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* ALSA: dice: fix error path to destroy initialized stream dataTakashi Sakamoto2018-04-261-1/+1
| | | | | | | | | | | | In error path of snd_dice_stream_init_duplex(), stream data for incoming packet can be left to be initialized. This commit fixes it. Fixes: 436b5abe2224 ('ALSA: dice: handle whole available isochronous streams') Cc: <stable@vger.kernel.org> # v4.6+ Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Skip jack and others for non-existing PCM streamsTakashi Iwai2018-04-251-1/+8
| | | | | | | | | | | | | | | | | | | | | | | | When CONFIG_SND_DYNAMIC_MINORS isn't set, there are only limited number of devices available, and HD-audio, especially with HDMI/DP codec, will fail to create more than two devices. The driver warns about the lack of such devices and skips the PCM device creations, but the HDMI driver still tries to create the corresponding JACK, SPDIF and ELD controls even for the non-existing PCM substreams. This results in confusion on user-space, and even may break the operation. Similarly, Intel HDMI/DP codec builds the ELD notification from i915 graphics driver, and this may be broken if a notification is sent for the non-existing PCM stream. This patch adds the check of the existence of the assigned PCM substream in the both scenarios above, and skips the further operation if the PCM substream is not assigned. Fixes: 9152085defb6 ("ALSA: hda - add DP MST audio support") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge tag 'asoc-fix-4.17-rc2' of ↵Takashi Iwai2018-04-2511-33/+85
|\ | | | | | | | | | | | | | | | | | | https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v4.17 A small batch of fixes collected since the merge window, none of which are particularly large or remarkable. They've all been cooking in -next for a while.
| * ASoC: msm8916-wcd-analog: use threaded context for mbhc eventsSrinivas Kandagatla2018-04-191-3/+6
| | | | | | | | | | | | | | | | | | | | As snd_soc_jack_report() can sleep, move handling of mbhc events to a thread context rather than in interrupt context. Fixes: de66b3455023 ('ASoC: codecs: msm8916-wcd-analog: add MBHC support') Reported-by: Bjorn Andersson <bjorn.andersson@linaro.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: topology: Check widget kcontrols before derefLiam Girdwood2018-04-171-2/+2
| | | | | | | | | | | | | | Validate the topology input before we dereference the pointer. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: topology: Fix bugs of freeing soc topologyYan Wang2018-04-171-1/+1
| | | | | | | | | | | | | | | | | | | | In snd_soc_tplg_component_remove(), it should compare index and not dobj->index with SND_SOC_TPLG_INDEX_ALL for removing all topology objects. Signed-off-by: Yan Wang <yan.wang@linux.intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: amd: acp-da7219-max98357: Make symbol da7219_dai_clk staticWei Yongjun2018-04-161-1/+1
| | | | | | | | | | | | | | | | | | | | Fixes the following sparse warning: sound/soc/amd/acp-da7219-max98357a.c:46:12: warning: symbol 'da7219_dai_clk' was not declared. Should it be static? Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: rt5514: Add the missing register in the readable tableoder_chiou@realtek.com2018-04-161-0/+3
| | | | | | | | | | | | | | The patch adds the missing register in the readable table. Signed-off-by: Oder Chiou <oder_chiou@realtek.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: adau17x1: Handling of DSP_RUN register during fw setupDanny Smith2018-04-132-7/+22
| | | | | | | | | | | | | | | | | | | | | | DSP_RUN needs to be disabled during firmware write otherwise we can end up with undefined behavior if writing to a dsp which is already running firmware. Signed-off-by: Danny Smith <dannys@axis.com> Signed-off-by: Robert Rosengren <robert.rosengren@axis.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: topology: fix some tiny memory leaksDan Carpenter2018-04-121-2/+6
| | | | | | | | | | | | | | | | These tiny memory leaks don't have a huge real life impact but they cause static checker warnings so let's fix them. