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2012-09-12ALSA: ice1724: Use linear scale for AK4396 volume control.Matteo Frigo
The AK4396 DAC has a linear-scale attentuator, but sound/pci/ice1712/prodigy_hifi.c used a log scale instead, which is not quite right. This patch restores the correct scale, borrowing from the ak4396 code in sound/pci/oxygen/oxygen.c. Signed-off-by: Matteo Frigo <athena@fftw.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-11ALSA: hda_intel: add position_fix quirk for Asus K53ECatalin Iacob
Commit c20c5a841cbe47f5b7812b57bd25397497e5fbc0 changed some chipsets to default to POS_FIX_COMBO so they now use POS_FIX_LPIB instead of POS_FIX_POSBUF. Since then I've been getting artifacts on playback, including repeated sounds on my Asus laptop. My hardware is Cougar Point which the commit log of c20c5a841cbe47f5b7812b57bd25397497e5fbc0 mentions as tested so POS_FIX_COMBO probably works in general but apparently it doesn't on Asus K53E therefore the need for the quirk. Signed-off-by: Catalin Iacob <iacobcatalin@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-11ALSA: compress_core: fix open flags test in snd_compr_open()Dan Carpenter
O_RDONLY is zero so the original test (f->f_flags & O_RDONLY) is always false and it will never do compress capture. The test for O_WRONLY is also slightly off. The original test would consider "->flags = (O_WRONLY | O_RDWR)" as write only instead of rejecting it as invalid. I've also removed the pr_err() because that could flood dmesg. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-10ALSA: hda - Fix Oops at codec reset/reconfigTakashi Iwai
snd_hda_codec_reset() calls restore_pincfgs() where the codec is powered up again, which eventually tries to resume and initialize via the callbacks of the codec. However, it's the place just after codec free callback, thus no codec callbacks should be called after that. On a codec like CS4206, it results in Oops due to the access in init callback. This patch fixes the issue by clearing the codec callbacks properly after freeing codec. Reported-by: Daniel J Blueman <daniel@quora.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-06ALSA: usb-audio: Fix bogus error messages for delay accountingTakashi Iwai
The recent fix for the missing fine delayed time adjustment gives strange error messages at each start of the playback stream, such as delay: estimated 0, actual 352 delay: estimated 353, actual 705 These come from the sanity check in retire_playback_urb(). Before the stream is activated via start_endpoints(), a few silent packets have been already sent. And at this point the delay account is still in the state as if the new packets are just queued, so the driver gets confused and spews the bogus error messages. For fixing the issue, we just need to check whether the received packet is valid, whether it's zero sized or not. Reported-by: Markus Trippelsdorf <markus@trippelsdorf.de> Cc: <stable@vger.kernel.org> [v3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-06ALSA: hda - Fix missing Master volume for STAC9200/925xTakashi Iwai
With the commit [2faa3bf: ALSA: hda - Rewrite the mute-LED hook with vmaster hook in patch_sigmatel.c], the former Master volume control was converted to PCM. This was supposed to be covered by the vmaster control. But due to the lack of "PCM" slave definition, this didn't happen properly. The patch fixes the missing entry. Reported-by: Andrew Shadura <bugzilla@tut.by> Cc: <stable@vger.kernel.org> [v3.4+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31ALSA: snd-usb: fix cross-interface streaming devicesDaniel Mack
Commit 68e67f40b ("ALSA: snd-usb: move calls to usb_set_interface") saved us some unnecessary calls to snd_usb_set_interface() but ignored the fact that there is at least one device out there which operates on two endpoint in different interfaces simultaniously. Take care for this by catching the case where data and sync endpoints are located on different interfaces and calling snd_usb_set_interface() between the start of the two endpoints. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Robert M. Albrecht <linux@romal.de> Cc: stable@kernel.org [v3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31ALSA: snd-usb: fix calls to next_packet_sizeDaniel Mack
