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2019-08-29ALSA: hda - Fix potential endless loop at applying quirksTakashi Iwai1-2/+2
Since the chained quirks via chained_before flag is applied before the depth check, it may lead to the endless recursive calls, when the chain were set up incorrectly. Fix it by moving the depth check at the beginning of the loop. Fixes: 1f57825077dc ("ALSA: hda - Add chained_before flag to the fixup entry") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-26ALSA: oxfw: fix to handle correct stream for PCM playbackTakashi Sakamoto1-1/+1
When userspace application calls ioctl(2) to configure hardware for PCM playback substream, ALSA OXFW driver handles incoming AMDTP stream. In this case, outgoing AMDTP stream should be handled. This commit fixes the bug for v5.3-rc kernel. Fixes: 4f380d007052 ("ALSA: oxfw: configure packet format in pcm.hw_params callback") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-25ALSA: seq: Fix potential concurrent access to the deleted poolTakashi Iwai3-2/+20
The input pool of a client might be deleted via the resize ioctl, the the access to it should be covered by the proper locks. Currently the only missing place is the call in snd_seq_ioctl_get_client_pool(), and this patch papers over it. Reported-by: syzbot+4a75454b9ca2777f35c7@syzkaller.appspotmail.com Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-22ALSA: usb-audio: Check mixer unit bitmap yet more strictlyTakashi Iwai1-8/+28
The bmControls (for UAC1) or bmMixerControls (for UAC2/3) bitmap has a variable size depending on both input and output pins. Its size is to fit with input * output bits. The problem is that the input size can't be determined simply from the unit descriptor itself but it needs to parse the whole connected sources. Although the uac_mixer_unit_get_channels() tries to check some possible overflow of this bitmap, it's incomplete due to the lack of the evaluation of input pins. For covering possible overflows, this patch adds the bitmap overflow check in the loop of input pins in parse_audio_mixer_unit(). Fixes: 0bfe5e434e66 ("ALSA: usb-audio: Check mixer unit descriptors more strictly") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-21ALSA: line6: Fix memory leak at line6_init_pcm() error pathTakashi Iwai1-9/+9
I forgot to release the allocated object at the early error path in line6_init_pcm(). For addressing it, slightly shuffle the code so that the PCM destructor (pcm->private_free) is assigned properly before all error paths. Fixes: 3450121997ce ("ALSA: line6: Fix write on zero-sized buffer") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-21ALSA: usb-audio: Fix invalid NULL check in snd_emuusb_set_samplerate()Takashi Iwai1-4/+4
The quirk function snd_emuusb_set_samplerate() has a NULL check for the mixer element, but this is useless in the current code. It used to be a check against mixer->id_elems[unitid] but it was changed later to the value after mixer_eleme_list_to_info() which is always non-NULL due to the container_of() usage. This patch fixes the check before the conversion. While we're at it, correct a typo in the comment in the function, too. Fixes: 8c558076c740 ("ALSA: usb-audio: Clean up mixer element list traverse") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-20ALSA: hda/ca0132 - Add new SBZ quirkPaweł Rekowski1-0/+1
This patch adds a new PCI subsys ID for the SBZ, as found and tested by me and some reddit users. Link: https://lore.kernel.org/lkml/20190819204008.14426-1-p.rekowski@gmail.com Signed-off-by: Paweł Rekowski <p.rekowski@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-20ALSA: usb-audio: Add implicit fb quirk for Behringer UFX1604Takashi Iwai1-0/+1
Behringer UFX1604 requires the similar quirk to apply implicit fb like another Behringer model UFX1204 in order to fix the noisy playback. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204631 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-19ALSA: hda - Fixes inverted Conexant GPIO mic mute ledJeronimo Borque1-8/+9
"enabled" parameter historically referred to the device input or output, not to the led indicator. After the changes added with the led helper functions the mic mute led logic refers to the led and not to the mic input which caused led indicator to be negated. Fixing logic in cxt_update_gpio_led and updated cxt_fixup_gpio_mute_hook Also updated debug messages to ease further debugging if necessary. Fixes: 184e302b46c9 ("ALSA: hda/conexant - Use the mic-mute LED helper") Suggested-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jeronimo Borque <jeronimo@borque.com.ar> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-15ALSA: usb-audio: Fix a stack buffer overflow bug in check_input_termHui Peng1-8/+27
`check_input_term` recursively calls itself with input from device side (e.g., uac_input_terminal_descriptor.bCSourceID) as argument (id). In `check_input_term`, if `check_input_term` is called with the same `id` argument as the caller, it triggers endless recursive call, resulting kernel space stack overflow. This patch fixes the bug by adding a bitmap to `struct mixer_build` to keep track of the checked ids and stop the execution if some id has been checked (similar to how parse_audio_unit handles unitid argument). Reported-by: Hui Peng <benquike@gmail.com> Reported-by: Mathias Payer <mathias.payer@nebelwelt.net> Signed-off-by: Hui Peng <benquike@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-14ALSA: usb-audio: Fix an OOB bug in parse_audio_mixer_unitHui Peng1-0/+2