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: Intel: atom: fix ACPI/PCI KconfigPierre-Louis Bossart2018-04-121-9/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The split between ACPI and PCI platforms generated issues with randconfig: with SND_SST_ATOM_HIFI2_PLATFORM_PCI=y and SND_SST_ATOM_HIFI2_PLATFORM=m, we get this module link failure: ERROR: "sst_context_init" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_context_cleanup" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_alloc_drv_context" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "intel_sst_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_configure_runtime_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! To keep things simple, let's expose two configs for SND_SST_ATOM_HIFI2_PLATFORM_PCI and SND_SST_ATOM_HIFI2_PLATFORM_ACPI, which select a common SND_SST_ATOM_HIFI2_PLATFORM option. To avoid breaking existing solutions with the semantics change, SND_SST_ATOM_HIFI2_PLATFORM_ACPI uses "default ACPI" so that "make oldnoconfig" and "make olddefconfig" still work as expected. Also remove mentions of Medfield while we are at it since it was removed recently. Reported-by: Arnd Bergmann <arnd@arndb.de> Fixes: 4772c16ede52 ("ASoC: Intel: Kconfig: Simplify-clarify ACPI/PCI dependencies") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com> Acked-By: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: rsnd: mark PM functions __maybe_unusedArnd Bergmann2018-04-121-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The suspend/resume callbacks are now optional, leading to a warning when they are unused: sound/soc/sh/rcar/core.c:1548:12: error: 'rsnd_resume' defined but not used [-Werror=unused-function] static int rsnd_resume(struct device *dev) ^~~~~~~~~~~ sound/soc/sh/rcar/core.c:1539:12: error: 'rsnd_suspend' defined but not used [-Werror=unused-function] static int rsnd_suspend(struct device *dev) This marks the as __maybe_unused to avoid the warning. Fixes: f8a9a29c4fe9 ("ASoC: rsnd: set pm_ops in hibernate-compatible way") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: fsl_ssi: Fix mode setting when changing channel numberNicolin Chen2018-04-121-3/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is a partial revert (in a cleaner way) of commit ebf08ae3bc90 ("ASoC: fsl_ssi: Keep ssi->i2s_net updated") to fix a regression at test cases when switching between mono and stereo audio. The problem is that ssi->i2s_net is initialized in set_dai_fmt() only, while this set_dai_fmt() is only called during the dai-link probe(). The original patch assumed set_dai_fmt() would be called during every playback instance, so it failed at the overriding use cases. This patch adds the local variable i2s_net back to let regular use cases still follow the mode settings from the set_dai_fmt(). Meanwhile, the original commit of keeping ssi->i2s_net updated was to make set_tdm_slot() clean by checking the ssi->i2s_net directly instead of reading SCR register. However, the change itself is not necessary (or even harmful) because the set_tdm_slot() might fail to check the slot number for Normal-Mode-None-Net settings while mono audio cases still need 2 slots. So this patch can also fix it. And it adds an extra line of comments to declare ssi->i2s_net does not reflect the register value but merely the initial setting from the set_dai_fmt(). Reported-by: Mika Penttilä <mika.penttila@nextfour.com> Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com> Tested-by: Mika Penttilä <mika.penttila@nextfour.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: fsl_esai: Fix divisor calculation failure at lower ratioNicolin Chen2018-04-121-0/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When the desired ratio is less than 256, the savesub (tolerance) in the calculation would become 0. This will then fail the loop- search immediately without reporting any errors. But if the ratio is smaller enough, there is no need to calculate the tolerance because PM divisor alone is enough to get the ratio. So a simple fix could be just to set PM directly instead of going into the loop-search. Reported-by: Marek Vasut <marex@denx.de> Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com> Tested-by: Marek Vasut <marex@denx.de> Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org
| * ASoC: dmic: Fix clock parentingTero Kristo2018-04-121-3/+11
| | | | | | | | | | | | | | | | | | | | | | | | In 4.16 the clock hierarchy got changed by a5c82a09d876 ARM: dts: omap4: add clkctrl nodes The fck of dmic is no longer a mux clock, it's parent is. Signed-off-by: Tero Kristo <t-kristo@ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org # 4.16+
