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2011-03-23ALSA: hda - Fix SPDIF out regression on ALC889Takashi Iwai1-1/+1
The commit 5a8cfb4e8ae317d283f84122ed20faa069c5e0c4 ALSA: hda - Use ALC_INIT_DEFAULT for really default initialization changed to use the default initialization method for ALC889, but this caused a regression on SPDIF output on some machines. This seems due to the COEF setup included in the default init procedure. For making SPDIF working again, the COEF-setup has to be avoided for the id 0889. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=24342 Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-23ALSA: usb-audio - Support for Boss JS-8 Jam StationKeith A. Milner1-0/+27
Signed-off-by: Keith A. Milner <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-23ALSA: usb-audio: add Cakewalk UM-1G supportClemens Ladisch1-0/+13
Add a quirk for the Cakewalk UM-1G USB MIDI interface in "advanced driver" mode. (It already works in standard mode.) Signed-off-by: Clemens Ladisch <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-23sound/oss/opl3: validate voice and channel indexesDan Rosenberg1-2/+13
User-controllable indexes for voice and channel values may cause reading and writing beyond the bounds of their respective arrays, leading to potentially exploitable memory corruption. Validate these indexes. Signed-off-by: Dan Rosenberg <[email protected]> Cc: [email protected] Signed-off-by: Takashi Iwai <[email protected]>
2011-03-23sound/oss: remove offset from load_patch callbacksDan Rosenberg5-26/+18
Was: [PATCH] sound/oss/midi_synth: prevent underflow, use of uninitialized value, and signedness issue The offset passed to midi_synth_load_patch() can be essentially arbitrary. If it's greater than the header length, this will result in a copy_from_user(dst, src, negative_val). While this will just return -EFAULT on x86, on other architectures this may cause memory corruption. Additionally, the length field of the sysex_info structure may not be initialized prior to its use. Finally, a signed comparison may result in an unintentionally large loop. On suggestion by Takashi Iwai, version two removes the offset argument from the load_patch callbacks entirely, which also resolves similar issues in opl3. Compile tested only. v3 adjusts comments and hopefully gets copy offsets right. Signed-off-by: Dan Rosenberg <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-23Merge branch 'topic/asoc' into for-linusTakashi Iwai5-4/+25
2011-03-23ALSA: HDA: Realtek: Avoid unnecessary volume control index on Surround/SideDavid Henningsson1-8/+17
Similar to commit 7e59e097c09b82760bb0fe08b0fa2b704d76c3f4, this patch avoids unnecessary volume control indices for more Realtek auto-parsers, e g the ALC66x family, on the "Surround" and "Side" controls. These indices cause these volume controls to be ignored by PulseAudio and vmaster and should be removed whenever possible. Cc: [email protected] Reported-by: Jan Losinski <[email protected]> Signed-off-by: David Henningsson <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-22ASoC: Support !REGULATOR build for sgtl5000Mark Brown1-0/+14
The regulator is optional depending on board design. Signed-off-by: Mark Brown <[email protected]> Acked-by: Liam Girdwood <[email protected]>
2011-03-22ALSA: hda - VIA: Fix VT1708 can't build up Headphone control issueLydia Wang1-3/+6
Since VT1708 didn't support the control of getting connection number, building of headphone control will fail in via_hp_build() function. Signed-off-by: Lydia Wang <[email protected]> Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-22ALSA: hda - VIA: Correct stream names for VT1818SLydia Wang1-0/+5
Correct stream names of analog playback and capture streams for VT1818S. Signed-off-by: Lydia Wang <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-22ALSA: hda - VIA: Fix codec type for VT1708BCE at the right timingLydia Wang1-4/+5
Add get_codec_type() in via_new_spec() function to make sure getting correct codec type before building mixer controls. Signed-off-by: Lydia Wang <[email protected]> Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-22ALSA: hda - VIA: Fix invalid A-A path volume adjust issueLydia Wang1-1/+8
