summaryrefslogtreecommitdiff
path: root/sound
AgeCommit message (Collapse)Author
2013-06-29ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam c310Takashi Iwai
commit 36691e1be6ec551eef4a5225f126a281f8c051c2 upstream. 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 Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-06-29ALSA: usb-audio: work around Android accessory firmware bugClemens Ladisch
commit 342cda29343a6272c630f94ed56810a76740251b upstream. 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> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-06-19ALSA: usb-audio - Fix invalid volume resolution on Logitech HD webcam c270Takashi Iwai
commit 11e7064f35bb87da8f427d1aa4bbd8b7473a3993 upstream. 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> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-06-19ALSA: usb-audio - Apply Logitech QuickCam Pro 9000 quirk only to audio ifaceTakashi Iwai
commit 8eafc0a161123d90617c9ca2eddfe87b382b1b89 upstream. ... instead of applying to all interfaces. Reference: http://forums.gentoo.org/viewtopic-p-6886404.html Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-06-19ALSA: usb-audio: avoid integer overflow in create_fixed_stream_quirk()Xi Wang
commit 8866f405efd4171f9d9c91901d2dd02f01bacb60 upstream. A malicious USB device could feed in a large nr_rates value. This would cause the subsequent call to kmemdup() to allocate a smaller buffer than expected, leading to out-of-bounds access. This patch validates the nr_rates value and reuses the limit introduced in commit 4fa0e81b ("ALSA: usb-audio: fix possible hang and overflow in parse_uac2_sample_rate_range()"). Signed-off-by: Xi Wang <xi.wang@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-06-19ALSA: usb-audio: fix possible hang and overflow in ↵Xi Wang
parse_uac2_sample_rate_range() commit 4fa0e81b83503900be277e6273a79651b375e288 upstream. A malicious USB device may feed in carefully crafted min/max/res values, so that the inner loop in parse_uac2_sample_rate_range() could run for a long time or even never terminate, e.g., given max = INT_MAX. Also nr_rates could be a large integer, which causes an integer overflow in the subsequent call to kmalloc() in parse_audio_format_rates_v2(). Thus, kmalloc() would allocate a smaller buffer than expected, leading to a memory corruption. To exploit the two vulnerabilities, an attacker needs physical access to the machine to plug in a malicious USB device. This patch makes two changes. 1) The type of "rate" is changed to unsigned int, so that the loop could stop once "rate" is larger than INT_MAX. 2) Limit nr_rates to 1024. Suggested-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Xi Wang <xi.wang@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-05-30ALSA: HDA: Fix Oops caused by dereference NULL pointerWang YanQing
commit 2195b063f6609e4c6268f291683902f25eaf9aa6 upstream. The interrupt handler azx_interrupt will call azx_update_rirb, which may call snd_hda_queue_unsol_event, snd_hda_queue_unsol_event will dereference chip->bus pointer. The problem is we alloc chip->bus in azx_codec_create which will be called after we enable IRQ and enable unsolicited event in azx_probe. This will cause Oops due dereference NULL pointer. I meet it, good luck:) [Rearranged the NULL check before the tracepoint and added another NULL check of bus->workq -- tiwai] Signed-off-by: Wang YanQing <udknight@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-05-30ASoC: wm8994: missing break in wm8994_aif3_hw_params()Dan Carpenter
commit 4495e46fe18f198366961bb2b324a694ef8a9b44 upstream. The missing break here means that we always return early and the function is a no-op. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-05-13vm: convert snd_pcm_lib_mmap_iomem() to vm_iomap_memory() helperLinus Torvalds
commit 0fe09a45c4848b5b5607b968d959fdc1821c161d upstream. This is my example conversion of a few existing mmap users. The pcm mmap case is one of the more straightforward ones. Acked-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-05-13ALSA: usb-audio: Fix autopm error during probingTakashi Iwai
