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2013-07-15ALSA: usx2y: Fix unlocked snd_pcm_stop() callTakashi Iwai
snd_pcm_stop() must be called in the PCM substream lock context. Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-15ALSA: ua101: Fix unlocked snd_pcm_stop() callTakashi Iwai
snd_pcm_stop() must be called in the PCM substream lock context. Cc: <stable@vger.kernel.org> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-15ALSA: 6fire: Fix unlocked snd_pcm_stop() callTakashi Iwai
snd_pcm_stop() must be called in the PCM substream lock context. Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-10ALSA: usb-audio: fix regression for fixed stream quirkEldad Zack
Commit 8f898e92aea2c24c7f379ee265d178f69ebb9c07 removed the redundant reads of bInterfaceProtocol from the descriptors, but introduced a regression to devices with quirks of type QUIRK_AUDIO_FIXED_ENDPOINT, since fp->protocol is not set in setup process. As a consequence, audio streams would not get initialized, as the following logs show: [ 48.923043] setting usb interface 3:1 [ 48.923056] Creating new capture data endpoint #81 [ 48.923484] 4:3:1: cannot set freq 48000 to ep 0x81 This patch sets fp->protocol in create_fixed_stream_quirk() and resolves the regression. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-28ALSA: usb-audio: Add Audio Advantage Micro IIPrzemek Rudy
This patch is adding extensive support (beside standard usb audio class) for Audio Advantage Micro II usb sound card. Features included: - Access to AES bits (so now sending the IEC61937 compliant stream is possible). - Mixer SPDIF control added to turn on/off the optical transmitter. Signed-off-by: Przemek Rudy <prudy1@o2.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-28Merge branch 'full-roland-support' of ↵Takashi Iwai
git://git.alsa-project.org/alsa-kprivate into for-next For adding support for many Roland and Yamaha devices: * 'full-roland-support' of git://git.alsa-project.org/alsa-kprivate: ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE ALSA: usb-audio: claim autodetected PCM interfaces all at once ALSA: usb-audio: remove superfluous Roland quirks ALSA: usb-audio: add MIDI port names for some Roland devices ALSA: usb-audio: add support for many Roland/Yamaha devices ALSA: usb-audio: detect implicit feedback on Roland devices ALSA: usb-audio: store protocol version in struct audioformat
2013-06-27ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTUREClemens Ladisch
The Roland Quad/Octo-Capture devices use some unknown vendor-specific mechanism to switch sample rates (and to manage other controls). To prevent the driver from attempting to use any other than the default 44.1 kHz sample rate, use quirks to hide the other alternate settings. Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27ALSA: usb-audio: claim autodetected PCM interfaces all at onceClemens Ladisch
snd_card_register() registers all devices newly added since the last call. However, the playback/capture streams are handled as one ALSA device, so the second /dev device will not be registered if the PCM streams are added in two steps. QUIRK_AUTODETECT caused the probe callback to be called once for each interface, which triggered this problem. Work around this by handling this like the composite quirk, i.e., autodetecting all other interfaces that might be used for PCM or MIDI. Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27ALSA: usb-audio: remove superfluous Roland quirksClemens Ladisch
Remove all quirks that are no longer needed now that the generic Roland quirks can handle the vendor-specific descriptors correctly. Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27ALSA: usb-audio: add MIDI port names for some Roland devicesClemens Ladisch
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27ALSA: usb-audio: add support for many Roland/Yamaha devicesClemens Ladisch
Add quirks to detect the various vendor-specific descriptors used by Roland and Yamaha in most of their recent USB audio and MIDI devices. Together with the previous patch, this should add audio/MIDI support for the following USB devices: - Edirol motion dive .tokyo performance package - Roland MC-808 Synthesizer - Roland BK-7m Synthesizer - Roland VIMA JM-5/8 Synthesizer - Roland SP-555 Sequencer - Roland V-Synth GT Synthesizer - Roland Music Atelier AT-75/100/300/350C/500/800/900/900C Organ - Edirol V-Mixer M-200i/300/380/400/480/R-1000 - BOSS GT-10B Effects Processor - Roland Fantom G6/G7/G8 Keyboard - Cakewalk Sonar V-Studio 20/100/700 Audio Interface - Roland GW-8 Keyboard - Roland AX-Synth Keyboard - Roland JUNO-Di/STAGE/Gi Keyboard - Roland VB-99 Effects Processor - Cakewalk UM-2G MIDI Interface - Roland A-500S Keyboard - Roland SD-50 Synthesizer - Roland OCTAPAD SPD-30 Controller - Roland Lucina AX-09 Synthesizer - BOSS BR-800 Digital Recorder - Roland DUO/TRI-CAPTURE (EX) Audio Interface - BOSS RC-300 Loop Station - Roland JUPITER-50/80 Keyboard - Roland R-26 Recorder - Roland SPD-SX Controller - BOSS JS-10 Audio Player - Roland TD-11/15/30 Drum Module - Roland A-49/88 Keyboard - Roland INTEGRA-7 Synthesizer - Roland R-88 Recorder Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27ALSA: usb-audio: detect implicit feedback on Roland devicesClemens Ladisch
