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* Merge branch 'topic/core-vuln-fixes' into for-linusTakashi Iwai2014-06-182-27/+52
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| * ALSA: control: Make sure that id->index does not overflowLars-Peter Clausen2014-06-181-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | The ALSA control code expects that the range of assigned indices to a control is continuous and does not overflow. Currently there are no checks to enforce this. If a control with a overflowing index range is created that control becomes effectively inaccessible and unremovable since snd_ctl_find_id() will not be able to find it. This patch adds a check that makes sure that controls with a overflowing index range can not be created. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: control: Handle numid overflowLars-Peter Clausen2014-06-181-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Each control gets automatically assigned its numids when the control is created. The allocation is done by incrementing the numid by the amount of allocated numids per allocation. This means that excessive creation and destruction of controls (e.g. via SNDRV_CTL_IOCTL_ELEM_ADD/REMOVE) can cause the id to eventually overflow. Currently when this happens for the control that caused the overflow kctl->id.numid + kctl->count will also over flow causing it to be smaller than kctl->id.numid. Most of the code assumes that this is something that can not happen, so we need to make sure that it won't happen Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: control: Don't access controls outside of protected regionsLars-Peter Clausen2014-06-181-5/+10
| | | | | | | | | | | | | | | | | | | | | | A control that is visible on the card->controls list can be freed at any time. This means we must not access any of its memory while not holding the controls_rw_lock. Otherwise we risk a use after free access. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: control: Fix replacing user controlsLars-Peter Clausen2014-06-181-16/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There are two issues with the current implementation for replacing user controls. The first is that the code does not check if the control is actually a user control and neither does it check if the control is owned by the process that tries to remove it. That allows userspace applications to remove arbitrary controls, which can cause a user after free if a for example a driver does not expect a control to be removed from under its feed. The second issue is that on one hand when a control is replaced the user_ctl_count limit is not checked and on the other hand the user_ctl_count is increased (even though the number of user controls does not change). This allows userspace, once the user_ctl_count limit as been reached, to repeatedly replace a control until user_ctl_count overflows. Once that happens new controls can be added effectively bypassing the user_ctl_count limit. Both issues can be fixed by instead of open-coding the removal of the control that is to be replaced to use snd_ctl_remove_user_ctl(). This function does proper permission checks as well as decrements user_ctl_count after the control has been removed. Note that by using snd_ctl_remove_user_ctl() the check which returns -EBUSY at beginning of the function if the control already exists is removed. This is not a problem though since the check is quite useless, because the lock that is protecting the control list is released between the check and before adding the new control to the list, which means that it is possible that a different control with the same settings is added to the list after the check. Luckily there is another check that is done while holding the lock in snd_ctl_add(), so we'll rely on that to make sure that the same control is not added twice. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: control: Protect user controls against concurrent accessLars-Peter Clausen2014-06-182-6/+26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The user-control put and get handlers as well as the tlv do not protect against concurrent access from multiple threads. Since the state of the control is not updated atomically it is possible that either two write operations or a write and a read operation race against each other. Both can lead to arbitrary memory disclosure. This patch introduces a new lock that protects user-controls from concurrent access. Since applications typically access controls sequentially than in parallel a single lock per card should be fine. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge tag 'asoc-v3.16-rc1' of ↵Takashi Iwai2014-06-1811-89/+135
|\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v3.16 Quite a few build coverage fixes in here among the usual small driver fixes includling the sigmadsp change from Lars - moving the driver to separate modules per bus (which is basically just code motion) avoids issues with some combinations of buses being enabled.