Modify vt_auto_create_analog_input_ctls() function to fix invalid a-a path volume adjust issue for VT1708S, VT1702 and VT1716S codecs. Signed-off-by: Lydia Wang <[email protected]> Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-22ALSA: hda - VIA: Add missing support for VT1718S in A-A pathLydia Wang1-0/+5
Modify mute_aa_path() function to support VT1718S codec. Signed-off-by: Lydia Wang <[email protected]> Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-22ALSA: hda - VIA: Fix independent headphone no sound issueLydia Wang1-1/+11
Modify via_independent_hp_put() function to support VT1718S and VT1812 codecs, and fix independent headphone no sound issue. Signed-off-by: Lydia Wang <[email protected]> Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-22ALSA: hda - VIA: Fix stereo mixer recording no sound issueLydia Wang1-3/+6
Modify function via_mux_enum_put() to fix stereo mixer recording no sound issue. Signed-off-by: Lydia Wang <[email protected]> Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-21ALSA: hda - Set EAPD for Realtek ALC665Andres Mejia1-0/+2
Set EAPD for Realtek ALC665 (Vendor Id: 0x10eSet EAPD for Realtek ALC665 (Vendor Id: 0x10ec0665). Signed-off-by: Andres Mejia <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-21ALSA: usb - Remove trailing spaces from USB card name stringsTakashi Iwai1-4/+18
Some USB devices give trailing spaces in strings returned from usb_string(). This confuses the automatic card-id creation, resulting always in "default". This patch fixes the behavior by removing trailing spaces. Signed-off-by: Takashi Iwai <[email protected]>
2011-03-18sound: read i_size with i_size_read()Xiaochen Wang1-1/+1
Convert direct read of inode->i_size to using i_size_read(). i_size_read is guaranteed to return a valid value and its caller does not need to use addtional locking. Signed-off-by: Xiaochen Wang <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-18ASoC: Remove bogus check for register validity in debugfs writeMark Brown1-2/+0
Since not all registers need to be cached and the cache is entirely optional anyway we shouldn't be checking that a register is in the cached range. If the register is invalid then the actual I/O code can determine that and report an error. Similarly, the step size can and should be enforced by the lower level code if it's important. Signed-off-by: Mark Brown <[email protected]> Acked-by: Liam Girdwood <[email protected]>
2011-03-18Merge branch 'topic/asoc' of ↵Mark Brown1-6/+36
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-2.6.39
2011-03-18Merge branch 'topic/misc' into for-linusTakashi Iwai91-6530/+9928
2011-03-18ALSA: sound/pci/asihpi: check adapter index in hpi_ioctlDan Rosenberg1-0/+5
The user-supplied index into the adapters array needs to be checked, or an out-of-bounds kernel pointer could be accessed and used, leading to potentially exploitable memory corruption. Signed-off-by: Dan Rosenberg <[email protected]> Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-18ALSA: aloop - Fix possible IRQ lock inversionTakashi Iwai1-10/+9
loopback_pos_update() can be called in the timer callback, thus the lock held should be irq-safe. Otherwise you'll get AB/BA deadlock together with substream->self_group.lock. Reported-and-tested-by: Knut Petersen <[email protected]> Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-16Merge branch 'topic/hda' into for-linusTakashi Iwai9-509/+417
2011-03-16Merge branch 'topic/asoc' into for-linusTakashi Iwai138-2589/+14087
2011-03-16ALSA: sound/core: merge list_del()/list_add_tail() to list_move_tail()Nicolas Kaiser1-5/+3
Merge list_del() + list_add_tail() to list_move_tail(). Signed-off-by: Nicolas Kaiser <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-16Merge branch 'for-2.6.39' of ↵Takashi Iwai1-6/+36
git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
2011-03-16ALSA: ctxfi - use list_move() instead of list_del()/list_add() combinationKirill A. Shutemov1-2/+1
Signed-off-by: Kirill A. Shutemov <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-16ALSA: firewire - msleep needs delay.hStephen Rothwell1-0/+1