All the Roland/Edirol/BOSS USB audio devices that need implicit feedback show this unambiguously in their descriptors, so it might be a good idea to let the driver detect this. This should make playback work correctly (at least with Jack) with the following devices: - BOSS GT-100 - BOSS JS-8 Jam Station - Edirol M-16DX - Roland GAIA SH-01 Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27ALSA: usb-audio: store protocol version in struct audioformatClemens Ladisch
Instead of reading bInterfaceProtocol from the descriptor whenever it's needed, store this value in the audioformat structure. Besides simplifying some code, this will allow us to correctly handle vendor- specific devices where the descriptors are marked with other values. Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-24Add M2Tech hiFace USB-SPDIF driverAntonio Ospite
Add driver for M2Tech hiFace USB-SPDIF interface and compatible devices. M2Tech hiFace and compatible devices offer a Hi-End S/PDIF Output Interface, see http://www.m2tech.biz/hiface.html The supported products are: * M2Tech Young * M2Tech hiFace * M2Tech North Star * M2Tech W4S Young * M2Tech Corrson * M2Tech AUDIA * M2Tech SL Audio * M2Tech Empirical * M2Tech Rockna * M2Tech Pathos * M2Tech Metronome * M2Tech CAD * M2Tech Audio Esclusive * M2Tech Rotel * M2Tech Eeaudio * The Chord Company CHORD * AVA Group A/S Vitus Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-21ALSA: usb: uniform style used in MODULE_SUPPORTED_DEVICE()Antonio Ospite
In sound/usb/card.c and sound/usb/misc/ua101.c there are no spaces between the vendor and the device names, use this style in the other drivers too. This also helps keeping consistency when new drivers copies from the ones already in the mainline tree. Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-21ALSA: snd-usb-6fire: use vmalloc buffersAntonio Ospite
For USB devices it's not necessary to allocate physically contiguous buffers. Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-21ALSA: snd-usb-caiaq: use vmalloc buffersAntonio Ospite
For USB devices it's not necessary to allocate physically contiguous buffers. Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-21ALSA: snd-usb-caiaq: remove the unused snd_card_used variableAntonio Ospite
The snd_card_used variable is only read but never written, remove it. Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-18ALSA: sound/usb/misc/ua101.c: convert __list_for_each usage to list_for_eachDave Jones
Signed-off-by: Dave Jones <davej@redhat.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-17ALSA: usx2y: remove some old dead codeDan Carpenter
USB_QUEUE_BULK isn't defined any more. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-17ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam c310Takashi Iwai
Just like the previous fix for LogitechHD Webcam c270 in commit 11e7064f35bb87da8f427d1aa4bbd8b7473a3993, c310 model also requires the same workaround for avoiding the kernel warning. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59741 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-17ALSA: usb-audio: work around Android accessory firmware bugClemens Ladisch
When the Android firmware enables the audio interfaces in accessory mode, it always declares in the control interface's baInterfaceNr array that interfaces 0 and 1 belong to the audio function. However, the accessory interface itself, if also enabled, already is at index 0 and shifts the actual audio interface numbers to 1 and 2, which prevents the PCM streaming interface from being seen by the host driver. To get the PCM interface interface to work, detect when the descriptors point to the (for this driver useless) accessory interface, and redirect to the correct one. Reported-by: Jeremy Rosen <jeremy.rosen@openwide.fr> Tested-by: Jeremy Rosen <jeremy.rosen@openwide.fr> Cc: <stable@vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-05ALSA: usb-audio - Fix invalid volume resolution on Logitech HD webcam c270Takashi Iwai
USB audio driver spews an error message when probing Logitech HD webcam c270: ALSA mixer.c:1300 usb_audio: Warning! Unlikely big volume range (=6144), cval->res is probably wrong. ALSA mixer.c:1304 usb_audio: [5] FU [Mic Capture Volume] ch = 1, val = 1536/7680/1 Obviously the device needs a fixed volume resolution (cval->res = 384) like other Logitech devices. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=821735 Reported-and-tested-by: Cristian Rodríguez <crrodriguez@opensuse.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-04ALSA: usb-audio - Apply Logitech QuickCam Pro 9000 quirk only to audio ifaceTakashi Iwai
