x HD-Audio: - Use audio component for i915 HDMI/DP hotplug handling - On-demand binding with i915 driver - bdl_pos_adj parameter adjustment for Baytrail controllers - Enable power_save_node for CX20722; this shouldn't lead to regression, hopefully - Kabylake HDMI/DP codec support - Quirks for Lenovo E50-80, Dell Latitude E-series, and other Dell machines - A few code refactoring FireWire: - Lots of code cleanup and refactoring - Integrate the support of SCS.1x devices into snd-oxfw driver; snd-scs1x driver is obsoleted USB-audio: - Fix possible NULL dereference at disconnection - A regression fix for Native Instruments devices Misc: - A few code cleanups of fm801 driver" * tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (722 commits) ALSA: timer: Code cleanup ALSA: timer: Harden slave timer list handling ALSA: hda - Add fixup for Dell Latitidue E6540 ALSA: timer: Fix race among timer ioctls ALSA: hda - add codec support for Kabylake display audio codec ALSA: timer: Fix double unlink of active_list ALSA: usb-audio: Fix mixer ctl regression of Native Instrument devices ALSA: hda - fix the headset mic detection problem for a Dell laptop ALSA: hda - Fix white noise on Dell Latitude E5550 ALSA: hda_intel: add card number to irq description ALSA: seq: Fix race at timer setup and close ALSA: seq: Fix missing NULL check at remove_events ioctl ALSA: usb-audio: Avoid calling usb_autopm_put_interface() at disconnect ASoC: hdac_hdmi: remove unused hdac_hdmi_query_pin_connlist ASoC: AMD: Add missing include file ALSA: hda - Fixup inverted internal mic for Lenovo E50-80 ALSA: usb: Add native DSD support for Oppo HA-1 ASoC: Make aux_dev more like a generic component ASoC: bcm2835: cleanup includes by ordering them alphabetically ASoC: AMD: Manage ACP 2.x SRAM banks power ...
| * ALSA: usb-audio: Fix mixer ctl regression of Native Instrument devicesTakashi Iwai2016-01-131-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | The commit [da6d276957ea: ALSA: usb-audio: Add resume support for Native Instruments controls] brought a regression where the Native Instrument audio devices don't get the correct value at update due to the missing shift at writing. This patch addresses it. Fixes: da6d276957ea ('ALSA: usb-audio: Add resume support for Native Instruments controls') Reported-and-tested-by: Owen Williams <owilliams@mixxx.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb-audio: Avoid calling usb_autopm_put_interface() at disconnectTakashi Iwai2016-01-121-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ALSA PCM may still have a leftover instance after disconnection and it delays its release. The problem is that the PCM close code path of USB-audio driver has a call of snd_usb_autosuspend(). This involves with the call of usb_autopm_put_interface() and it may lead to a kernel Oops due to the NULL object like: BUG: unable to handle kernel NULL pointer dereference at 0000000000000190 IP: [<ffffffff815ae7ef>] usb_autopm_put_interface+0xf/0x30 PGD 0 Call Trace: [<ffffffff8173bd94>] snd_usb_autosuspend+0x14/0x20 [<ffffffff817461bc>] snd_usb_pcm_close.isra.14+0x5c/0x90 [<ffffffff8174621f>] snd_usb_playback_close+0xf/0x20 [<ffffffff816ef58a>] snd_pcm_release_substream.part.36+0x3a/0x90 [<ffffffff816ef6b3>] snd_pcm_release+0xa3/0xb0 [<ffffffff816debb0>] snd_disconnect_release+0xd0/0xe0 [<ffffffff8114d417>] __fput+0x97/0x1d0 [<ffffffff8114d589>] ____fput+0x9/0x10 [<ffffffff8109e452>] task_work_run+0x72/0x90 [<ffffffff81088510>] do_exit+0x280/0xa80 [<ffffffff8108996a>] do_group_exit+0x3a/0xa0 [<ffffffff8109261f>] get_signal+0x1df/0x540 [<ffffffff81040903>] do_signal+0x23/0x620 [<ffffffff8114c128>] ? do_readv_writev+0x128/0x200 [<ffffffff810012e1>] prepare_exit_to_usermode+0x91/0xd0 [<ffffffff810013ba>] syscall_return_slowpath+0x9a/0x120 [<ffffffff817587cd>] ? __sys_recvmsg+0x5d/0x70 [<ffffffff810d2765>] ? ktime_get_ts64+0x45/0xe0 [<ffffffff8115dea0>] ? SyS_poll+0x60/0xf0 [<ffffffff818d2327>] int_ret_from_sys_call+0x25/0x8f We have already a check of disconnection in snd_usb_autoresume(), but the check is missing its counterpart. The fix is just to put the same check in snd_usb_autosuspend(), too. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=109431 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb: Add native DSD support for Oppo HA-1Jurgen Kramer2016-01-111-0/+1