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| *-------. \ Merge remote-tracking branches 'asoc/fix/fsl-dma', 'asoc/fix/fsl-spdif', ↵Mark Brown2014-06-1610-76/+119
| |\ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | 'asoc/fix/pxa', 'asoc/fix/rcar' and 'asoc/fix/sigmadsp' into asoc-linus
| | | | | | * | ASoC: sigmadsp: Split regmap and I2C support into separate modulesLars-Peter Clausen2014-06-066-65/+107
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When the SigmaDSP module is built-in, but the I2C core is build as a module we'll get a undefined reference: sound/built-in.o: In function `sigma_action_write_i2c': :(.text+0x5d8d4): undefined reference to `i2c_master_send' This can happen if a audio driver that is using the regmap SigmaDSP interface is built into the kernel, but core I2C support is build as a module. To fix this split the SigmaDSP module into three modules, one module providing the core infrastructure and two small modules implementing the regmap and I2C interfaces. This allows e.g. the core infrastructure and regmap support to be built into the kernel while I2C support can still be build as a module. Fixes: dab464b60 ("ASoC: Add ADAU1361/ADAU1761 audio CODEC support") Reported-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | | | * | | ASoC: rsnd: fixup index of src/dst mod when captureKuninori Morimoto2014-06-121-1/+1
| | | | | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Index of dma name should use -1, not +1 when capture case. Thank you Dan. Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | | * | | ASoC: MMP audio needs sram supportArnd Bergmann2014-06-051-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | From e7a94bb7fb871c73cc85712d89c1f48d0271c1be Mon Sep 17 00:00:00 2001 From: Arnd Bergmann <arnd@arndb.de> Date: Thu, 5 Jun 2014 12:31:28 +0200 Subject: [PATCH] ASoC: MMP audio needs sram support Building the pxa/mmp audio driver without support for the mmp sram driver enabled results in this link error: sound/built-in.o: In function `mmp_pcm_free_dma_buffers': :(.text+0x3e734): undefined reference to `sram_get_gpool' sound/built-in.o: In function `mmp_pcm_new': :(.text+0x3e7c0): undefined reference to `sram_get_gpool' The sram driver is cannot be manually enabled and needs to be turned on by selecting MMP_SRAM from each module that needs it, which is what this patch does. Ideally, MMP should move over to the generic SRAM support, but for the moment, we can avoid the build error. Signed-off-by: Arnd Bergmann <arnd@arndb.de> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Haojian Zhuang <haojian.zhuang@gmail.com> Cc: Qiao Zhou <zhouqiao@marvell.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | | * | | ASoC: pxa: add I2C dependencies as neededArnd Bergmann2014-06-031-5/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We have in the past added 'depends on I2C' for some of the PXA boards after hitting randconfig build bugs. I have seens a couple of new bugs in this area during the linux-next cycle for 3.16, after it became possible to build some more PXA machines with I2C disabled. To shut this up for good, this adds the dependency to every board that uses I2C as the interface to the codec. I have gone through all board files and verified that they all either use AC97 or I2C, and this annotates the latter. Some of these already enable I2C from mach-pxa/Kconfig, but since that can change it's better to be explicit here. The link error that can result otherwise happens when CONFIG_I2C is set to 'm' and the codec driver is built-in as a result of being selected by the platform specific glue. Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | * | | | ASoC: fsl_spdif: Fix integer overflow when calculating divisorsAnssi Hannula2014-06-091-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The calculation code does u64 = (u32 - u32) * 100000; The 64 bits are of no help here as the type is casted only after the multiplication, and therefore the result may overflow, possibly causing inoptimal or wrong clock setup in an unfortunate case (the maximum result value of the first substraction is currently 47999). Fix the code to cast before multiplication. Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi> Acked-by: Nicolin Chen <Guangyu.Chen@freescale.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | * | | | ASoC: fsl_spdif: Fix incorrect usage of regmap_read()Nicolin Chen2014-06-091-1/+1