fixes this error: sound/firewire/fcp.c: In function 'fcp_avc_transaction': sound/firewire/fcp.c:103: error: implicit declaration of function 'msleep' Signed-off-by: Stephen Rothwell <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-15ALSA: firewire-lib, firewire-speakers: handle packet queueing errorsClemens Ladisch3-13/+37
Add an AMDTP stream error state that occurs when we fail to queue another packet. In this case, the stream is stopped, and the error can be reported when the application tries to restart the PCM stream. Signed-off-by: Clemens Ladisch <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-15ALSA: firewire-lib: allocate DMA buffer separatelyClemens Ladisch3-5/+16
For correct cache coherency on some architectures, DMA buffers must be allocated in a different cache line than data that is concurrently used by the CPU. Signed-off-by: Clemens Ladisch <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-15ALSA: firewire-lib: use no-info SYT for packets without SYT sampleClemens Ladisch1-4/+8
In non-blocking mode, the SYT_INTERVAL is larger than the number of audio frames in each packet, so there are packets that do not contain any frame to which the SYT could be applied. For these packets, the SYT must not be the timestamp of the next valid SYT frame, but the special no-info SYT value. This fixes broken playback on the FireWave at 44.1 kHz. Signed-off-by: Clemens Ladisch <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-15ALSA: add LaCie FireWire Speakers/Griffin FireWave Surround driverClemens Ladisch20-4/+2651
Add a driver for two playback-only FireWire devices based on the OXFW970 chip. v2: better AMDTP API abstraction; fix fw_unit leak; small fixes v3: cache the iPCR value v4: FireWave constraints; fix fw_device reference counting; fix PCR caching; small changes and fixes v5: volume/mute support; fix crashing due to pcm stop races v6: fix build; one-channel volume for LaCie v7: use signed values to make volume (range checks) work; fix function block IDs for volume/mute; always use channel 0 for LaCie volume Signed-off-by: Clemens Ladisch <[email protected]> Acked-by: Stefan Richter <[email protected]> Tested-by: Jay Fenlason <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-14ALSA: hda - Remove an unused variable in patch_realtek.cTakashi Iwai1-1/+0
Signed-off-by: Takashi Iwai <[email protected]>
2011-03-14ALSA: hda - pin-adc-mux-dmic auto-configuration of 92HD8X codecsVitaliy Kulikov1-73/+149
This patch replaces use of the harcoded arrays of pins, muxes, digital mics and adcs with the auto-generated ones using codec parsing and auto-discovers all actually connected digital mic pins on 92HD8X-like codecs This patch also adds the support for d-mic on pin 0x20. Signed-off-by: Vitaliy Kulikov <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-14ALSA: hda - fix digital mic selection in mixer on 92HD8X codecsVitaliy Kulikov1-3/+8
When the mux for digital mic is different from the mux for other mics, the current auto-parser doesn't handle them in a right way but provides only one mic. This patch fixes the issue. Signed-off-by: Vitaliy Kulikov <[email protected]> Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-14ALSA: hda - Move default input-src selection to init partTakashi Iwai1-12/+5
Move the default input-src selection code for alc268/269 to the init part instead of the parser. The input-src selection might be overwritten by init verbs. Signed-off-by: Takashi Iwai <[email protected]>
2011-03-14ALSA: hda - Initialize special cases for input src in init phaseTakashi Iwai1-3/+16
Currently some special handling for the unusual case like dual-ADCs or a single-input-src is done in the tree-parse time in set_capture_mixer(). But this setup could be overwritten by static init verbs. This patch moves the initialization into the init phase so that such input-src setup won't be lost. Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-14ALSA: ctxfi - Clear input settings before initializationPrzemyslaw Bruski1-0/+2
Clear input settings before initialization. Signed-off-by: Przemyslaw Bruski <[email protected]> Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-14ALSA: ctxfi - Fix SPDIF status retrievalPrzemyslaw Bruski1-14/+5
SDPIF status retrieval always returned the default settings instead of the actual ones. Signed-off-by: Przemyslaw Bruski <[email protected]> Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-14ALSA: ctxfi - Fix incorrect SPDIF status bit maskPrzemyslaw Bruski1-1/+1