| | | | | | | | | | | | | | | | | | This patch adds native DSD support for the Oppo HA-1. It uses a XMOS chipset but they use their own vendor ID. Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * Merge branch 'for-linus' into for-nextTakashi Iwai2015-12-235-12/+44
| |\ | | | | | | | | | | | | Conflicts: drivers/gpu/drm/i915/intel_pm.c
| * | ALSA: usb-audio: use list_for_each_entry_continue_reverseGeliang Tang2015-12-221-4/+2
| | | | | | | | | | | | | | | | | | | | | | | | For better readability, use list_for_each_entry_continue_reverse() in have_dup_chmap(). Signed-off-by: Geliang Tang <geliangtang@163.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usb-audio: constify usb_protocol_ops structuresJulia Lawall2015-12-111-12/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The usb_protocol_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usx2y: fix inconsistent indenting on if statementColin Ian King2015-12-021-1/+1
| | | | | | | | | | | | | | | | | | | | | minor change, indenting is one tab out. Signed-off-by: Colin Ian King <colin.king@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: midi: constify snd_rawmidi_global_ops structuresJulia Lawall2015-11-221-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The snd_rawmidi_global_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: ua101: replace le16_to_cpu() with usb_endpoint_maxp()Cheah Kok Cheong2015-11-161-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit 939f325f4a0f ("usb: add usb_endpoint_maxp() macro") and commit 29cc88979a88 ("USB: use usb_endpoint_maxp() instead of le16_to_cpu()") introduced a new helper macro. This trivial patch convert remaining users found in ua101 driver. Signed-off-by: Cheah Kok Cheong <thrust73@gmail.com> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | Merge branch 'for-linus' of ↵Linus Torvalds2016-01-151-1/+1
|\ \ \ | |_|/ |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial Pull trivial tree updates from Jiri Kosina. * 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: floppy: make local variable non-static exynos: fixes an incorrect header guard dt-bindings: fixes some incorrect header guards cpufreq-dt: correct dead link in documentation cpufreq: ARM big LITTLE: correct dead link in documentation treewide: Fix typos in printk Documentation: filesystem: Fix typo in fs/eventfd.c fs/super.c: use && instead of & for warn_on condition Documentation: fix sysfs-ptp lib: scatterlist: fix Kconfig description
| * | treewide: Fix typos in printkMasanari Iida2015-12-081-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | This patch fix multiple spelling typos found in various part of kernel. Signed-off-by: Masanari Iida <standby24x7@gmail.com> Acked-by: Randy Dunlap <rdunlap@infradead.org> Signed-off-by: Jiri Kosina <jkosina@suse.cz>
* | | ALSA: usb-audio: Add sample rate inquiry quirk for AudioQuest DragonFlyAnssi Hannula2015-12-141-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Avoid getting sample rate on AudioQuest DragonFly as it is unsupported and causes noisy "cannot get freq at ep 0x1" messages when playback starts. Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: usb-audio: Add a more accurate volume quirk for AudioQuest DragonFlyAnssi Hannula2015-12-144-12/+43
| |/ |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | AudioQuest DragonFly DAC reports a volume control range of 0..50 (0x0000..0x0032) which in USB Audio means a range of 0 .. 0.2dB, which is obviously incorrect and would cause software using the dB information in e.g. volume sliders to have a massive volume difference in 100..102% range. Commit 2d1cb7f658fb ("ALSA: usb-audio: add dB range mapping for some devices") added a dB range mapping for it with range 0..50 dB. However, the actual volume mapping seems to be neither linear volume nor linear dB scale, but instead quite close to the cubic mapping e.g. alsamixer uses, with a range of approx. -53...0 dB. Replace the previous quirk with a custom dB mapping based on some basic output measurements, using a 10-item range TLV (which will still fit in alsa-lib MAX_TLV_RANGE_SIZE). Tested on AudioQuest DragonFly HW v1.2. The quirk is only applied if the range is 0..50, so if this gets fixed/changed in later HW revisions it will no longer be applied. v2: incorporated Takashi Iwai's suggestion for the quirk application method Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: work around CH345 input SysEx corruptionClemens Ladisch2015-11-161-0/+42