* | ALSA: hda/realtek - change the location for one of two front micsKailang Yang2018-04-251-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | On this Lenovo ThinkCentre machine. There are two front mics, we change the location for one of them. Relation: f33f79f3d0e5 ("ALSA: hda/realtek - change the location for one of two front microphones") Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: rme9652: Hardening for potential Spectre v1Takashi Iwai2018-04-251-2/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | As recently Smatch suggested, one place in RME9652 driver may expand the array directly from the user-space value with speculation: sound/pci/rme9652/rme9652.c:2074 snd_rme9652_channel_info() warn: potential spectre issue 'rme9652->channel_map' (local cap) This patch puts array_index_nospec() for hardening against it. BugLink: https://marc.info/?l=linux-kernel&m=152411496503418&w=2 Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hdspm: Hardening for potential Spectre v1Takashi Iwai2018-04-251-10/+14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As recently Smatch suggested, a couple of places in HDSP MADI driver may expand the array directly from the user-space value with speculation: sound/pci/rme9652/hdspm.c:5717 snd_hdspm_channel_info() warn: potential spectre issue 'hdspm->channel_map_out' (local cap) sound/pci/rme9652/hdspm.c:5734 snd_hdspm_channel_info() warn: potential spectre issue 'hdspm->channel_map_in' (local cap) This patch puts array_index_nospec() for hardening against them. BugLink: https://marc.info/?l=linux-kernel&m=152411496503418&w=2 Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: asihpi: Hardening for potential Spectre v1Takashi Iwai2018-04-252-5/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | As recently Smatch suggested, a couple of places in ASIHPI driver may expand the array directly from the user-space value with speculation: sound/pci/asihpi/hpimsginit.c:70 hpi_init_response() warn: potential spectre issue 'res_size' (local cap) sound/pci/asihpi/hpioctl.c:189 asihpi_hpi_ioctl() warn: potential spectre issue 'adapters' This patch puts array_index_nospec() for hardening against them. BugLink: https://marc.info/?l=linux-kernel&m=152411496503418&w=2 Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: opl3: Hardening for potential Spectre v1Takashi Iwai2018-04-251-2/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | As recently Smatch suggested, one place in OPL3 driver may expand the array directly from the user-space value with speculation: sound/drivers/opl3/opl3_synth.c:476 snd_opl3_set_voice() warn: potential spectre issue 'snd_opl3_regmap' This patch puts array_index_nospec() for hardening against it. BugLink: https://marc.info/?l=linux-kernel&m=152411496503418&w=2 Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda: Hardening for potential Spectre v1Takashi Iwai2018-04-251-1/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As recently Smatch suggested, one place in HD-audio hwdep ioctl codes may expand the array directly from the user-space value with speculation: sound/pci/hda/hda_local.h:467 get_wcaps() warn: potential spectre issue 'codec->wcaps' As get_wcaps() itself is a fairly frequently called inline function, and there is only one single call with a user-space value, we replace only the latter one to open-code locally with array_index_nospec() hardening in this patch. BugLink: https://marc.info/?l=linux-kernel&m=152411496503418&w=2 Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: seq: oss: Hardening for potential Spectre v1Takashi Iwai2018-04-254-40/+55
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As Smatch recently suggested, a few places in OSS sequencer codes may expand the array directly from the user-space value with speculation, namely there are a significant amount of references to either info->ch[] or dp->synths[] array: sound/core/seq/oss/seq_oss_event.c:315 note_on_event() warn: potential spectre issue 'info->ch' (local cap) sound/core/seq/oss/seq_oss_event.c:362 note_off_event() warn: potential spectre issue 'info->ch' (local cap) sound/core/seq/oss/seq_oss_synth.c:470 snd_seq_oss_synth_load_patch() warn: potential spectre issue 'dp->synths' (local cap) sound/core/seq/oss/seq_oss_event.c:293 note_on_event() warn: potential spectre issue 'dp->synths' sound/core/seq/oss/seq_oss_event.c:353 note_off_event() warn: potential spectre issue 'dp->synths' sound/core/seq/oss/seq_oss_synth.c:506 snd_seq_oss_synth_sysex() warn: potential spectre issue 'dp->synths' sound/core/seq/oss/seq_oss_synth.c:580 snd_seq_oss_synth_ioctl() warn: potential spectre issue 'dp->synths' Although all these seem doing only the first load without further reference, we may want to stay in a safer side, so hardening with array_index_nospec() would still make sense. We may put array_index_nospec() at each place, but here we take a different approach: - For dp->synths[], change the helpers to retrieve seq_oss_synthinfo pointer directly instead of the array expansion at each place - For info->ch[], harden in a normal way, as there are only a couple of places As a result, the existing helper, snd_seq_oss_synth_is_valid() is replaced with snd_seq_oss_synth_info(). Also, we cover MIDI device where a similar array expansion is done, too, although it wasn't reported by Smatch. BugLink: https://marc.info/?l=linux-kernel&m=152411496503418&w=2 Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: seq: oss: Fix unbalanced use lock for synth MIDI deviceTakashi Iwai2018-04-251-4/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | When get_synthdev() is called for a MIDI device, it returns the fixed midi_synth_dev without the use refcounting. OTOH, the caller is supposed to unreference unconditionally after the usage, so this would lead to unbalanced refcount. This patch corrects the behavior and keep up the refcount balance also for the MIDI synth device. Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/realtek - Update ALC255 depop optimizeKailang Yang2018-04-251-0/+2
| | | | | | | | | | | | | | | | | | Add ALC255 its own depop functions for alc_init and alc_shutup. Assign it to ALC256 usage. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/realtek - Add some fixes for ALC233Kailang Yang2018-04-251-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fill COEF to change EAPD to verb control. Assigned codec type. This is an additional fix over 92f974df3460 ("ALSA: hda/realtek - New vendor ID for ALC233"). [ More notes: according to Kailang, the chip is 10ec:0235 bonding for ALC233b, which is equivalent with ALC255. It's only used for Lenovo. The chip needs no alc_process_coef_fw() for headset unlike ALC255. ] Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: pcm: Change return type to vm_fault_tSouptick Joarder2018-04-251-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | Use new return type vm_fault_t for fault handler. For now, this is just documenting that the function returns a VM_FAULT value rather than an errno. Once all instances are converted, vm_fault_t will become a distinct type. Commit 1c8f422059ae ("mm: change return type to vm_fault_t") Signed-off-by: Souptick Joarder <jrdr.linux@gmail.com> Reviewed-by: Matthew Wilcox <mawilcox@microsoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usx2y: Change return type to vm_fault_tSouptick Joarder2018-04-253-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | Use new return type vm_fault_t for fault handler. For now, this is just documenting that the function returns a VM_FAULT value rather than an errno. Once all instances are converted, vm_fault_t will become a distinct type. Commit 1c8f422059ae ("mm: change return type to vm_fault_t") Signed-off-by: Souptick Joarder <jrdr.linux@gmail.com> Reviewed-by: Matthew Wilcox <mawilcox@microsoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: ADC3: Fix channel mapping conversion for ADC3.Michael Drake2018-04-241-1/+1
| | | | | | | | | | | | | | | | | | | | The channel mapping is defined by bChRelationship, not bChPurpose. Fixes: 9a2fe9b801f5 ("ALSA: usb: initial USB Audio Device Class 3.0 support") Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Signed-off-by: Michael Drake <michael.drake@codethink.co.uk> Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: dice: fix OUI for TC groupTakashi Sakamoto2018-04-241-1/+1
| | | | | | | | | | | | | | | | | | | | | | OUI for TC Electronic is 0x000166, for TC GROUP A/S. 0x001486 is for Echo Digital Audio Corporation. Fixes: 7cafc65b3aa1 ('ALSA: dice: force to add two pcm devices for listed models') Cc: <stable@vger.kernel.org> # v4.6+ Reference: http://standards-oui.ieee.org/oui/oui.txt Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Skip broken EU on Dell dock USB-audioTakashi Iwai2018-04-241-0/+3
| | | | | | | | | | | | | | | | | | | | | | The Dell Dock USB-audio device with 0bda:4014 is behaving notoriously bad, and we have already applied some workaround to avoid the firmware hiccup. Yet we still need to skip one thing, the Extension Unit at ID 4, which doesn't react correctly to the mixer ctl access. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1090658 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Fix missing endian conversionTakashi Iwai2018-04-241-2/+2
| | | | | | | | | | | | | | | | | | | | | | The UAC2 jack detection support introduced the bmControls checks in a couple of places, but they forgot the endian conversion; the bmControls of UAC2 terminal descriptor is __le16, not a byte like in UAC1. Fixes: 5a222e849452 ("ALSA: usb-audio: UAC2 jack detection") Tested-by: Andrew Chant <achant@google.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Fix forgotten conversion of control query functionsTakashi Iwai2018-04-231-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The recent code refactoring made the argument for some helper functions to be the explicit UAC_CS_* and UAC2_CS_* value instead of 0-based offset. However, there was one place left forgotten, and it caused a regression on some devices appearing as the inconsistent mixer setup. This patch corrects the forgotten conversion. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=199449 Fixes: 21e9b3e931f7 ("ALSA: usb-audio: fix uac control query argument") Tested-by: Nazar Mokrynskyi <nazar@mokrynskyi.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: control: Fix missing __user annotationTakashi Iwai2018-04-231-1/+1