| | | | |/ / | | | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We should not copy the return value into this val since it's supposed to get the value of the register not the success result of regmap_read(). Thus fix it. Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | * | | | ASoC: fsl: Fix build problemGuenter Roeck2014-06-121-2/+2
| | |/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit 432481220 (ASoC: fsl-ssi: Use regmap) removed struct ccsr_ssi. Unfortunately, the structure is still used. This causes mpc85xx_smp_defconfig and mpc85xx_defconfig builds to fail with sound/soc/fsl/fsl_dma.c:926:50: error: invalid use of undefined type 'struct ccsr_ssi' dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0); ound/soc/fsl/fsl_dma.c:927:50: error: invalid use of undefined type 'struct ccsr_ssi' dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0); Fix by using constants, similar to original commit. Cc: Markus Pargmann <mpa@pengutronix.de> Signed-off-by: Guenter Roeck <linux@roeck-us.net> Signed-off-by: Mark Brown <broonie@linaro.org>
| * | | | Merge remote-tracking branch 'asoc/fix/core' into asoc-linusMark Brown2014-06-161-13/+16
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| | * | | ASoC: dapm: Make sure register value is in sync with DAPM kcontrol stateJarkko Nikula2014-06-091-13/+16
| | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit c9e065c27fe9 ("ASoC: dapm: Make sure to always update the DAPM graph in _put_volsw()") stopped updating register values in those cases where initial after boot state of kcontrol appears to not change but where register value still needs update because it is not in sync with the kcontrol state. Fix this by doing snd_soc_test_bits() unconditionally as it was before but by using separate flags for kcontrol and register state changes. This allow both DAPM graph to be updated when disabling auto-muted control and update register if it is out-of-sync in respect of kcontrol state. Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
* | | | drm/i915, HD-audio: Don't continue probing when nomodeset is givenTakashi Iwai2014-06-163-9/+14
|/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When a machine is booted with nomodeset option, i915 driver skips the whole initialization. Meanwhile, HD-audio tries to bind wth i915 just by request_symbol() without knowing that the initialization was skipped, and eventually it hits WARN_ON() in i915_request_power_well() and i915_release_power_well() wrongly but still continues probing, even though it doesn't work at all. In this patch, both functions are changed to return an error in case of uninitialized state instead of WARN_ON(), so that HD-audio driver can give up HDMI controller initialization at the right time. Acked-by: Daniel Vetter <daniel.vetter@ffwll.ch> Cc: <stable@vger.kernel.org> [3.15] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | Merge tag 'sound-fix-3.16-rc1' of ↵Linus Torvalds2014-06-1314-54/+145
|\ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "Most of changes are small and easy cleanup or fixes: - a few HD-audio Realtek codec fixes and quirks - Intel HDMI audio fixes for Broadwell and Haswell / ValleyView - FireWire sound stack cleanups - a couple of sequencer core fixes - compress ABI fix for 64bit - conversion to modern ktime*() API" * tag 'sound-fix-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (23 commits) ALSA: hda/realtek - Add more entry for enable HP mute led ALSA: hda - Add quirk for external mic on Lifebook U904 ALSA: hda - fix a fixup value for codec alc293 in the pin_quirk table ALSA: intel8x0: Use ktime and ktime_get() ALSA: core: Use ktime_get_ts() ALSA: hda - verify pin:converter connection on unsol event for HSW and VLV ALSA: compress: Cancel the optimization of compiler and fix the size of struct for all platform. ALSA: hda - Add quirk for ABit AA8XE Revert "ALSA: hda - mask buggy stream DMA0 for Broadwell display controller" ALSA: hda - using POS_FIX_LPIB on Broadwell HDMI Audio ALSA: hda/realtek - Add support of ALC667 codec ALSA: hda/realtek - Add more codec rename ALSA: hda/realtek - New vendor ID for ALC233 ALSA: hda - add two new pin tables ALSA: hda/realtek - Add support of ALC891 codec ALSA: seq: Continue broadcasting events to ports if one of them fails ALSA: bebob: Remove unused function prototype ALSA: fireworks: Remove meaningless mutex_destroy() ALSA: fireworks: Remove a constant over width to which it's applied ALSA: fireworks: Improve comments about Fireworks transaction ...