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | One of the many faults of the QinHeng CH345 USB MIDI interface chip is that it does not handle received SysEx messages correctly -- every second event packet has a wrong code index number, which is the one from the last seen message, instead of 4. For example, the two messages "FE F0 01 02 03 04 05 06 07 08 09 0A 0B 0C 0D 0E F7" result in the following event packets: correct: CH345: 0F FE 00 00 0F FE 00 00 04 F0 01 02 04 F0 01 02 04 03 04 05 0F 03 04 05 04 06 07 08 04 06 07 08 04 09 0A 0B 0F 09 0A 0B 04 0C 0D 0E 04 0C 0D 0E 05 F7 00 00 05 F7 00 00 A class-compliant driver must interpret an event packet with CIN 15 as having a single data byte, so the other two bytes would be ignored. The message received by the host would then be missing two bytes out of six; in this example, "F0 01 02 03 06 07 08 09 0C 0D 0E F7". These corrupted SysEx event packages contain only data bytes, while the CH345 uses event packets with a correct CIN value only for messages with a status byte, so it is possible to distinguish between these two cases by checking for the presence of this status byte. (Other bugs in the CH345's input handling, such as the corruption resulting from running status, cannot be worked around.) Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: stable@vger.kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: prevent CH345 multiport output SysEx corruptionClemens Ladisch2015-11-164-0/+16
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The CH345 USB MIDI chip has two output ports. However, they are multiplexed through one pin, and the number of ports cannot be reduced even for hardware that implements only one connector, so for those devices, data sent to either port ends up on the same hardware output. This becomes a problem when both ports are used at the same time, as longer MIDI commands (such as SysEx messages) are likely to be interrupted by messages from the other port, and thus to get lost. It would not be possible for the driver to detect how many ports the device actually has, except that in practice, _all_ devices built with the CH345 have only one port. So we can just ignore the device's descriptors, and hardcode one output port. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: stable@vger.kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: add packet size quirk for the Medeli DD305Clemens Ladisch2015-11-161-0/+1
| | | | | | | | | | | | Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb: Add native DSD support for Aune X1SJurgen Kramer2015-11-091-0/+1
|/ | | | | | | | This patch adds native DSD support for the Aune X1S 32BIT/384 DSD DAC Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: USB-audio: Remove mixer entry from Zoom R16/24 quirkRicard Wanderlof2015-10-191-7/+0
| | | | | | | | The device has no mixer (and identifies itself as such), so just skip the mixer definition. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: USB-audio: Adjust max packet size calculation for tx_length_quirkRicard Wanderlof2015-10-191-2/+10
| | | | | | | | | | | | | | | | For the Zoom R16/24 (tx_length_quirk set), when calculating the maximum sample frequency, consideration must be made for the fact that four bytes of the packet contain a length descriptor and consequently must not be counted as part of the audio data. This is corroborated by the wMaxPacketSize for this device, which is 108 bytes according for the USB playback endpoint descriptor. The frame size is 8 bytes (2 channels of 4 bytes each), and the 108 bytes thus work out as 13 * 8 + 4, i.e. corresponding to 13 frames plus the additional 4 byte length descriptor. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: USB-audio: Add quirk for Zoom R16/24 playbackRicard Wanderlof2015-10-197-9/+61
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The Zoom R16/24 have a nonstandard playback format where each isochronous packet contains a length descriptor in the first four bytes. (Curiously, capture data does not contain this and requires no quirk.) The quirk involves adding the extra length descriptor whenever outgoing isochronous packets are generated, both in pcm.c (outgoing audio) and endpoint.c (silent data). In order to make the quirk as unintrusive as possible, for pcm.c:prepare_playback_urb(), the isochronous packet descriptors are initially set up in the same way no matter if the quirk is enabled or not. Once it is time to actually copy the data into the