In order to support devices with implicit feedback streaming models, packet sizes are now stored with each individual urb, and the PCM handling code which fills the buffers purely relies on the size fields now. However, calling snd_usb_audio_next_packet_size() for all possible packets in an URB at once, prior to letting the PCM code do its job does in fact not lead to the same behaviour than what the old code did: The PCM code will break its loop once a period boundary is reached, consequently using up less packets that it really could. As snd_usb_audio_next_packet_size() implements a feedback mechanism to the endpoints phase accumulator, the number of calls to that function matters, and when called too often, the data rate runs out of bounds. Fix this by making the next_packet function public, and call it from the PCM code as before if the packet data sizes are not defined. Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: stable@kernel.org [v3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31ALSA: snd-usb: restore delay informationDaniel Mack
Parts of commit 294c4fb8 ("ALSA: usb: refine delay information with USB frame counter") were unfortunately lost during the refactoring of the snd-usb driver in 3.5. This patch adds them back, restoring the correct delay information behaviour. Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: stable@kernel.org [3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31ALSA: snd-usb: use list_for_each_safe for endpoint resourcesPavel Roskin
snd_usb_endpoint_free() frees the structure that contains its argument. Signed-off-by: Pavel Roskin <proski@gnu.org> Cc: stable@vger.kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-30ALSA: snd-usb: Fix URB cancellation at stream startDaniel Mack
Commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream") fixed a scheduling-while-atomic bug that happened when snd_usb_endpoint_start was called from the trigger callback, which is an atmic context. However, the patch breaks the idea of the endpoints reference counting, which is the reason why the driver has been refactored lately. Revert that commit and let snd_usb_endpoint_start() take care of the URB cancellation again. As this function is called from both atomic and non-atomic context, add a flag to denote whether the function may sleep. Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: stable@kernel.org [3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-28ALSA: hda - Don't trust codec EPSS bit for IDT 92HD83xx & coTakashi Iwai
These codecs seem reporting EPSS but require longer delay for the proper D3 transition. For example, D3_STOP_CLOCK_OK bit won't be set correctly even after D3. In this patch, codec->epss flag is overridden for avoid the misbehavior. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-28ALSA: hda - Avoid unnecessary parameter read for EPSSTakashi Iwai
EPSS parameter should be static, so we can read it once and remember. This also allows more easily to override the wrong EPSS capability reported from a codec by changing the flag in the codec initialization step. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-22ALSA: hda - Do not set GPIOs for speakers on IDT if there are no speakersDavid Henningsson
This fixes an issue with a machine where there were no speakers, but GPIO0 had to be data=1 for the headphone to be functioning. I'm not sure if we need a more advanced patch to solve all possible cases, but if so, this patch would still provide a minor optimisation. BugLink: https://bugs.launchpad.net/bugs/1040077 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-21ALSA: snd-als100: fix suspend/resumeOndrej Zary
snd_card_als100_probe() does not set pcm field in struct snd_sb. As a result, PCM is not suspended and applications don't know that they need to resume the playback. Tested with Labway A381-F20 card (ALS120). Signed-off-by: Ondrej Zary <linux@rainbow-software.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20ALSA: hda - Fix leftover codec->power_transitionTakashi Iwai
When the codec turn-on operation is canceled by the immediate power-on, the driver left the power_transition flag as is. This caused the persistent avoidance of power-save behavior. Cc: <stable@vger.kernel.org> [v3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20Merge tag 'asoc-3.6' of ↵Takashi Iwai
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Additional updates for 3.6 A batch more bugfixes, all driver-specific and fairly small and unremarkable in a global context. The biggest batch are for the newly added Arizona drivers.
2012-08-20Merge branch 'topic/ca0132-fix' into for-linusTakashi Iwai
This is a series of fixes for CA0132, especially the missing SPDIF I/O and the mixer build errors.