| * | | ALSA: hda/realtek - Add more entry for enable HP mute ledKailang Yang2014-06-131-0/+14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | More HP machine need mute led support. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda - Add quirk for external mic on Lifebook U904David Henningsson2014-06-131-0/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | According to the bug reporter (Данило Шеган), the external mic starts to work and has proper jack detection if only pin 0x19 is marked properly as an external headset mic. AlsaInfo at https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1328587/+attachment/4128991/+files/AlsaInfo.txt Cc: stable@vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1328587 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda - fix a fixup value for codec alc293 in the pin_quirk tableHui Wang2014-06-131-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The fixup value for codec alc293 was set to ALC269_FIXUP_DELL1_MIC_NO_PRESENCE by a mistake, if we don't fix it, the Dock mic will be overwriten by the headset mic, this will make the Dock mic can't work. Cc: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: intel8x0: Use ktime and ktime_get()Thomas Gleixner2014-06-121-6/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | do_posix_clock_monotonic_gettime() is a leftover from the initial posix timer implementation which maps to ktime_get_ts() and returns the monotonic time in a timespec. Use ktime based ktime_get() and use the ktime_delta_us() function to calculate the delta instead of open coding the timespec math. Signed-off-by: Thomas Gleixner <tglx@linutronix.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: core: Use ktime_get_ts()Thomas Gleixner2014-06-121-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | do_posix_clock_monotonic_gettime() is a leftover from the initial posix timer implementation which maps to ktime_get_ts(). Signed-off-by: Thomas Gleixner <tglx@linutronix.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda - verify pin:converter connection on unsol event for HSW and VLVMengdong Lin2014-06-121-1/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch will verify the pin's coverter selection for an active stream when an unsol event reports this pin becomes available again after a display mode change or hot-plug event. For Haswell+ and Valleyview: display mode change or hot-plug can change the transcoder:port connection and make all the involved audio pins share the 1st converter. So the stream using 1st convertor will flow to multiple pins but active streams using other converters will fail. This workaround is to assure the pin selects the right conveter and an assigned converter is not shared by other unused pins. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda - Add quirk for ABit AA8XEDavid Henningsson2014-06-101-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Bios does not set up the pin config default correctly (everything is set to zero). Reporter claims that 6stack-dig and 6stack-automute solve the problem. Alsa-info at http://www.alsa-project.org/db/?f=376c0804cbdde90bcd2cb94799407cb1cacf5d05 BugLink: https://bugs.launchpad.net/bugs/1319291 Reported-by: Stefano Statuti <stefano.statuti@hotmail.it> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | Revert "ALSA: hda - mask buggy stream DMA0 for Broadwell display controller"Libin Yang2014-06-091-6/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This reverts commit 7189eb9b8f7962474956196c301676470542f253. It will use LPIB to get the DMA position on Broadwell HDMI Audio. Signed-off-by: Libin Yang <libin.yang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda - using POS_FIX_LPIB on Broadwell HDMI AudioLibin Yang2014-06-091-1/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Broadwell HDMI can't use position buffer reliably, force to use LPIB Signed-off-by: Libin Yang <libin.yang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda/realtek - Add support of ALC667 codecKailang Yang2014-06-061-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | New codec suooprt of ALC667. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda/realtek - Add more codec renameKailang Yang2014-06-061-0/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Some vendor has special bonding options. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda/realtek - New vendor ID for ALC233Kailang Yang2014-06-061-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is compatible with ALC255. It is use for Lenovo. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda - add two new pin tablesHui Wang2014-06-061-6/+41
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | These two new pin tables can fix headset mic problems for several new Dell machines. And also delete some machines from old quirk table since the existing pin talbes already cover them. Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda/realtek - Add support of ALC891 codecKailang Yang2014-06-051-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | New codec support for ALC891. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: seq: Continue broadcasting events to ports if one of them failsAdam Goode2014-06-041-12/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Sometimes PORT_EXIT messages are lost when a process is exiting. This happens if you subscribe to the announce port with client A, then subscribe to the announce port with client B, then kill client A. Client B will not see the PORT_EXIT message because client A's port is closing and is earlier in the announce port subscription list. The for each loop will try to send the announcement to client A and fail, then will stop trying to broadcast to other ports. Killing B works fine since the announcement will already have gone to A. The CLIENT_EXIT message does not get lost. How to reproduce problem: *** termA $ aseqdump -p 0:1 0:1 Port subscribed 0:1 -> 128:0 *** termB $ aseqdump -p 0:1 *** termA 0:1 Client start client 129 0:1 Port start 129:0 0:1 Port subscribed 0:1 -> 129:0 *** termB 0:1 Port subscribed 0:1 -> 129:0 *** termA ^C *** termB 0:1 Client exit client 128 <--- expected Port exit as well (before client exit) Signed-off-by: Adam Goode <agoode@google.