... instead of applying to all interfaces. Reference: http://forums.gentoo.org/viewtopic-p-6886404.html Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-03ALSA: usb-audio: fix Roland/Cakewalk UM-3G supportClemens Ladisch
Commit 927c9423dd5f2d1c0b93d5e694ab84b4a5559713 (ALSA: usb-audio: add Edirol UM-3G support) used a wrong quirk type, which would make the driver refuse to attach with the error message "MIDIStreaming interface descriptor not found". Cc: <stable@vger.kernel.org> # 3.3 and later Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-05-23ALSA: usb-6fire: Modify firmware version checkTorsten Schenk
Check only the uppermost 16 bits instead of the whole 32 bits of the version information. Do this because all firmware version tested with this version information worked correctly and the strict check causes problems for several users. Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-05-17ALSA: usb-audio: proc: use found syncmaxsize to determine feedback formatTorstein Hegge
freqshift is only set for the data endpoint and syncmaxsize is only set for the sync endpoint. This results in a syncmaxsize of zero used in the proc output feedback format calculation, which gives a feedback format incorrectly shown as 8.16 for UAC2 devices. As neither the data nor the sync endpoint gives all the relevant content, output the two combined. Also remove the sync_endpoint "packet size" which is always zero and the sync_endpoint "momentary freq" which is constant. Tested with UAC2 async and UAC1 adaptive, not tested with UAC1 async. Reported-by: B. Zhang <bb.zhang@free.fr> Signed-off-by: Torstein Hegge <hegge@resisty.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-30ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatchEldad Zack
Current code does this: be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]) Which is effectively (neglecting the index): be16_to_cpu(be16_to_cpu(*((u16 *) buf))) This means the int16 in the buffer is not converted at all. Daniel Mack confirmed that the driver works on little endian CPUs, leading to the conclusion that the device-side structure is actually little endian. This changes the code to use le16_to_cpu(). Caught by sparse. Acked-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-29ALSA: pcm_format_to_bits strong-typed conversionEldad Zack
Add a function to handle conversion from snd_pcm_format_t to bitwise with proper typing. Change such conversions to use this function and silence sparse warnings. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-29ALSA: USB: adjust for changed 3.8 USB APIClemens Ladisch
The recent changes in the USB API ("implement new semantics for URB_ISO_ASAP") made the former meaning of the URB_ISO_ASAP flag the default, and changed this flag to mean that URBs can be delayed. This is not the behaviour wanted by any of the audio drivers because it leads to discontinuous playback with very small period sizes. Therefore, our URBs need to be submitted without this flag. Reported-by: Joe Rayhawk <jrayhawk@fairlystable.org> Cc: <stable@vger.kernel.org> # 3.8 only Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-26ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sourcesDavid Henningsson
The Scarlett 2i2 seems to take almost 500 ms to set the sample rate, even if the clock is currently set to that value. This patch speeds up prepare of the device, by avoiding setting the clock to something it already is. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-25ALSA: usb-audio: USB quirk for Yamaha THR10CTrulan Martin
This patch adds a USB quirk for the Yamaha THR10C amp. Signed-off-by: Trulan Martin <trulanm@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-25ALSA: usb-audio: USB quirk for Yamaha THR5ATrulan Martin
This patch adds a USB quirk for the Yamaha THR5A amp. Signed-off-by: Trulan Martin <trulanm@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-25ALSA: usb-audio: USB quirk for Yamaha THR10Trulan Martin
This patch adds a USB quirk for the Yamaha THR10 amp. Signed-off-by: Trulan Martin <trulanm@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-25ALSA: usb-audio: Fix autopm error during probingTakashi Iwai
We've got strange errors in get_ctl_value() in mixer.c during probing, e.g. on Hercules RMX2 DJ Controller: ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4 ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4 .... It turned out that the culprit is autopm: snd_usb_autoresume() returns -ENODEV when called during card->probing = 1. Since the call itself during card->probing = 1 is valid, let's fix the return value of snd_usb_autoresume() as success. Reported-and-tested-by: Daniel Schürmann <daschuer@mixxx.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-25ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINTDaniel Mack