commit 60af3d037eb8c670dcce31401501d1271e7c5d95 upstream. 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> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-05-13ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINTDaniel Mack
commit ebfc594c02148b6a85c2f178cf167a44a3c3ce10 upstream. 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> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-05-13ASoC: max98088: Fix logging of hardware revision.Dylan Reid
commit 98682063549bedd6e2d2b6b7222f150c6fbce68c upstream. The hardware revision of the codec is based at 0x40. Subtract that before convering to ASCII. The same as it is done for 98095. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-05-13ALSA: usb-audio: disable autopm for MIDI devicesClemens Ladisch
commit cbc200bca4b51a8e2406d4b654d978f8503d430b upstream. 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 Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-05-13ALSA: usb: Add quirk for 192KHz recording on E-Mu devicesCalvin Owens
commit 1539d4f82ad534431cc67935e8e442ccf107d17d upstream. 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> Signed-off-by: Takashi Iwai <tiwai@suse.de> [bwh: Backported to 3.2: adjust filenames, context] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-04-25ALSA: hda - fix typo in proc outputDavid Henningsson
commit aeb3a97222832e5457c4b72d72235098ce4bfe8d upstream. Rename "Digitial In" to "Digital In". This function is only used for proc output, so should not cause any problems to change. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-04-25ALSA: hda - Enabling Realtek ALC 671 codecRainer Koenig
commit 1d87caa69c04008e09f5ff47b5e6acb6116febc7 upstream. * Added the device ID to the modalias list and assinged ALC662 patches for it * Added 4 port support for the device ID 0671 in alc662_parse_auto_config Signed-off-by: Rainer Koenig <Rainer.Koenig@ts.fujitsu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-04-25ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is runningAlban Bedel
commit f1ca493b0b5e8f42d3b2dc8877860db2983f47b6 upstream. The Charge Pump needs the DSP clock to work properly, without it the bypass to HP/LINEOUT is not working properly. This requirement is not mentioned in the datasheet but has been confirmed by Mark Brown from Wolfson. Signed-off-by: Alban Bedel <alban.bedel@avionic-design.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-04-25ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*Eldad Zack
commit 889d66848b12d891248b03abcb2a42047f8e172a upstream. 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> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-04-10ASoC: dma-sh7760: Fix compile errorLars-Peter Clausen
commit 417a1178f1bf3cdc606376b3ded3a22489fbb3eb upstream. The dma-sh7760 currently fails with the following compile error: sound/soc/sh/dma-sh7760.c:346:2: error: unknown field 'pcm_ops' specified in initializer sound/soc/sh/dma-sh7760.c:346:2: warning: initialization from incompatible pointer type sound/soc/sh/dma-sh7760.c:347:2: error: unknown field 'pcm_new' specified in initializer sound/soc/sh/dma-sh7760.c:347:2: warning: initialization makes integer from pointer without a cast sound/soc/sh/dma-sh7760.c:348:2: error: unknown field 'pcm_free' specified in initializer sound/soc/sh/dma-sh7760.c:348:2: warning: initialization from incompatible pointer type sound/soc/sh/dma-sh7760.c: In function 'sh7760_soc_platform_probe': sound/soc/sh/dma-sh7760.c:353:2: warning: passing argument 2 of 'snd_soc_register_platform' from incompatible pointer type include/sound/soc.h:368:5: note: expected 'struct snd_soc_platform_driver *' but argument is of type 'struct snd_soc_platform *' This is due the misnaming of the snd_soc_platform_driver type name and 'ops' field. The issue was introduced in commit f0fba2a("ASoC: multi-component - ASoC Multi-Component Support"). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-04-10ASoC: imx-ssi: Fix occasional AC97 reset failureSascha Hauer
commit b6e51600f4e983e757b1b6942becaa1ae7d82e67 upstream. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> [bwh: Backported to 3.2: adjust filename] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-03-27ALSA: hda - Fix typo in checking IEC958 emphasis bitTakashi Iwai