2012-08-20ALSA: hda - don't create dysfunctional mixer controls for ca0132David Henningsson
It's possible that these amps are settable somehow, e g through secret codec verbs, but for now, don't create the controls (as they won't be working anyway, and cause errors in amixer). Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/1038651 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20ALSA: sound/ppc/snd_ps3.c: fix error return codeJulia Lawall
Initialize ret before returning on failure, as done elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20ALSA: sound/pci/rme9652/hdspm.c: fix error return codeJulia Lawall
Convert a nonnegative error return code to a negative one, as returned elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20ALSA: sound/pci/sis7019.c: fix error return codeJulia Lawall
Initialize rc before returning on failure, as done elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20ALSA: sound/pci/ctxfi/ctatc.c: fix error return codeJulia Lawall
Initialize err before returning on failure, as done elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20ALSA: sound/atmel/ac97c.c: fix error return codeJulia Lawall
In the first case, the second test of whether retval is negative is redundant. It is dropped and the previous and subsequent tests are combined. In the second case, add an initialization of retval on failure of ioremap. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20ALSA: sound/atmel/abdac.c: fix error return codeJulia Lawall
Initialize retval before returning from a failed call to ioremap. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20sound: oss/sb_audio: prevent divide by zero bugDan Carpenter
Speed comes from get_user() in audio_ioctl(). We use it to set the "s" variable before clamping it to valid values so it could lead to a divide by zero bug. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-17ASoC: wm9712: Fix inverted capture volumeMark Brown
The capture volume increases with the register value so it shouldn't be flagged as inverted. Reported-by: Christoph Fritz <chf.fritz@googlemail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-17ASoC: wm9712: Fix microphone source selectionMark Brown
Currently the microphone input source is not selectable as while there is a DAPM widget it's not connected to anything so it won't be properly instantiated. Add something more correct for the input structure to get things going, even though it's not hooked into the rest of the routing map and so won't actually achieve anything except allowing the relevant register bits to be written. Reported-by: Christop Fritz <chf.fritz@googlemail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
2012-08-17ASoC: wm5102: Remove DRC2Mark Brown
It will be removed from future device revisions. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-16ALSA: hda - Don't send invalid volume knob command on IDT 92hd75bxxDavid Henningsson
Instead of blindly initializing a volume knob widget, first check that there actually is a volume knob widget. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-16ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture streamTakashi Iwai
A PCM capture stream on usb-audio causes a scheduling-while-atomic BUG, as reported in the bugzilla entry below. It's because snd_usb_endpoint_start() is called at first at trigger START for a capture stream, and this function contains the left-over EP deactivation codes. The problem doesn't happen for a playback stream because the function is called at PCM prepare time, which can sleep. This patch fixes the BUG by moving the EP deactivation code into the PCM prepare callback. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46011 Cc: <stable@vger.kernel.org> [v3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-14ALSA: lx6464es: Add a missing error checkTakashi Iwai
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=44541 Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-14ALSA: hda - Fix 'Beep Playback Switch' with no underlying mute switchDavid Henningsson
Some Conexant devices (e g CX20590) have no mute capability on their Beep widgets. This patch makes sure we don't try setting mutes on those widgets. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-13ASoC: jack: Always notify full jack statusMark Brown
Don't just notify for the bits we've updated, notify the full state of the jack otherwise users might get confused by misleading reports. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-13ASoC: wm5110: Add missing input PGA routesMark Brown
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-13ASoC: wm5102: Add missing input PGA routesMark Brown
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-13ALSA: hda - fix Copyright debug messageWang Xingchao