The USB_DT_CS_ENDPOINT class-specific endpoint descriptor is usually stuffed directly after the standard USB endpoint descriptor, and this is where the driver currently expects it to be. There are, however, devices in the wild that have it the other way around in their descriptor sets, so the USB_DT_CS_ENDPOINT comes *before* the standard enpoint. Devices known to implement it that way are "Sennheiser BTD-500" and Plantronics USB headsets. When the driver can't find the USB_DT_CS_ENDPOINT, it won't be able to change sample rates, as the bitmask for the validity of this command is storen in bmAttributes of that descriptor. Fix this by searching the entire interface instead of just the extra bytes of the first endpoint, in case the latter fails. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: Torstein Hegge <hegge@resisty.net> Reported-and-tested-by: Yves G <alsa-user@vivigatt.com> Cc: stable@kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-22ALSA: snd-usb-audio: set the timeout for usb control set messages to 5000 msDaniel Schürmann
Set the timeout for USB control set messages according to the USB 2 spec, using the macros from include/linux/usb.h. The get timout becomes 5000 ms even though it is 500 ms in the spec. This patch is required to run the Hercules RMX2 which needs a timeout of 1240 ms. More notes from author: I still distinguish between set and get but as long both are 5000 ms GCC will remove it anyway. IMHO this is more easy read and there is no need to explain why we use a get timeout for set messages. Signed-off-by: Daniel Schürmann <daschuer@mixxx.org> Acked-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-18Merge tag 'asoc-v3.10-2' of ↵Takashi Iwai
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: More updates for v3.10 The main additional change here is Lars-Peter's DMA work plus the platform conversions which have been tested - getting this in mainline will make life easier for development after the merge window. These factor a large chunk of code out of the drivers for the platforms using dmaengine, greatly simplifying development.
2013-04-18ALSA: snd-usb: add quirks handler for DSD streamsDaniel Mack
Unfortunately, none of the UAC standards provides a way to identify DSD (Direct Stream Digital) formats. Hence, this patch adds a quirks handler to identify USB interfaces that are capable of handling DSD. That quirks handler can augment the already parsed formats bit-field, by any of the new SNDRV_PCM_FMTBIT_DSD_{U8_U16} and setting the dsd_dop flag in the audio format, if the driver should take care for the DOP byte stuffing. The only devices that are known to work with this are the ones with a 'Playback Designs' vendor id. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-18ALSA: snd-usb: add support for bit-reversed byte formatsDaniel Mack
There is quite some confusion around the bit-ordering in DSD samples, and no general agreement that defines whether hardware is supposed to expect the oldest sample in the MSB or the LSB of a byte. ALSA will hence set the rule that on the software API layer, bytes always carry the oldest bit in the most significant bit of a byte, and the driver has to translate that at runtime in order to match the hardware layout. This patch adds support for this by adding a boolean flag to the audio format struct. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-18ALSA: snd-usb: add support for DSD DOP stream transportDaniel Mack
In order to provide a compatibility way for pushing DSD samples through ordinary PCM channels, the "DoP open Standard" was invented. See http://www.dsd-guide.com for the official document. The host is required to stuff DSD marker bytes (0x05, 0xfa, alternating) in the MSB of 24 bit wide samples on the bus, in addition to the 16 bits of actual DSD sample payload. To support this, the hardware and software stride logic in the driver has to be tweaked a bit, as we make the userspace believe we're operating on 16 bit samples, while we in fact push one more byte per channel down to the hardware. The DOP runtime information is stored in struct snd_usb_substream, so we can keep track of our state across multiple calls to prepare_playback_urb_dsd_dop(). Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-18ALSA: snd-usb: use ep->stride from urb callbacksDaniel Mack
For normal PCM transfer, this change has no effect, as the endpoint's stride is always frame_bits/8. For DSD DOP streams, however, which is added later, the hardware stride differs from the software stride, and the endpoint has the correct information in these cases. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-15ALSA: usb-audio: disable autopm for MIDI devicesClemens Ladisch