outgoing packet buffer (together with the added length descriptors) the isochronous descriptors are adjusted in order take the increased payload length into account. For endpoint.c:prepare_silent_urb() it makes more sense to modify the actual function, partly because the function is less complex to start with and partly because it is not as time-critical as prepare_playback_urb() (whose bulk is run with interrupts disabled), so the (minute) additional time spent in the non-quirk case is motivated by the simplicity of having a single function for all cases. The quirk is controlled by the new tx_length_quirk member in struct snd_usb_substream and struct snd_usb_audio, which is conveyed to pcm.c and endpoint.c from quirks.c in a similar manner to the txfr_quirk member in the same structs. In contrast to txfr_quirk however, the quirk is enabled directly in quirks.c:create_standard_audio_quirk() by checking the USB ID in that function. Another option would be to introduce a new QUIRK_AUDIO_ZOOM_INTERFACE or somesuch, which would have made the quirk very plain to see in the quirk table, but it was felt that the additional code needed to implement it this way would just make the implementation more complex with no real gain. Tested with a Zoom R16, both by doing capture and playback separately using arecord and aplay (8 channel capture and 2 channel playback, respectively), as well as capture and playback together using Ardour, as well as Audacity and Qtractor together with jackd. The R24 is reportedly compatible with the R16 when used as an audio interface. Both devices share the same USB ID and have the same number of inputs (8) and outputs (2). Therefore "R16/24" is mentioned throughout the patch. Regression tested using an Edirol UA-5 in both class compliant (16-bit) and "advanced" (24 bit, forces the use of quirks) modes. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Tested-by: Panu Matilainen <pmatilai@laiskiainen.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: USB-audio: Add offset parameter to copy_to_urb()Ricard Wanderlof2015-10-191-6/+6
| | | | | | | | Preparation for adding Zoom R16/24 quirk. No functional change. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: USB-audio: Break out creation of silent urbs from prepare_outbound_urb()Ricard Wanderlof2015-10-191-19/+27
| | | | | | | | Refactoring in preparation for adding Zoom R16/24 quirk. No functional change. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: USB-audio: Also move out hwptr_done wrap from prepare_playback_urb()Ricard Wanderlof2015-10-191-3/+6
| | | | | | | | Refactoring in preparation for adding Zoom R16/24 quirk. No functional change. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: USB-audio: Break out copying to urb from prepare_playback_urb()Ricard Wanderlof2015-10-191-14/+21
| | | | | | | | Refactoring in preparation for adding Zoom R16/24 quirk. No functional change. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: USB-audio: Add support for Novation Nocturn MIDIcontrol surfaceRicard Wanderlof2015-10-161-0/+9
| | | | | | | | | | The Nocturn needs the MIDI_RAW_BYTES quirk, like other Novation devices. Tested that the Nocturn shows up in aconnect, and that it can be used as a control surface (using the xtor synthesizer patch editor). Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Fix max packet size calculation for USB audioRicard Wanderlof2015-10-131-2/+17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Rounding must take place before multiplication with the frame size, since each packet contains a whole number of frames. We must also properly consider the data interval, as a larger data interval will result in larger packets, which, depending on the sampling frequency, can result in packet sizes that are less than integral multiples of the packet size for a lower data interval. Detailed explanation and rationale: The code before this commit had the following expression on line 613 to calculate the maximum isochronous packet size: maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) >> (16 - ep->datainterval); Here, ep->freqmax is the maximum assumed sample frequency, calculated from the nominal sample frequency plus 25%. It is ultimately derived from ep->freqn, which is in the units of frames per packet, from get_usb_full_speed_rate() or usb_high_speed_rate(), as applicable, in Q16.16 format. The expression essentially adds the Q16.16 