commit a686fd141e20244ad75f80ad54706da07d7bb90a upstream. There is a typo in convert_to_spdif_status() about checking the emphasis IEC958 status bit. It should check the given value instead of the resultant value. Reported-by: Martin Weishart <martin.weishart@telosalliance.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-03-27ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()Daniel Mack
commit 83ea5d18d74f032a760fecde78c0210f66f7f70c upstream. Creation of individual mixer controls may fail, but that shouldn't cause the entire mixer creation to fail. Even worse, if the mixer creation fails, that will error out the entire device probing. All the functions called by parse_audio_unit() should return -EINVAL if they find descriptors that are unsupported or believed to be malformed, so we can safely handle this error code as a non-fatal condition in snd_usb_mixer_controls(). That fixes a long standing bug which is commonly worked around by adding quirks which make the driver ignore entire interfaces. Some of them might now be unnecessary. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: Rodolfo Thomazelli <pe.soberbo@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-03-27ALSA: snd-usb: mixer: propagate errors up the call chainDaniel Mack
commit 4d7b86c98e445b075c2c4c3757eb6d3d6efbe72e upstream. In check_input_term() and parse_audio_feature_unit(), propagate the error value that has been returned by a failing function instead of -EINVAL. That helps cleaning up the error pathes in the mixer. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-03-27ALSA: hda/cirrus - Fix the digital beep registrationTakashi Iwai
commit a86b1a2cd2f81f74e815e07f756edd7bc5b6f034 upstream. The argument passed to snd_hda_attach_beep_device() is a widget NID while spec->beep_amp holds the composed value for amp controls. Signed-off-by: Takashi Iwai <tiwai@suse.de> [bwh: Backported to 3.2: adjust context] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-03-20ALSA: seq: Fix missing error handling in snd_seq_timer_open()Takashi Iwai
commit 66efdc71d95887b652a742a5dae51fa834d71465 upstream. snd_seq_timer_open() didn't catch the whole error path but let through if the timer id is a slave. This may lead to Oops by accessing the uninitialized pointer. BUG: unable to handle kernel NULL pointer dereference at 00000000000002ae IP: [<ffffffff819b3477>] snd_seq_timer_open+0xe7/0x130 PGD 785cd067 PUD 76964067 PMD 0 Oops: 0002 [#4] SMP CPU 0 Pid: 4288, comm: trinity-child7 Tainted: G D W 3.9.0-rc1+ #100 Bochs Bochs RIP: 0010:[<ffffffff819b3477>] [<ffffffff819b3477>] snd_seq_timer_open+0xe7/0x130 RSP: 0018:ffff88006ece7d38 EFLAGS: 00010246 RAX: 0000000000000286 RBX: ffff88007851b400 RCX: 0000000000000000 RDX: 000000000000ffff RSI: ffff88006ece7d58 RDI: ffff88006ece7d38 RBP: ffff88006ece7d98 R08: 000000000000000a R09: 000000000000fffe R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: ffff8800792c5400 R14: 0000000000e8f000 R15: 0000000000000007 FS: 00007f7aaa650700(0000) GS:ffff88007f800000(0000) GS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00000000000002ae CR3: 000000006efec000 CR4: 00000000000006f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 Process trinity-child7 (pid: 4288, threadinfo ffff88006ece6000, task ffff880076a8a290) Stack: 0000000000000286 ffffffff828f2be0 ffff88006ece7d58 ffffffff810f354d 65636e6575716573 2065756575712072 ffff8800792c0030 0000000000000000 ffff88006ece7d98 ffff8800792c5400 ffff88007851b400 ffff8800792c5520 Call Trace: [<ffffffff810f354d>] ? trace_hardirqs_on+0xd/0x10 [<ffffffff819b17e9>] snd_seq_queue_timer_open+0x29/0x70 [<ffffffff819ae01a>] snd_seq_ioctl_set_queue_timer+0xda/0x120 [<ffffffff819acb9b>] snd_seq_do_ioctl+0x9b/0xd0 [<ffffffff819acbe0>] snd_seq_ioctl+0x10/0x20 [<ffffffff811b9542>] do_vfs_ioctl+0x522/0x570 [<ffffffff8130a4b3>] ? file_has_perm+0x83/0xa0 [<ffffffff810f354d>] ? trace_hardirqs_on+0xd/0x10 [<ffffffff811b95ed>] sys_ioctl+0x5d/0xa0 [<ffffffff813663fe>] ? trace_hardirqs_on_thunk+0x3a/0x3f [<ffffffff81faed69>] system_call_fastpath+0x16/0x1b Reported-and-tested-by: Tommi Rantala <tt.rantala@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-03-20ALSA: vmaster: Fix slave change notificationTakashi Iwai