Commit 88a8516a2128 (ALSA: usbaudio: implement USB autosuspend) introduced autopm for all USB audio/MIDI devices. However, many MIDI devices, such as synthesizers, do not merely transmit MIDI messages but use their MIDI inputs to control other functions. With autopm, these devices would get powered down as soon as the last MIDI port device is closed on the host. Even some plain MIDI interfaces could get broken: they automatically send Active Sensing messages while powered up, but as soon as these messages cease, the receiving device would interpret this as an accidental disconnection. Commit f5f165418cab (ALSA: usb-audio: Fix missing autopm for MIDI input) introduced another regression: some devices (e.g. the Roland GAIA SH-01) are self-powered but do a reset whenever the USB interface's power state changes. To work around all this, just disable autopm for all USB MIDI devices. Reported-by: Laurens Holst Cc: <stable@vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-13ALSA: usb: Add quirk for 192KHz recording on E-Mu devicesCalvin Owens
When recording at 176.2KHz or 192Khz, the device adds a 32-bit length header to the capture packets, which obviously needs to be ignored for recording to work properly. Userspace expected: L0 L1 L2 R0 R1 R2 ...but actually got: R2 L0 L1 L2 R0 R1 Also, the last byte of the length header being interpreted as L0 of the first sample caused spikes every 0.5ms, resulting in a loud 16KHz tone (about the highest 'B' on a piano) being present throughout captures. Tested at all sample rates on an E-Mu 0404USB, and tested for regressions on a generic USB headset. Signed-off-by: Calvin Owens <jcalvinowens@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-10ALSA: snd-usb: Playback Design: use usb_set_inferface quirk from more locationsDaniel Mack
It turns out the devices from Playback Design need the delay quirk after usb_set_interface from clocks.c as well. Make it a proper quirks function and factor out the code to quirks.c. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-07ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*Eldad Zack
The usb_control_msg() function expects __u16 types and performs the endianness conversions by itself. However, in three places, a conversion is performed before it is handed over to usb_control_msg(), which leads to a double conversion (= no conversion): * snd_usb_nativeinstruments_boot_quirk() * snd_nativeinstruments_control_get() * snd_nativeinstruments_control_put() Caught by sparse: sound/usb/mixer_quirks.c:512:38: warning: incorrect type in argument 6 (different base types) sound/usb/mixer_quirks.c:512:38: expected unsigned short [unsigned] [usertype] index sound/usb/mixer_quirks.c:512:38: got restricted __le16 [usertype] <noident> sound/usb/mixer_quirks.c:543:35: warning: incorrect type in argument 5 (different base types) sound/usb/mixer_quirks.c:543:35: expected unsigned short [unsigned] [usertype] value sound/usb/mixer_quirks.c:543:35: got restricted __le16 [usertype] <noident> sound/usb/mixer_quirks.c:543:56: warning: incorrect type in argument 6 (different base types) sound/usb/mixer_quirks.c:543:56: expected unsigned short [unsigned] [usertype] index sound/usb/mixer_quirks.c:543:56: got restricted __le16 [usertype] <noident> sound/usb/quirks.c:502:35: warning: incorrect type in argument 5 (different base types) sound/usb/quirks.c:502:35: expected unsigned short [unsigned] [usertype] value sound/usb/quirks.c:502:35: got restricted __le16 [usertype] <noident> Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Acked-by: Daniel Mack <zonque@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: UAC2: support read-only freq controlEldad Zack
Some clocks might be read-only, e.g., external clocks (see also UAC2 4.7.2.1). In this case, setting the sample frequency will always fail (even if the rate is equal to the current clock rate), therefore do not write, but read the value and compare to the requested rate. If the clock is read only, avoid reading it twice. If it doesn't match, return -ENXIO since the clock is invalid for this configuration. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: show err in set_sample_rate_v2 debugEldad Zack
Show the error code returned from the USB subsystem in the debug messages. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: UAC2: auto clock selection module paramEldad Zack
Add a module param to disable auto clock selection. This is provided for users that expect the audio stream to fail when the clock source is invalid (e.g., the word clock was unintentionally disconnected). Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: UAC2: try to find and switch to valid clockEldad Zack
If a selector is available on a device, it may be pointing to a clock source which is currently invalid. If there is a valid clock source which can be selected, switch to it. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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