equivalent of 0.999... (i.e. the largest number less than one) to the sample rate, in order to get a rate whose integer part is rounded up from the fractional value. The multiplication with (frame_bits >> 3) yields the number of bytes in a packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back to an integer, taking into consideration the bDataInterval field of the endpoint descriptor (which describes how often isochronous packets are transmitted relative to the (micro)frame rate (125us or 1ms, for USB high speed and full speed, respectively)). For this discussion we will initially assume a bDataInterval of 0, so the second line of the expression just converts the Q16.16 value to an integer. In order to illustrate the problem, we will set frame_bits 64, which corresponds to a frame size of 8 bytes. The problem here is twofold. First, the rounding operation consists of the addition of 0x0.ffff and subsequent conversion to integer, but as the expression stands, the conversion to integer is done after multiplication with the frame size, rather than before. This results in the resulting maxsize becoming too large. Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is 0xc0000 (see usb_high_speed_rate()). Add 25% (line 612) and we get 0xf0000. The calculated maxsize is then ((0xf0000 + 0x0ffff) * 8) >> 16 = 127 . However, if we do the number of bytes calculation in a less obscure way it's more apparent what the true corresponding packet size is: we get ceil(96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612, and the 8000 is the number of isochronous packets per second on a high speed USB connection (125 us microframe interval). This is fixed by performing the complete rounding operation prior to multiplication with the frame rate. The second problem is that when considering the ep->datainterval, this must be done before rounding, in order to take the advantage of the fact that if the number of bytes per packet is not an integer, the resulting rounded-up integer is not necessarily a factor of two when the data interval is increased by the same factor. For instance, assuming a freqency of 41 kHz, the resulting bytes-per-packet value for USB high speed is 41 kHz / 8000 = 5.125, or 0x52000 in Q16.16 format. With a data interval of 1 (ep->datainterval = 0), this means that 6 frames per packet are needed, whereas with a data interval of 2 we need 10.25, i.e. 11 frames needed. Rephrasing the maxsize expression to: maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * (frame_bits >> 3); for the above 96 kHz example we instead get ((0xf0000 + 0xffff) >> 16) * 8 = 120 which is the correct value. We can also do the calculation with a non-integer sample rate which is when rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn = 0x58333, and resulting ep->freqmax 0x58333 * 1.25 = 0x6e3ff (rounded down)): Original maxsize = ((0x6e3ff + 0xffff) * 8) << 16 = 63 (63.124.. rounded down) True maxsize = ceil(44100 * 1.25 / 8000) * 8 = 7 * 8 = 56 New maxsize = ((0x6e3ff + 0xffff) >> 16) * 8 = 7 * 8 = 56 This is also corroborated by the wMaxPacketSize check on line 616. Assume that wMaxPacketSize = 104, with ep->maxpacksize then having the same value. As 104 < 127, we get maxsize = 104. ep->freqmax is then recalculated to (104 / 8) << 16 = 0xd0000 . Putting that rate into the original maxsize calculation yields a maxsize of ((0xd0000 + 0xffff) * 8) >> 16 = 111 (with decimals 111.99988). Clearly, we should get back the 104 here, which we would with the new expression: ((0xd0000 + 0xffff) >> 16) * 8 = 104 . (The error has not been a problem because it only results in maxsize being a bit too big which just wastes a couple of bytes, either as a result of the first maxsize calculation, or because the resulting calculation will hit the wMaxPacketSize value before the packet is too big, resulting in fixing the size to wMaxPacketSize even though the packet is actually not too long.) Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Allow any MIDI endpoint to drive use of interrupt transfer ↵Keith A. Milner2015-10-111-4/+7
