summaryrefslogtreecommitdiffstats
path: root/sound
AgeCommit message (Collapse)AuthorFilesLines
2012-08-31ALSA: snd-usb: fix cross-interface streaming devicesDaniel Mack1-0/+15
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 Mack3-13/+8
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 Mack1-3/+26
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 Roskin1-2/+2
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 Mack3-11/+15
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 Iwai1-0/+1
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 Iwai2-2/+9
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 Henningsson1-0/+3
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 Zary1-1/+1
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 Iwai1-0/+2
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 Iwai32-267/+160
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 Iwai3-145/+99
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 Henningsson1-0/+8
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 Lawall1-0/+1
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 Lawall1-1/+1
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 Lawall1-2/+3
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 Lawall1-1/+3
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 Lawall1-8/+4
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 Lawall1-0/+1
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 Carpenter1-1/+3
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 Brown1-1/+1
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 Brown1-2/+17
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 Brown1-16/+0
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 Henningsson1-4/+5
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 Iwai2-4/+3
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 Iwai1-0/+2
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 Henningsson1-2/+11
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 Brown1-1/+1
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 Brown1-0/+12
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-13ASoC: wm5102: Add missing input PGA routesMark Brown1-0/+9
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-13ALSA: hda - fix Copyright debug messageWang Xingchao1-1/+1
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 Kamat1-1/+1
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 Lin1-1/+1
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 Henningsson1-6/+14
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 Carpenter1-1/+1
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 Brown1-4/+6
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 Ralston1-0/+9
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 Iwai7-9/+9
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 Rattray1-0/+2
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 Brown1-15/+0
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 Jiang1-0/+7
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 Bedia1-2/+8
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 Henningsson1-0/+8
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 Iwai1-6/+14
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>
2012-08-08ALSA: hda - Use the standard PCM ops for CA0132Takashi Iwai1-113/+29
Now with the workaround using codec->pcm_format_first flag, we can clean up the home-baked codes in patch_ca0132.c. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-08ALSA: hda - Fix superfluous "-in" suffix from CA0132 capture itemsTakashi Iwai1-2/+2
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-08ALSA: hda - Add codec->pcm_format_first flagTakashi Iwai2-24/+46
Introduced a new flag to set up the PCM stream format at first before the stream_id and channel tag. Some codecs (e.g. CA0132) seem preferring this over stream_id -> format order. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-08ASoC: imx-ssi: Remove mono supportFabio Estevam1-2/+3
Playing a mono track results in incorrect playback rate, ie, the audio is played at a faster rate. Remove mono support in the driver by setting 'channes_min' to dual-channel and this allows mono tracks to be played correctly. Reported-by: Gaëtan Carlier <gcembed@gmail.com> Tested-by: Gaëtan Carlier <gcembed@gmail.com> Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-08ASoC: mxs: Fix the name of the SoC familyFabio Estevam1-1/+1
SND_SOC_MXS_SGTL5000 is used on MXS boards, so fix the SoC family name. Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-08ALSA: hda - Fix double quirk for Quanta FL1 / Lenovo IdeapadDavid Henningsson1-2/+3
The same ID is twice in the quirk table, so the second one is not used. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Cc: <stable@vger.kernel.org> [v3.2+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-10 09:31:54 +0000