| | | | | | | | | | | | | | | | | | There is one place missing __user annotation to the pointer used by the recent code refactoring. Reported by sparse. Fixes: 450296f305f1 ("ALSA: control: code refactoring TLV ioctl handler") Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: core: Report audio_tstamp in snd_pcm_sync_ptrDavid Henningsson2018-04-231-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | It looks like a simple mistake that this struct member was forgotten. Audio_tstamp isn't used much, and on some archs (such as x86) this ioctl is not used by default, so that might be the reason why this has slipped for so long. Fixes: 4eeaaeaea1ce ("ALSA: core: add hooks for audio timestamps") Signed-off-by: David Henningsson <diwic@ubuntu.com> Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Cc: <stable@vger.kernel.org> # v3.8+ Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: pcm: Return negative delays from SNDRV_PCM_IOCTL_DELAY.Jeffery Miller2018-04-232-15/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The commit c2c86a97175f ("ALSA: pcm: Remove set_fs() in PCM core code") changed SNDRV_PCM_IOCTL_DELAY to return an inconsistent error instead of a negative delay. Originally the call would succeed and return the negative delay. The Chromium OS Audio Server (CRAS) gets confused and hangs when the error is returned instead of the negative delay. Help CRAS avoid the issue by rolling back the behavior to return a negative delay instead of an error. Fixes: c2c86a97175f ("ALSA: pcm: Remove set_fs() in PCM core code") Signed-off-by: Jeffery Miller <jmiller@neverware.com> Cc: <stable@vger.kernel.org> # v4.13+ Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge tag 'sound-4.17-rc2' of ↵Linus Torvalds2018-04-214-8/+18
|\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "A few small fixes: - a fix for the NULL-dereference in rawmidi compat ioctls, triggered by fuzzer - HD-audio Realtek codec quirks, a VIA controller fixup - a long-standing bug fix in LINE6 MIDI" * tag 'sound-4.17-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: rawmidi: Fix missing input substream checks in compat ioctls ALSA: hda/realtek - adjust the location of one mic ALSA: hda/realtek - set PINCFG_HEADSET_MIC to parse_flags ALSA: hda - New VIA controller suppor no-snoop path ALSA: line6: Use correct endpoint type for midi output
| * | ALSA: rawmidi: Fix missing input substream checks in compat ioctlsTakashi Iwai2018-04-191-6/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some rawmidi compat ioctls lack of the input substream checks (although they do check only for rfile->output). This many eventually lead to an Oops as NULL substream is passed to the rawmidi core functions. Fix it by adding the proper checks before each function call. The bug was spotted by syzkaller. Reported-by: syzbot+f7a0348affc3b67bc617@syzkaller.appspotmail.com Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda/realtek - adjust the location of one micHui Wang2018-04-191-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There are two front mics on this machine, if we don't adjust the location for one of them, they will have the same mixer name, pulseaudio can't handle this situation. After applying this FIXUP, they will have different mixer name, then pulseaudio can handle them correctly. Cc: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda/realtek - set PINCFG_HEADSET_MIC to parse_flagsHui Wang2018-04-191-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Otherwise, the pin will be regarded as microphone, and the jack name is "Mic Phantom", it is always on in the pulseaudio even nothing is plugged into the jack. So the UI is confusing to users since the microphone always shows up in the UI even there is no microphone plugged. After adding this flag, the jack name is "Headset Mic Phantom", then the pulseaudio can handle its detection correctly. Fixes: f0ba9d699e5c ("ALSA: hda/realtek - Fix Dell headset Mic can't record") Cc: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - New VIA controller suppor no-snoop pathDavid Wang2018-04-161-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch is used to tell kernel that new VIA HDAC controller also support no-snoop path. [ minor coding style fix by tiwai ] Signed-off-by: David Wang <davidwang@zhaoxin.