| | | | | | | | | | | | | | | | | | | | | on newer Roland devices This patch enables interrupt transfer mode for MIDI ports on newer Boss/Roland devices such as the GT-100/001 which support interrupt transfer on both IN and OUT MIDI endpoints. Previously this wasn't being enabled for these devices as the code was specifically looking for the scenario where the IN endpoint supported interrupt transfer and the OUT endpoint was bulk transfer. Newer devices support interrupt transfer for both endpoints. This has been tested on Boss devices GT-001, BR-80 and JS-8 and Roland VS-20. It would benefit from some regresison testing with other devices if possible. Signed-off-by: Keith A. Milner <maillist@superlative.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: harmless underflow in snd_audigy2nx_led_put()Dan Carpenter2015-09-281-1/+1
| | | | | | | | | | We want to verify that "value" is either zero or one, so we test if it is greater than one. Unfortunately, this is a signed int so it could also be negative. I think this is harmless but it introduces a static checker warning. Let's make "value" unsigned. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Change internal PCM orderJohan Rastén2015-09-071-1/+9
| | | | | | | | | | | | | | | New PCMs will now be added to the end of the chip's PCM list instead of to the front. This changes the way streams are combined so that the first capture stream will now be merged with the first playback stream instead of the last. This fixes a problem with ASUS U7. Cards with one playback stream and cards without capture streams should be unaffected by this change. Exception added for M-Audio Audiophile USB (tm) since it seems to have a fix to swap capture stream numbering in alsa-lib conf/cards/USB-audio.conf Signed-off-by: Johan Rastén <johan@oljud.se> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: correct the value cache check.Yao-Wen Mao2015-08-281-1/+1
| | | | | | | | The check of cval->cached should be zero-based (including master channel). Signed-off-by: Yao-Wen Mao <yaowen@google.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Handle normal and auto-suspend equallyTakashi Iwai2015-08-261-20/+10
| | | | | | | | | | In theory, the device may get suspended even at runtime PM suspend. Currently we don't save the mixer state for autopm, and it may bring inconsistency. This patch removes the special handling for autosuspend. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Replace probing flag with active refcountTakashi Iwai2015-08-262-9/+4
| | | | | | We can use active refcount for preventing autopm during probe. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Avoid nested autoresume callsTakashi Iwai2015-08-267-143/+145
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | After the recent fix of runtime PM for USB-audio driver, we got a lockdep warning like: ============================================= [ INFO: possible recursive locking detected ] 4.2.0-rc8+ #61 Not tainted --------------------------------------------- pulseaudio/980 is trying to acquire lock: (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] but task is already holding lock: (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] This comes from snd_usb_autoresume() invoking down_read() and it's used in a nested way. Although it's basically safe, per se (as these are read locks), it's better to reduce such spurious warnings. The read lock is needed to guarantee the execution of "shutdown" (cleanup at disconnection) task after all concurrent tasks are finished. This can be implemented in another better way. Also, the current check of chip->in_pm isn't good enough for protecting the racy execution of multiple auto-resumes. This patch rewrites the logic of snd_usb_autoresume() & co; namely, - The recursive call of autopm is avoided by the new refcount, chip->active. The chip->in_pm flag is removed accordingly. - Instead of rwsem, another refcount, chip->usage_count, is introduced for tracking the period to delay the shutdown procedure. At the last clear of this refcount, wake_up() to the shutdown waiter is called. - The shutdown flag is replaced with shutdown atomic count; this is for reducing the lock. - Two new helpers are introduced to simplify the management of these refcounts; snd_usb_lock_shutdown() increases the usage_count, checks the shutdown state, and does autoresume. snd_usb_unlock_shutdown() does the opposite. Most of mixer and other codes just need this, and simply returns an error if it receives an error from lock. Fixes: 9003ebb13f61 ('ALSA: usb-audio: Fix runtime PM unbalance') Reported-and-tested-by: Alexnader Kuleshov <kuleshovmail@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'for-linus' into for-nextTakashi Iwai2015-08-212-1/+2
|\