As spec said, 1 indicates no copyright is asserted. Signed-off-by: Wang Xingchao <xingchao.wang@intel.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-10ASoC: Samsung: Fix build errorSachin Kamat
Fixes the following build error: In file included from arch/arm/mach-exynos/include/mach/dma.h:24:0, from arch/arm/plat-samsung/include/plat/dma-ops.h:17, from arch/arm/plat-samsung/include/plat/dma.h:128, from sound/soc/samsung/pcm.c:23: arch/arm/plat-samsung/include/plat/dma-pl330.h:106:8: error: redefinition of ‘struct s3c2410_dma_client’ arch/arm/plat-samsung/include/plat/dma.h:40:8: note: originally defined here make[3]: *** [sound/soc/samsung/pcm.o] Error 1 Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com> Acked-by: Kukjin Kim <kgene.kim@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-10ALSA : hda - bug fix on checking the supported power states of a codecMengdong Lin
The return value of snd_hda_param_read() is -1 for an error, otherwise it's the supported power states of a codec. The supported power states is a 32-bit value. Bit 31 will be set to 1 if the codec supports EPSS, thus making "sup" negative. And the bit 28:5 is reserved as "0". So a negative value other than -1 shall be further checked. Please refer to High-Definition spec 7.3.4.12 "Supported Power States", thanks! Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-10ALSA: hda - Fix panned "Beep Playback Switch"David Henningsson
When "Beep Playback Switch" had a different value on left and right channels (such as muting left but not right, or vice versa), this could result in the right channel being ignored. This patch enables beep to be sounding from right channel only, and also give correct result back to userspace (e g amixer). Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-10ALSA: cs46xx - signedness bug in snd_cs46xx_codec_read()Dan Carpenter
This function returns its own error codes instead of normal negative error codes. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-09ASoC: core: Upgrade the severity of probe deferral errors to dev_err()Mark Brown
In the past when ASoC had a custom probe deferral mechanism people complained about the logspam it generated and didn't want to know about the fact that we were doing probe deferral so all the error messages for it were at dev_dbg(), making diagnostics hard. Now that we have probe deferral as an accepted thing and it's generating log messages anyway there's no need to worry about this so upgrade the severity of all the probe deferral sources to dev_err() so that they are displayed by default. Also add one for missing aux_devs since there wasn't one. Reported-by: Russell King <linux@arm.linux.org.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-09ALSA: hda_intel: Add Device IDs for Intel Lynx Point-LP PCHJames Ralston
This patch adds the Intel HD Audio Device IDs for the Intel Lynx Point-LP PCH Signed-off-by: James Ralston <james.d.ralston@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-09ALSA: platform: Check CONFIG_PM_SLEEP instead of CONFIG_PMTakashi Iwai
When CONFIG_PM is set but CONFIG_PM_SLEEP is unset, SIMPLE_DEV_PM_OPS() ignores the given functions, and this leads to compile warnings. For avoiding this, simply check CONFIG_PM_SLEEP instead of CONFIG_PM. Reported-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-09ASoC: wm8994: Add missing dapm routes for WM8958 rev AChris Rattray
Signed-off-by: Chris Rattray <crattray@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-09ASoC: wm8962: Don't duplicate bias level management in resumeMark Brown
The core will bring the bias level up for us since we use idle_bias_off, duplicating this may be harmful. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-09ASoC: bfin: fix memory leak in sport3 controller driverScott Jiang
Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-09ASoC: Davinci: McASP: Flush the FIFO before enablingVaibhav Bedia
FIFO should be flushed before it is enabled for the first time. This fixes the I/O errors reported by the ASoC core on a fresh boot Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com> Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-09ALSA: hda - Fix pop noise in headphones on S3 for Asus X55A, X55VDavid Henningsson
To turn off pin control for the pin was tested, and helped against this issue. BugLink: https://bugs.launchpad.net/bugs/1034779 Tested-by: Chih-Hsyuan Ho <chih.ho@canonical.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-08ALSA: hda - Add missing SPDIF I/O setup for CA0132Takashi Iwai
CA0132 driver had some codes to handle the S/PDIF I/O, but the actual setups of pins and converters were missing. Now the pins are added. Also, fixed a few points triggering invalid codec verbs and mixer elements since the digital I/O audio widgets on CA0132 have no amp. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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