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: line6: Use correct endpoint type for midi outputFabián Inostroza2018-04-121-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Sending MIDI messages to a PODxt through the USB connection shows "usb_submit_urb failed" in dmesg and the message is not received by the POD. The error is caused because in the funcion send_midi_async() in midi.c there is a call to usb_sndbulkpipe() for endpoint 3 OUT, but the PODxt USB descriptor shows that this endpoint it's an interrupt endpoint. Patch tested with PODxt only. [ The bug has been present from the very beginning in the staging driver time, but Fixes below points to the commit moving to sound/ directory so that the fix can be cleanly applied -- tiwai ] Fixes: 61864d844c29 ("ALSA: move line6 usb driver into sound/usb") Signed-off-by: Fabián Inostroza <fabianinostroza@udec.cl> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | Merge tag 'drm-fixes-for-v4.17-rc1' of ↵Linus Torvalds2018-04-121-3/+1
|\ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://people.freedesktop.org/~airlied/linux Pull drm fixes from Dave Airlie: "One omap, and one alsa pm fix (we merged the breaking patch via drm tree). Otherwise it's two bunches of amdgpu fixes, removing an unneeded file, some DC fixes, HDMI audio regression fix, and some vega12 fixes" * tag 'drm-fixes-for-v4.17-rc1' of git://people.freedesktop.org/~airlied/linux: (27 commits) Revert "drm/amd/display: disable CRTCs with NULL FB on their primary plane (V2)" Revert "drm/amd/display: fix dereferencing possible ERR_PTR()" drm/amd/display: Fix regamma not affecting full-intensity color values drm/amd/display: Fix FBC text console corruption drm/amd/display: Only register backlight device if embedded panel connected drm/amd/display: fix brightness level after resume from suspend drm/amd/display: HDMI has no sound after Panel power off/on drm/amdgpu: add MP1 and THM hw ip base reg offset drm/amdgpu: fix null pointer panic with direct fw loading on gpu reset drm/radeon: add PX quirk for Asus K73TK drm/omap: fix crash if there's no video PLL drm/amdgpu: Fix memory leaks at amdgpu_init() error path drm/amdgpu: Fix PCIe lane width calculation drm/radeon: Fix PCIe lane width calculation drm/amdgpu/si: implement get/set pcie_lanes asic callback drm/amdgpu: Add support for SRBM selection v3 Revert "drm/amdgpu: Don't change preferred domian when fallback GTT v5" drm/amd/powerply: fix power reading on Fiji drm/amd/powerplay: Enable ACG SS feature drm/amdgpu/sdma: fix mask in emit_pipeline_sync ...
| * \ \ Merge tag 'drm-misc-next-fixes-2018-04-04' of ↵Dave Airlie2018-04-111-3/+1
| |\ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://anongit.freedesktop.org/drm/drm-misc into drm-next hda_intel: Don't declare azx PM ops if VGA_SWITCHEROO configured (Lukas) Cc: Lukas Wunner <lukas@wunner.de> Cc: Takashi Iwai <tiwai@suse.de> * tag 'drm-misc-next-fixes-2018-04-04' of git://anongit.freedesktop.org/drm/drm-misc: ALSA: hda - Silence PM ops build warning
| | * | | ALSA: hda - Silence PM ops build warningLukas Wunner2018-03-291-3/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The system sleep PM ops azx_suspend() and azx_resume() were previously called by vga_switcheroo, but commit 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") removed their invocation. Unfortunately the commit neglected to update the #ifdef surrounding the two functions, so if CONFIG_PM_SLEEP is *not* enabled but all three of CONFIG_PM, CONFIG_VGA_SWITCHEROO and CONFIG_SND_HDA_CODEC_HDMI *are* enabled, the compiler now emits the following warning: sound/pci/hda/hda_intel.c:1024:12: warning: 'azx_resume' defined but not used [-Wunused-function] static int azx_resume(struct device *dev) ^~~~~~~~~~ sound/pci/hda/hda_intel.c:989:12: warning: 'azx_suspend' defined but not used [-Wunused-function] static int azx_suspend(struct device *dev) ^~~~~~~~~~~ Silence by updating the #ifdef. Because the #ifdef block now uses the same condition as the one immediately succeeding it, the two blocks can be collapsed together, shaving off another two lines. Fixes: 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") Reviewed-by: Takashi Iwai <tiwai@suse.de> Reported-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Lukas Wunner <lukas@wunner.de> Link: https://patchwork.kernel.org/patch/10313441/ Link: https://patchwork.freedesktop.org/patch/msgid/b8e70e34a9acbd4f0a1a6c7673cea96888ae9503.1522323444.git.lukas@wunner.de
* | | | | Merge tag 'sound-fix-4.17-rc1' of ↵Linus Torvalds2018-04-103-77/+58