| * ALSA: usb: Add native DSD support for Gustard DAC-X20UJurgen Kramer2015-08-211-0/+1
| | | | | | | | | | | | | | | | This patch adds native DSD support for the Gustard DAC-X20U. Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb-audio: Fix runtime PM unbalanceTakashi Iwai2015-08-191-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The fix for deadlock in PM in commit [1ee23fe07ee8: ALSA: usb-audio: Fix deadlocks at resuming] introduced a new check of in_pm flag. However, the brainless patch author evaluated it in a wrong way (logical AND instead of logical OR), thus usb_autopm_get_interface() is wrongly called at probing, leading to unbalance of runtime PM refcount. This patch fixes it by correcting the logic. Reported-by: Hans Yang <hansy@nvidia.com> Fixes: 1ee23fe07ee8 ('ALSA: usb-audio: Fix deadlocks at resuming') Cc: <stable@vger.kernel.org> [v3.15+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Recurse before saving terminal propertiesJulian Scheel2015-08-191-5/+11
| | | | | | | | | | | | | | | | | | | | | | | | The input terminal parser recurses into the referenced clock entity to verify it is existant and thus the terminal descriptor is valid. The actual property values of the term instance which is initially parsed must not be overriden by the recursion. For this to work the term properties have to be assigned after recursing into the referenced clock entity descriptors. Signed-off-by: Julian Scheel <julian@jusst.de> Acked-by: Daniel Mack <daniel@zonque.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb: handle descriptor with SYNC_NONE illegal valuePierre-Louis Bossart2015-08-161-2/+16
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The M-Audio Transit exposes an interface with a SYNC_NONE attribute. This is not a valid value according to the USB audio classspec. However there is a sync endpoint associated to this record. Changing the logic to try to use this sync endpoint allows for seamless transitions between altset 2 and altset 3. If any errors happen, the behavior remains the same. $ more /proc/asound/card1/stream0 M-Audio Transit USB at usb-0000:00:14.0-2, full speed : USB Audio Playback: Status: Stop Interface 1 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (ADAPTIVE) Rates: 48001 - 96000 (continuous) Interface 1 Altset 2 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (NONE) Rates: 8000 - 48000 (continuous) Interface 1 Altset 3 Format: S16_LE Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb: fix corrupted pointers due to interface setting changePierre-Louis Bossart2015-08-161-0/+14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When a transition occurs between alternate settings that do not use the same synchronization method, the substream pointers were not reset. This prevents audio from being played during the second transition. Identified and tested with M-Audio Transit device (0763:2006 Midiman M-Audio Transit) Details of the issue: First playback to adaptive endpoint: $ aplay -Dhw:1,0 ~/24_96.wav Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes, Rate 96000 Hz, Stereo [ 3169.297556] usb 1-2: setting usb interface 1:1 [ 3169.297568] usb 1-2: Creating new playback data endpoint #3 [ 3169.298563] usb 1-2: Setting params for ep #3 (type 0, 3 urbs), ret=0 [ 3169.298574] usb 1-2: Starting data EP @ffff880035fc8000 first playback to asynchronous endpoint: $ aplay -Dhw:1,0 ~/16_48.wav Playing WAVE '/home/plb/16_48.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo [ 3204.520251] usb 1-2: setting usb interface 1:3 [ 3204.520264] usb 1-2: Creating new playback data endpoint #3 [ 3204.520272] usb 1-2: Creating new capture sync endpoint #83 [ 3204.521162] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0 [ 3204.521177] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0 [ 3204.521182] usb 1-2: Starting data EP @ffff880035fce000 [ 3204.521204] usb 1-2: Starting sync EP @ffff8800bd616000 second playback to adaptive endpoint: no audio and error on terminal: $ aplay -Dhw:1,0 ~/24_96.wav Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes, Rate 96000 Hz, Stereo aplay: pcm_write:1939: write error: Input/output error [ 3239.483589] usb 1-2: setting usb interface 1:1 [ 3239.483601] usb 1-2: Re-using EP 3 in iface 1,1 @ffff880035fc8000 [ 3239.484590] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0 [ 3239.484606] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0 This last line shows that a sync endpoint is used when it shouldn't. The sync endpoint is no longer valid and the pointers are corrupted Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Fix parameter block size for UAC2 control requestsJulian Scheel2015-08-142-20/+46