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: bebob: Remove unused function prototypeTakashi Sakamoto2014-06-041-2/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_bebob_stream_map() is not defined. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: fireworks: Remove meaningless mutex_destroy()Takashi Sakamoto2014-06-041-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently mutex_destroy() is called in module's cleanup function. But after cleaned up, this mutex is automatically released. So this function call is meaningless. [fixed a typo in changelog by tiwai] Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: fireworks: Remove a constant over width to which it's appliedTakashi Sakamoto2014-06-041-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The constants of enum snd_efw_grp_type is for struct snd_efw_phys_grp.type. But this member is 1 byte. Although the value is between 0x00-0xff, a constant has 0x10000. This constant is meaningless. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: fireworks: Improve comments about Fireworks transactionTakashi Sakamoto2014-06-041-8/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | It includes descriptions to cause misreading. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: fireworks: Use safer way to arrange ring buffer pointerTakashi Sakamoto2014-06-042-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | To reverse a pointer for the ring buffer, subtraction by buffer size is better than assignment to the beginning of the buffer. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: fireworks/bebob: Shorten critical section for stream_stop_duplex()Takashi Sakamoto2014-06-042-4/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | All assignment for local variables in these functions are not related to critical section. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: seq: correctly detect input buffer overflowAdam Goode2014-06-041-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_seq_event_dup returns -ENOMEM in some buffer-full conditions, but usually returns -EAGAIN. Make -EAGAIN trigger the overflow condition in snd_seq_fifo_event_in so that the fifo is cleared and -ENOSPC is returned to userspace as stated in the alsa-lib docs. Signed-off-by: Adam Goode <agoode@google.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | Merge tag 'sound-3.16-rc1' of ↵Linus Torvalds2014-06-04300-5237/+35826
|\ \ \ \ | |/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into next Pull sound updates from Takashi Iwai: "At this time, majority of changes come from ASoC world while we got a few new drivers in other places for FireWire and USB. There have been lots of ASoC core cleanups / refactoring, but very little visible to external users. ASoC: - Support for specifying aux CODECs in DT - Removal of the deprecated mux and enum macros - More moves towards full componentisation - Removal of some unused I/O code - Lots of cleanups, fixes and enhancements to the davinci, Freescale, Haswell and Realtek drivers - Several drivers exposed directly in Kconfig for use with simple-card - GPIO descriptor support for jacks - More updates and fixes to the Freescale SSI, Intel and rsnd drivers - New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and ST STA350, Analog Devices ADAU1361, ADAU1381, ADAU1761 and ADAU1781, and Realtek RT5677 HD-audio: - Clean up Dell headset quirks - Noise fixes for Dell and Sony laptops - Thinkpad T440 dock fix - Realtek codec updates (ALC293,ALC233,ALC3235) - Tegra HD-audio HDMI support FireWire-audio: - FireWire audio stack enhancement (AMDTP, MIDI), support for incoming isochronous stream and duplex streams with timestamp synchronization - BeBoB-based devices support - Fireworks-based device support USB-audio: - Behringer BCD2000 USB device support Misc: - Clean up of a few old drivers, atmel, fm801, etc" * tag 'sound-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (480 commits) ASoC: Fix wrong argument for card remove callbacks ASoC: free jack GPIOs before the sound card is freed ALSA: firewire-lib: Remove a comment about restriction of asynchronous operation ASoC: cache: Fix error code when not using ASoC level cache ALSA: hda/realtek - Fix COEF widget NID for ALC260 replacer fixup ALSA: hda/realtek - Correction of fixup codes for PB V7900 laptop ALSA: firewire-lib: Use IEC 61883-6 compliant labels for Raw Audio data ASoC: add RT5677 CODEC driver ASoC: intel: The Baytrail/MAX98090 driver depends on I2C ASoC: rt5640: Add the function "get_clk_info" to RL6231 shared support ASoC: rt5640: Add the function of the PLL clock calculation to RL6231 shared support ASoC: rt5640: Add RL6231 class device shared support for RT5640, RT5645 and RT5651 ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error. ASoC: Add helper functions to cast from DAPM context to CODEC/platform ALSA: bebob: sizeof() vs ARRAY_SIZE() typo ASoC: wm9713: correct mono out PGA sources ALSA: synth: emux: soundfont.c: Cleaning up memory leak ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320 ASoC: fsl-ssi: Use regmap ASoC: fsl-ssi: reorder and document fsl_ssi_private ...