commit 2069d483b39a603a5f3428a19d3b4ac89aa97f48 upstream. When a value of a vmaster slave control is changed, the ctl change notification is sometimes ignored. This happens when the master control overrides, e.g. when the corresponding master control is muted. The reason is that slave_put() returns the value of the actual slave put callback, and it doesn't reflect the virtual slave value change. This patch fixes the function just to return 1 whenever a slave value is changed. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-03-06ALSA: hda - hdmi: ELD shouldn't be valid after unplugDavid Henningsson
commit bbfd8a19b6913f50a362457c34d49bfafe5e456e upstream. Currently, eld_valid is never set to false, except at kernel module load time. This patch makes sure that eld is no longer valid when the cable is (hot-)unplugged. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-03-06ALSA: ali5451: remove irq enabling in pointer callbackDenis Efremov
commit dacae5a19b4cbe1b5e3a86de23ea74cbe9ec9652 upstream. snd_ali_pointer function is called with local interrupts disabled. However it seems very strange to reenable them in such way. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Denis Efremov <yefremov.denis@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-03-06ALSA: rme32.c irq enabling after spin_lock_irqDenis Efremov
commit f49a59c4471d81a233e09dda45187cc44fda009d upstream. According to the other code in this driver and similar code in rme96 it seems, that spin_lock_irq in snd_rme32_capture_close function should be paired with spin_unlock_irq. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Denis Efremov <yefremov.denis@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-03-06ALSA: aloop: Fix Oops while PM resumeTakashi Iwai
commit edac894389f9c9de2a1368c78809c824b343f3a5 upstream. snd-aloop driver has no proper PM implementation, thus the PM resume may trigger Oops due to leftover timer instance. This patch adds the missing suspend/resume implementation. Reported-and-tested-by: El boulangero <elboulangero@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> [bwh: Backported to 3.2: adjust context] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-03-06ALSA: hda - Release assigned pin/cvt at error path of hdmi_pcm_open()Takashi Iwai
commit 2ad779b7329d6894a80df94e693e72eaa0d56790 upstream. If the driver detects and invalid ELD, it gives an open error. But it forgot to release the assigned pin, converter and spdif ctls before returning. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-03-06ALSA: usb-audio: fix Roland A-PRO supportClemens Ladisch
commit 7da58046482fceb17c4a0d4afefd9507ec56de7f upstream. The quirk for the Roland/Cakewalk A-PRO keyboards accidentally used the wrong interface number, which prevented the driver from attaching to the device. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-02-06ALSA: usb-audio: Fix regression by disconnection-race-fix patchTakashi Iwai
[NOTE: the regression below is found only in 3.2-3.4 stable trees, so there is no upstream commit corresponding to this patch] The recent fix for the race at disconnection of usb-audio devices (upstream commit 978520b7) triggers Oops when a device is unplugged while playing on 3.2 and 3.4 kernels. The culprit is that the shutdown flag check was wrongly added around the urb deactivation code snippet. The urb deactivation code has to be performed even after the device disconnected. Otherwise it remains undead and pokes the wild access in the end. The regression fix is simply reverting the shutdown flag check in that code. Reported-and-tested-by: Chris J Arges <christopherarges@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-02-06ALSA: usb - fix race in creation of M-Audio Fast track pro driverDavid Henningsson