|/ | | | | | | | | | | | USB Audio Class version 2.0 supports three different parameter block sizes for CUR requests, which are 1 byte (5.2.3.1 Layout 1 Parameter Block), 2 bytes (5.2.3.2 Layout 2 Parameter Block) and 4 bytes (5.2.3.3 Layout 3 Parameter Block). Use the correct size according to the specific control as it was already done for UACv1. The allocated block size for control requests is increased to support the 4 byte worst case. Signed-off-by: Julian Scheel <julian@jusst.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: add dB range mapping for some devicesYao-Wen Mao2015-07-291-0/+24
| | | | | | | | Add the correct dB ranges of Bose Companion 5 and Drangonfly DAC 1.2. Signed-off-by: Yao-Wen Mao <yaowen@google.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: line6: Fix -EBUSY error during active monitoringTakashi Iwai2015-07-141-7/+2
| | | | | | | | | | | When a monitor stream is active, the next PCM stream access results in EBUSY error because of the check in line6_stream_start(). Fix this by just skipping the submission of pending URBs when the stream is already running instead. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=101431 Cc: <stable@vger.kernel.org> # v4.0+ Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Add MIDI support for Steinberg MI2/MI4Dominic Sacré2015-07-011-0/+68
| | | | | | | | | | | | | | | | The Steinberg MI2 and MI4 interfaces are compatible with the USB class audio spec, but the MIDI part of the devices is reported as a vendor specific interface. This patch adds entries to quirks-table.h to recognize the MIDI endpoints. Audio functionality was already working and is unaffected by this change. Signed-off-by: Dominic Sacré <dominic.sacre@gmx.de> Signed-off-by: Albert Huitsing <albert@huitsing.nl> Acked-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Set correct type for some UAC2 mixer controls.Johan Rastén2015-06-111-3/+3
| | | | | | | | | Changed ctl type for Input Gain Control and Input Gain Pad Control to USB_MIXER_S16 as per section 5.2.5.7.11-12 in the USB Audio Class 2.0 definition. Signed-off-by: Johan Rastén <johan@oljud.se> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'for-linus' into for-nextTakashi Iwai2015-06-093-8/+13
|\ | | | | | | | | | | Resolve the non-trivial conflict due to the hdac regmap API changes. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb-audio: add native DSD support for JLsounds I2SoverUSBJurgen Kramer2015-06-081-2/+3
| | | | | | | | | | | | | | | | This patch adds native DSD support for the XMOS based JLsounds I2SoverUSB board Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb-audio: fix missing input volume controls in MAYA44 USB(+)Clemens Ladisch2015-06-031-6/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The driver worked around an error in the MAYA44 USB(+)'s mixer unit descriptor by aborting before parsing the missing field. However, aborting parsing too early prevented parsing of the other units connected to this unit, so the capture mixer controls would be missing. Fix this by moving the check for this descriptor error after the parsing of the unit's input pins. Reported-by: nightmixes <nightmixes@gmail.com> Tested-by: nightmixes <nightmixes@gmail.com> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb-audio: add MAYA44 USB+ mixer control namesClemens Ladisch2015-06-031-0/+5
| | | | | | | | | | | | | | | | | | | | Add mixer control names for the ESI Maya44 USB+ (which appears to be identical width the AudioTrak Maya44 USB). Reported-by: nightmixes <nightmixes@gmail.com> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb-audio: don't try to get Outlaw RR2150 sample rateEric Wong2015-05-301-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This quirk allows us to avoid the noisy: current rate 0 is different from the runtime rate message every time playback starts. While USB DAC in the RR2150 supports reading the sample rate, it never returns a sample rate other than zero in my observation with common sample rates. Signed-off-by: Eric Wong <normalperson@yhbt.net> Cc: Joe Turner <joe@oampo.co.uk> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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