SPDIF status mask creation was incorrect. Signed-off-by: Przemyslaw Bruski <[email protected]> Cc: <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-14ALSA: ctxfi - Fix microphone boost codes/commentsPrzemyslaw Bruski1-10/+18
microphone boost was set at +12dB, not +20dB (like in Windows driver and in adc_conf structure declaration), some comments added. Signed-off-by: Przemyslaw Bruski <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-11ALSA: atiixp - Fix wrong time-out checks during ac-link resetTakashi Iwai2-2/+2
The time-out in snd_atiixp_aclink_reset() is wrongly checked, and it resulted in exiting from the loop at the first iteration. Reported-by: Amir Shamsuddin <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-11ALSA: intel8x0m: append 'm' to "r_intel8x0"Paul Bolle1-3/+3
Appending an 'm' will distinguish it from a similar struct in intel8x0.c Signed-off-by: Paul Bolle <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-11ALSA: intel8x0m: add 'm' as "suffix" to static functionsPaul Bolle1-49/+49
Adding an 'm' will distinguish them from identical names in intel8x0.c. Signed-off-by: Paul Bolle <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-11ALSA: intel8x0m: wait a bit before warm reset checkPaul Bolle1-1/+2
At every resume a laptop I use prints this message (at KERN_ERR level): ALSA sound/pci/intel8x0m.c:904: AC'97 warm reset still in progress? [0x2] The thing to note here is that 0x2 corresponds to ICH_AC97COLD. Ie, what seems to be happening is that the register involved indicated a warm reset for some time (as the ICH_AC97WARM bit was set) but by the time the warning is printed, and that same register is checked again, that bit is already cleared and only the ICH_AC97COLD bit is still set. It turns out a warm reset needs some time to settle, but it is currently checked right away. The test therefore fails the first time it is done and schedule_timeout_uninterruptible() will be called. Once we return from that jiffies is already (far) past end_time on this laptop, so we exit the loop, print a warning, and exit the function while the warm reset actually succeeded. A way to fix this is to call usleep_range() after writing to the register involved. A handful of tests suggest 500 usecs is a safe value. (This might punish the "finish cold reset" case, but on this laptop such a cold reset apparently never happens, so I can't say for sure.) While we're at it drop the extra single tick from end_time, as it looks rather silly. Signed-off-by: Paul Bolle <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-11ALSA: usbaudio: implement USB autosuspendOliver Neukum6-22/+118
Devices are autosuspended if no pcm nor midi channel is open Mixer devices may be opened. This way they are active when in use to play or record sound, but can be suspended while users have a mixer application running. [Small clean-ups using static inline by tiwai] Signed-off-by: Oliver Neukum <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-11ALSA: usbaudio: fix suspend/resumeOliver Neukum3-3/+34
- ESHUTDOWN must be correctly handled - the optional interrupt endpoint's URB must be stopped and restarted Signed-off-by: Oliver Neukum <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2011-03-11Merge branch 'fix/misc' into topic/miscTakashi Iwai46-172/+348
2011-03-11ASoC: mini2440: Fix uda134x codec problem.Marek Belisko3-2/+11
ASoC audio for mini2440 platform in current kenrel doesn't work. First problem is samsung_asoc_dma device is missing in initialization. Next problem is with codec. Codec is initialized but never probed because no platform_device exist for codec driver. It leads to errors during codec binding to asoc dai. Next problem was platform data which was passed from board to asoc main driver but not passed to codec when called codec_soc_probe(). Following patch should fix issues. But not sure if in correct way. Please review. Signed-off-by: Marek Belisko <[email protected]> Acked-by: Liam Girdwood <[email protected]> Signed-off-by: Mark Brown <[email protected]> Cc: [email protected]
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