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* 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
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| * 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>
| * ALSA: usb-audio: Add mic volume fix quirk for Logitech Quickcam FusionWolfram Sang2015-05-291-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | Fix this from the logs: usb 7-1: New USB device found, idVendor=046d, idProduct=08ca ... usb 7-1: Warning! Unlikely big volume range (=3072), cval->res is probably wrong. usb 7-1: [5] FU [Mic Capture Volume] ch = 1, val = 4608/7680/1 Signed-off-by: Wolfram Sang <wsa@the-dreams.de> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge branch 'for-linus' into for-nextTakashi Iwai2015-05-291-0/+2
|\| | | | | | | Merge back the latest HD-audio stuff for further development.
| * ALSA: usb-audio: Add quirk for MS LifeCam HD-3000Vittorio G (VittGam)2015-05-241-0/+1
| | | | | | | | | | | | | | | | | | | | Microsoft LifeCam HD-3000 (045e:0779) needs a similar quirk for suppressing the unsupported sample rate inquiry. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481 Cc: <stable@vger.kernel.org> Signed-off-by: Vittorio Gambaletta <linuxbugs@vittgam.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb-audio: Add quirk for MS LifeCam StudioTakashi Iwai2015-05-191-0/+1
| | | | | | | | | | | | | | | | | | Microsoft LifeCam Studio (045e:0772) needs a similar quirk for suppressing the wrong sample rate inquiry. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: bcd2000: Make local data staticTakashi Iwai2015-05-261-1/+1
|/ | | | | | | Spotted by sparse: sound/usb/bcd2000/bcd2000.c:73:1: warning: symbol 'devices_used' was not declared. Should it be static? Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Fix audio output on Roland SC-D70 sound moduleTakamichi Horikawa2015-04-212-29/+6
| | | | | | | | | | | | | | | | | | | | Roland SC-D70 reports its device class as vendor specific class and the quirk QUIRK_AUDIO_FIXED_ENDPOINT was used for audio output. In the quirks table the sampling rate was hard-coded to 44100 Hz and therefore not worked when the sound module was in 48000 Hz mode. In this change the quirk is changed to QUIRK_AUDIO_STANDARD_INTERFACE but as the sound module reports incorrect bSubframeSize in its descriptors, additional change is made in format.c to detect it and to override it (which uses the existing code for Edirol SD-90). Tested both when the sound module was in 44100 Hz mode and 48000 Hz mode and both audio input and output. MIDI related part of the driver is not touched. Signed-off-by: Takamichi Horikawa <takamichiho@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'for-next' into for-linusTakashi Iwai2015-04-131-23/+17
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| * Merge branch 'for-linus' into for-nextTakashi Iwai2015-04-081-2/+7
| |\ | | | | | | | | | | | | | | | | | | Back merge HD-audio quirks to for-next branch, so that we can apply a couple of more quirks. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * \ Merge branch 'for-linus' into for-nextTakashi Iwai2015-03-161-0/+30
| |\ \
| * \ \ Merge branch 'for-linus' into for-nextTakashi Iwai2015-03-091-3/+3
| |\ \ \ | | | | | | | | | | | | | | | | | | | | Merging the HD-audio fixes back to base devel branch for further working on it.
| * | | | ALSA: usb-audio: Check Marantz/Denon USB DACs in a single placeTakashi Iwai2015-03-041-23/+17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There are three places doing the same check. Let's make them together. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | ALSA: usb-audio: Don't attempt to get Microsoft Lifecam Cinema sample rateAdam Honse2015-04-121-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Adds Microsoft LifeCam Cinema USB ID to the snd_usb_get_sample_rate_quirk list as the Lifecam Cinema does not appear to support getting the sample rate. Fixes the issue where the LifeCam Cinema would wait for USB timeout and log the message "cannot get freq at ep 0x82" when accessed. Addresses bug report https://bugzilla.kernel.org/show_bug.cgi?id=95961. Signed-off-by: Adam Honse <calcprogrammer1@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | ALSA: usb - Creative USB X-Fi Pro SB1095 volume knob supportDmitry M. Fedin2015-04-091-0/+1
| |_|_|/ |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Adds an entry for Creative USB X-Fi to the rc_config array in mixer_quirks.c to allow use of volume knob on the device. Adds support for newer X-Fi Pro card, known as "Model No. SB1095" with USB ID "041e:3237" Signed-off-by: Dmitry M. Fedin <dmitry.fedin@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: usb-audio: don't try to get Benchmark DAC1 sample rateEric Wong2015-04-041-2/+7
| |_|/ |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Adding this quirk allows us to avoid the noisy "cannot get freq at ep 0x1" message in dmesg output every time playback starts. This ought to affect other Benchmark DAC1 variations using the same "Microchip Technology, Inc." chip as well, but I have only tested with the "Pre" variant. Signed-off-by: Eric Wong <normalperson@yhbt.net> Cc: Joe Turner <joe@oampo.co.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: snd-usb: add quirks for Roland UA-22Daniel Mack2015-03-121-0/+30
| |/ |/| | | | | | | | | | | | | | | | | | | | | | | The device complies to the UAC1 standard but hides that fact with proprietary descriptors. The autodetect quirk for Roland devices catches the audio interface but misses the MIDI part, so a specific quirk is needed. Signed-off-by: Daniel Mack <daniel@zonque.org> Reported-by: Rafa Lafuente <rafalafuente@gmail.com> Tested-by: Raphaël Doursenaud <raphael@doursenaud.fr> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: line6: Clamp values correctlyTakashi Iwai2015-03-051-3/+3
|/ | | | | | | | The usages of clamp() macro in sound/usb/line6/playback.c are just wrong, the low and high values are swapped. Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb: Fix support for Denon DA-300USB DAC (ID 154e:1003)Frank C Guenther2015-02-171-0/+3
| | | | | | | | | | | | | | | | Fix problem where playback of Denon DA-300USB DAC sometimes does not start and leads to error messages like "clock source 41 is not valid, cannot use". Solution: Treat this device the same as other Denon/Marantz devices in sound/usb/quirks.c. Tested with both PCM and DSD formats. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=93261 Signed-off-by: Frank C Guenther <bugzilla.frnkcg@spamgourmet.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Don't attempt to get Lifecam HD-5000 sample rateJoe Turner2015-02-173-0/+12
| | | | | | | | | | | Adds a quirk to disable the check that the sample rate has been set correctly, as the Lifecam does not support getting the sample rate. This means that we don't need to wait for the USB timeout when attempting to get the sample rate. Waiting for the timeout causes problems in some applications, which give up on the device acquisition process before it has had time to complete, resulting in no sound. [minor tidy up by tiwai] Signed-off-by: Joe Turner <joe@oampo.co.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: line6: Improve line6_read/write_data() interfacesChris Rorvick2015-02-122-8/+14
| | | | | | | | | | The address cannot be negative so make it unsigned. Also, an unsigned int is always sufficient for the length, so no need to overdo it with a size_t. Finally, add in range checks to see if the values passed in actually fit where they are used. Signed-off-by: Chris Rorvick <chris@rorvick.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: line6: toneport: Use explicit type for firmware versionChris Rorvick2015-02-111-1/+1
| | | | | | | | | The firmware version is a single byte so have the variable type agree. Since the address to this member is passed to the read function, using an int is not even portable. Signed-off-by: Chris Rorvick <chris@rorvick.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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