commit b98ae2729dea161edc96c9d177459b6c28bcbba5 upstream. A patch in the 3.2 kernel caused regression with hotplugging the M-Audio Fast track pro, or sound after suspend. I don't have the device so I haven't done a full analysis, but it seems userspace (both udev and pulseaudio) got confused when a card was created, immediately destroyed, and then created again. However, at least one person in the bug report (martin djfun) reports that this patch resolves the issue for him. It also leaves a message in the log: "snd-usb-audio: probe of 1-1.1:1.1 failed with error -5" which is a bit misleading. It is better than non-working audio, but maybe there's a more elegant solution? BugLink: https://bugs.launchpad.net/bugs/1095315 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-02-06ALSA: hda - Fix non-snoop page handlingTakashi Iwai
commit 9ddf1aeb2134e72275c97a2c6ff2e3eb04f2f27a upstream. For non-snoop mode, we fiddle with the page attributes of CORB/RIRB and the position buffer, but also the ring buffers. The problem is that the current code blindly assumes that the buffer is contiguous. However, the ring buffers may be SG-buffers, thus a wrong vmapped address is passed there, leading to Oops. This patch fixes the handling for SG-buffers. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=800701 Signed-off-by: Takashi Iwai <tiwai@suse.de> [bwh: Backported to 3.2: open-code snd_pcm_get_dma_buf()] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-02-06ALSA: usb-audio: fix invalid length check for RME and other UAC 2 devicesClemens Ladisch
commit d56268fb108c7c21e19933588ca4d94652585183 upstream. Commit 23caaf19b11e (ALSA: usb-mixer: Add support for Audio Class v2.0) forgot to adjust the length check for UAC 2.0 feature unit descriptors. This would make the code abort on encountering a feature unit without per-channel controls, and thus prevented the driver to work with any device having such a unit, such as the RME Babyface or Fireface UCX. Reported-by: Florian Hanisch <fhanisch@uni-potsdam.de> Tested-by: Matthew Robbetts <wingfeathera@gmail.com> Tested-by: Michael Beer <beerml@sigma6audio.de> Cc: Daniel Mack <daniel@caiaq.de> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-02-06ALSA: hda - Add Conexant CX20755/20756/20757 codec IDsTakashi Iwai
commit 42c364ace52ae6b4699105b39f2559c256b6cd4c upstream. These are just compatible with other CX2075x codecs. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-02-06ALSA: hda/conexant - Correct vendor IDs for new codecsTakashi Iwai
commit 2d825fd82eb765412a558a56e193b77117d56699 upstream. Never trust datasheet... Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-02-06ALSA: hda - Add Conexant CX20751/2/3/4 codec supportTakashi Iwai
commit 61d648fb4726f8a89c07cd1904f9c2e11bf26df5 upstream. These are almost compatible with the older Conexant codecs. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-02-06Revert "ALSA: hda - Shut up pins at power-saving mode with Conexnat codecs"David Henningsson
commit 7ed4165e2d01bdbbb4c1086eb73eadf0f64cbbf0 upstream. This reverts commit 697c373e34613609cb5450f98b91fefb6e910588. The original patch was meant to remove clicking, but in fact caused even more clicking instead. Thanks to c4pp4 for doing most of the work with this bug. BugLink: https://bugs.launchpad.net/bugs/886975 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> [bwh: Backported to 3.2: adjust context] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-02-06ALSA: pxa27x: fix ac97 warm resetMike Dunn
commit 3b4bc7bccc7857274705b05cf81a0c72cfd0b0dd upstream. This patch fixes some code that implements a work-around to a hardware bug in the ac97 controller on the pxa27x. A bug in the controller's warm reset functionality requires that the mfp used by the controller as the AC97_nRESET line be temporarily reconfigured as a generic output gpio (AF0) and manually held high for the duration of the warm reset cycle. This is what was done in the original code, but it was broken long ago by commit fb1bf8cd ([ARM] pxa: introduce processor specific pxa27x_assert_ac97reset()) which changed the mfp to a GPIO input instead of a high output. The fix requires the ac97 controller to obtain the gpio via gpio_request_one(), with arguments that configure the gpio as an output initially driven high. Tested on a palm treo 680 machine. Reportedly, this broken code only prevents a warm reset on hardware that lacks a pull-up on the line, which appears to be the case for me. Signed-off-by: Mike Dunn <mikedunn@newsguy.com> Signed-off-by: Igor Grinberg <grinberg@compulab.co.il> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> [bwh: Backported to 3.2: adjust context] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-02-06ALSA: pxa27x: fix ac97 cold resetMike Dunn