| * | | ASoC: Fix wrong argument for card remove callbacksTakashi Iwai2014-06-037-9/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The commit [e1d4d3c8: ASoC: free jack GPIOs before the sound card is freed] introduced snd_soc_card remove callbacks to a few drivers, but they are implemented with a wrong argument type. The callback should receive snd_soc_card pointer instead of snd_soc_pcm_runtime. Fixes: e1d4d3c854f2 ('ASoC: free jack GPIOs before the sound card is freed') Acked-by: Mark Brown <broonie@linaro.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | Merge tag 'asoc-v3.16-2' of ↵Takashi Iwai2014-06-0387-932/+9417
| |\ \ \ | | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: Final updates for v3.16 A few more updates from the last week of development, nothing too exciting. Highlights include: - GPIO descriptor support for jacks - More updates and fixes to the Freescale SSI, Intel and rsnd drivers. - New drivers for Analog Devices ADAU1361, ADAU1381, ADAU1761 and ADAU1781, and Realtek RT5677.
| | * | ASoC: free jack GPIOs before the sound card is freedStephen Warren2014-06-037-34/+69
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is the same change as commit fb6b8e71448a "ASoC: tegra: free jack GPIOs before the sound card is freed", but applied to all other ASoC machine drivers where code inspection indicates the same problem exists. That commit's description is: ========== snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to generate an initial jack status report. If sound card initialization fails, that work item needs to be cancelled, so it doesn't run after the card has been freed. Specifically, freeing the card calls snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which is called from the work queue item. snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine drivers do call this function in the platform driver remove() callback. However, this happens after the sound card is freed, at least when the card is freed due to errors late during snd_soc_instantiate_card(). This leaves a window where the work item can execute after the card is freed. In next-20140522, sound card initialization does fail for unrelated reasons, and hits the problem described above. To solve this, fix the Tegra ASoC machine drivers to clean up the Jack GPIOs during the snd_soc_card's .remove() callback, which is executed before the overall card object is freed. also, guard the cleanup call based on whether we actually setup up the GPIOs in the first place. Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove function to match where the GPIOs get set up. However, there is no such callback. ========== Note that I have not even compile-tested this in most cases, since most of the drivers rely on specific mach-* support I don't have enabled, and don't support COMPILE_TEST. Testing by the relevant board maintainers would be useful. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | |
| | | \
| | *-. \ Merge remote-tracking branches 'asoc/topic/wm8804' and 'asoc/topic/wm9713' ↵Mark Brown2014-06-033-5/+19
| | |\ \ \ | | | | | | | | | | | | | | | | | | into asoc-next
| | | | * | ASoC: wm9713: correct mono out PGA sourcesMatt Reimer2014-06-011-2/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The mono output PGA input only has four possible sources, so omit the rest. Signed-off-by: Matt Reimer <mreimer@sdgsystems.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | * | | ASoC: wm8804: Allow control of master clock divider in PLL generationDaniel Matuschek2014-05-292-3/+18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using an additional mclk_div divider, it is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. The behavior has to be controlled by the sound card driver, because some sample frequency share the same setting. e.g. 192kHz and 96kHz use 24.576MHz master clock. The only difference is the MCLK divider. Signed-off-by: Daniel Matuschek <daniel@matuschek.net> Tested-by: Florian Meier <florian.meier@koalo.de> Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | * | | | Merge remote-tracking branch 'asoc/topic/tegra' into asoc-nextMark Brown2014-06-034-14/+45
| | |\ \ \ \
| | | * | | | ASoC: tegra: free jack GPIOs before the sound card is freedStephen Warren2014-05-264-14/+45
| | | | |/ / | | | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to generate an initial jack status report. If sound card initialization fails, that work item needs to be cancelled, so it doesn't run after the card has been freed. Specifically, freeing the card calls snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which is called from the work queue item. snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine drivers do call this function in the platform driver remove() callback. However, this happens after the sound card is freed, at least when the card is freed due to errors late during snd_soc_instantiate_card(). This leaves a window where the work item can execute after the card is freed. In next-20140522, sound card initialization does fail for unrelated reasons, and hits the problem described above. To solve this, fix the Tegra ASoC machine drivers to clean up the Jack GPIOs during the snd_soc_card's .remove() callback, which is executed before the overall card object is freed. also, gGuard the cleanup call based on whether we actually setup up the GPIOs in the first place. Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove function to match where the GPIOs get set up. However, there is no such callback. This change fixes all Tegra machine drivers. By code inspection, I believe some non-Tegra machine drivers have the same issue. I'll send a patch for that separately, once this is reviewed. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@linaro.org>
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