|\ \ \ \ \ | | |_|/ / | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "The main purpose of this pull request is a fix for a regression in the recent PCM OSS emulation code that may lead to RCU stall. Since syzkaller hits this too often, I send the pull request now with a minimal collection. Possibly another pull request may follow before RC1. The other fixes here are for USB-audio class 2 and 3 to improve the parser for the clock descriptors. These are rather cleanups but good for security, too. Last but not least, another included fix is the trivial one to remove superfluous WARN_ON() that annoyed syzbot" * tag 'sound-fix-4.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: pcm: Remove WARN_ON() at snd_pcm_hw_params() error ALSA: pcm: Fix endless loop for XRUN recovery in OSS emulation ALSA: usb-audio: Add sanity checks in UAC3 clock parsers ALSA: usb-audio: More strict sanity checks for clock parsers ALSA: usb-audio: Refactor clock finder helpers
| * | | | ALSA: pcm: Remove WARN_ON() at snd_pcm_hw_params() errorTakashi Iwai2018-04-091-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_pcm_hw_params() (more exactly snd_pcm_hw_params_choose()) contains a check of the return error from snd_pcm_hw_param_first() and _last() with snd_BUG_ON() -- i.e. it may trigger WARN_ON() depending on the kconfig. This was a valid check in the past, as these functions shouldn't return any error if the parameters have been already refined via snd_pcm_hw_refine() beforehand. However, the recent rewrite introduced a kmalloc() in snd_pcm_hw_refine() for removing VLA, and this brought a possibility to trigger an error. As a result, syzbot caught lots of superfluous kernel WARN_ON() and paniced via fault injection. As the WARN_ON() is no longer valid with the introduction of kmalloc(), let's drop snd_BUG_ON() check, in order to make the world peaceful place again. Reported-by: syzbot+803e0047ac3a3096bb4f@syzkaller.appspotmail.com Fixes: 5730f9f744cf ("ALSA: pcm: Remove VLA usage") Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: pcm: Fix endless loop for XRUN recovery in OSS emulationTakashi Iwai2018-04-071-2/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The commit 02a5d6925cd3 ("ALSA: pcm: Avoid potential races between OSS ioctls and read/write") split the PCM preparation code to a locked version, and it added a sanity check of runtime->oss.prepare flag along with the change. This leaded to an endless loop when the stream gets XRUN: namely, snd_pcm_oss_write3() and co call snd_pcm_oss_prepare() without setting runtime->oss.prepare flag and the loop continues until the PCM state reaches to another one. As the function is supposed to execute the preparation unconditionally, drop the invalid state check there. The bug was triggered by syzkaller. Fixes: 02a5d6925cd3 ("ALSA: pcm: Avoid potential races between OSS ioctls and read/write") Reported-by: syzbot+150189c103427d31a053@syzkaller.appspotmail.com Reported-by: syzbot+7e3f31a52646f939c052@syzkaller.appspotmail.com Reported-by: syzbot+4f2016cf5185da7759dc@syzkaller.appspotmail.com Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: usb-audio: Add sanity checks in UAC3 clock parsersTakashi Iwai2018-04-071-3/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The UAC3 clock parser codes lack of the sanity checks for malformed descriptors like UAC2 parser does. Without it, the driver may lead to a potential crash. Fixes: 9a2fe9b801f5 ("ALSA: usb: initial USB Audio Device Class 3.0 support") Tested-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: usb-audio: More strict sanity checks for clock parsersTakashi Iwai2018-04-071-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The sanity checks introduced for malformed descriptors loosely check the given descriptor size, although the size greater than the defined description is invalid. It was due to a concern of any funky firmware in the actual products. But this doesn't look hitting, and any sane products must have the defined descriptors. So in this patch, we make the validators more strict, allowing only with the defined descriptor sizes. The value in clock selector validator is corrected from 5 to 7 to count the two unlisted fields after baCSourceID[]. Suggested-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: usb-audio: Refactor clock finder helpersTakashi Iwai2018-04-071-74/+53
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There are lots of open-coded functions to find a clock source, selector and multiplier. Now there are both v2 and v3, so six variants. This patch refactors the code to use a common helper for the main loop, and define each validator function for each target. There is no functional change. Fixes: 9a2fe9b801f5 ("ALSA: usb: initial USB Audio Device Class 3.0 support") Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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