commit 41b645c8624df6ace020a8863ad1449d69140f7d upstream. Cold reset on the pxa27x currently fails and pxa2xx_ac97_try_cold_reset: cold reset timeout (GSR=0x44) appears in the kernel log. Through trial-and-error (the pxa270 developer's manual is mostly incoherent on the topic of ac97 reset), I got cold reset to complete by setting the WARM_RST bit in the GCR register (and later noticed that pxa3xx does this for cold reset as well). Also, a timeout loop is needed to wait for the reset to complete. Tested on a palm treo 680 machine. Signed-off-by: Mike Dunn <mikedunn@newsguy.com> Acked-by: Igor Grinberg <grinberg@compulab.co.il> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-02-06ASoC: wm2000: Fix sense of speech clarity enableMark Brown
commit 267f8fa2e1eef0612b2007e1f1846bcbc35cc1fa upstream. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-02-06ASoC: wm5100: Remove DSP B and left justified formatsMark Brown
commit 5f960294e2031d12f10c8488c3446fecbf59628d upstream. These are not supported Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-01-03ALSA: hda - Fix the wrong pincaps set in ALC861VD dallas/hp fixupTakashi Iwai
commit b78562b10fa66175e30b76073e32a0ad8d92aa83 upstream. The workaround to force VREF50 for dallas/hp model with ALC861VD was introduced in commit 8fdcb6fe4204bdb4c6991652717ab5063751414e, but it contained wrong pincap override bits. This patch fixes to exclude VREF80 pincap bit correctly. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-01-03ALSA: hda - Always turn on pins for HDMI/DPTakashi Iwai
commit 6169b673618bf0b2518ce413b54925782a603f06 upstream. We've seen the broken HDMI *video* output on some machines with GM965, and the debugging session pointed that the culprit is the disabled audio output pins. Toggling these pins dynamically on demand caused flickering of HDMI TV. This patch changes the behavior to keep the pin ON constantly. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=51421 Signed-off-by: Takashi Iwai <tiwai@suse.de> [bwh: Backported to 3.2: adjust context] Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-01-03ALSA: hda - Fix pin configuration of HP Pavilion dv7Takashi Iwai
commit 8ae5865ec77c22462c736846a0679947a6953548 upstream. Fix the quirk entry for HP Pavilion dv7 in order to make the bass speaker working. Reported-and-tested-by: Tomas Pospisek <tpo2@sourcepole.ch> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-01-03ALSA: hda - Add stereo-dmic fixup for Acer Aspire One 522Takashi Iwai
commit 63a077e27648b4043b1ca1b4e29f0c42d99616b6 upstream. Acer Aspire One 522 has the infamous digital mic unit that needs the phase inversion fixup for stereo. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=715737 Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-01-03ALSA: usb-audio: Fix missing autopm for MIDI inputTakashi Iwai
commit f5f165418cabf2218eb466c0e94693b8b1aee88b upstream. The commit [88a8516a: ALSA: usbaudio: implement USB autosuspend] added the support of autopm for USB MIDI output, but it didn't take the MIDI input into account. This patch adds the following for fixing the autopm: - Manage the URB start at the first MIDI input stream open, instead of the time of instance creation - Move autopm code to the common substream_open() - Make snd_usbmidi_input_start/_stop() more robust and add the running state check Reviewd-by: Clemens Ladisch <clemens@ladisch.de> Tested-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
2013-01-03ALSA: usb-audio: Avoid autopm calls after disconnectionTakashi Iwai
commit 59866da9e4ae54819e3c4e0a8f426bdb0c2ef993 upstream. Add a similar protection against the disconnection race and the invalid use of usb instance after disconnection, as well as we've done for the USB audio PCM. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=51201 Reviewd-by: Clemens Ladisch <clemens@ladisch.de> Tested-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Ben Hutchings <ben@decadent.org.uk>