The `uac_mixer_unit_descriptor` shown as below is read from the device side. In `parse_audio_mixer_unit`, `baSourceID` field is accessed from index 0 to `bNrInPins` - 1, the current implementation assumes that descriptor is always valid (the length of descriptor is no shorter than 5 + `bNrInPins`). If a descriptor read from the device side is invalid, it may trigger out-of-bound memory access. ``` struct uac_mixer_unit_descriptor { __u8 bLength; __u8 bDescriptorType; __u8 bDescriptorSubtype; __u8 bUnitID; __u8 bNrInPins; __u8 baSourceID[]; } ``` This patch fixes the bug by add a sanity check on the length of the descriptor. Reported-by: Hui Peng <benquike@gmail.com> Reported-by: Mathias Payer <mathias.payer@nebelwelt.net> Cc: <stable@vger.kernel.org> Signed-off-by: Hui Peng <benquike@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-14ALSA: hda - Add a generic reboot_notifyHui Wang4-15/+22
Make codec enter D3 before rebooting or poweroff can fix the noise issue on some laptops. And in theory it is harmless for all codecs to enter D3 before rebooting or poweroff, let us add a generic reboot_notify, then realtek and conexant drivers can call this function. Cc: stable@vger.kernel.org Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-14ALSA: hda - Let all conexant codec enter D3 when rebootingHui Wang1-9/+0
We have 3 new lenovo laptops which have conexant codec 0x14f11f86, these 3 laptops also have the noise issue when rebooting, after letting the codec enter D3 before rebooting or poweroff, the noise disappers. Instead of adding a new ID again in the reboot_notify(), let us make this function apply to all conexant codec. In theory make codec enter D3 before rebooting or poweroff is harmless, and I tested this change on a couple of other Lenovo laptops which have different conexant codecs, there is no side effect so far. Cc: stable@vger.kernel.org Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-13ALSA: hda/realtek - Add quirk for HP Envy x360Takashi Iwai1-0/+1
HP Envy x360 (AMD Ryzen-based model) with 103c:8497 needs the same quirk like HP Spectre x360 for enabling the mute LED over Mic3 pin. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204373 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-10ALSA: hda - Fix a memory leak bugWenwen Wang1-1/+1
In snd_hda_parse_generic_codec(), 'spec' is allocated through kzalloc(). Then, the pin widgets in 'codec' are parsed. However, if the parsing process fails, 'spec' is not deallocated, leading to a memory leak. To fix the above issue, free 'spec' before returning the error. Fixes: 352f7f914ebb ("ALSA: hda - Merge Realtek parser code to generic parser") Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-09ALSA: hda - Apply workaround for another AMD chip 1022:1487Takashi Iwai1-0/+3
MSI MPG X570 board is with another AMD HD-audio controller (PCI ID 1022:1487) and it requires the same workaround applied for X370, etc (PCI ID 1022:1457). BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-08ALSA: firewire: fix a memory leak bugWenwen Wang1-1/+1
In iso_packets_buffer_init(), 'b->packets' is allocated through kmalloc_array(). Then, the aligned packet size is checked. If it is larger than PAGE_SIZE, -EINVAL will be returned to indicate the error. However, the allocated 'b->packets' is not deallocated on this path, leading to a memory leak. To fix the above issue, free 'b->packets' before returning the error code. Fixes: 31ef9134eb52 ("ALSA: add LaCie FireWire Speakers/Griffin FireWave Surround driver") Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu> Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Cc: <stable@vger.kernel.org> # v2.6.39+ Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-08sound: fix a memory leak bugWenwen Wang1-1/+2
In sound_insert_unit(), the controlling structure 's' is allocated through kmalloc(). Then it is added to the sound driver list by invoking __sound_insert_unit(). Later on, if __register_chrdev() fails, 's' is removed from the list through __sound_remove_unit(). If 'index' is not less than 0, -EBUSY is returned to indicate the error. However, 's' is not deallocated on this execution path, leading to a memory leak bug. To fix the above issue, free 's' before -EBUSY is returned. Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-07ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)Takashi Iwai3-2/+70
A long-time problem on the recent AMD chip (X370, X470, B450, etc with PCI ID 1022:1457) with Realtek codecs is the crackled or distorted sound for capture streams, as well as occasional playback hiccups. After lengthy debugging sessions, the workarounds we've found are like the following: - Set up the proper driver caps for this controller, similar as the other AMD controller. - Correct the DMA position reporting with the fixed FIFO size, which is similar like as workaround used for VIA chip set. - Even after the position correction, PulseAudio still shows mysterious stalls of playback streams when a capture is triggered in timer-scheduled mode. Since we have no clear way to eliminate the stall, pass the BATCH PCM flag for PA to suppress the tsched mode as a temporary workaround. This patch implements the workarounds. For the driver caps, it defines a new preset, AXZ_DCAPS_PRESET_AMD_SB. It enables the FIFO- corrected position reporting (corresponding to the new position_fix=6) and enforces the SNDRV_PCM_INFO_BATCH flag. Note that the current implementation is merely a workaround. Hopefully we'll find a better alternative in future, especially about removing the BATCH flag hack again. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-07ALSA: hiface: fix multiple memory leak bugsWenwen Wang1-3/+8
In hiface_pcm_init(), 'rt' is firstly allocated through kzalloc(). Later on, hiface_pcm_init_urb() is invoked to initialize 'rt->out_urbs[i]'. In hiface_pcm_init_urb(), 'rt->out_urbs[i].buffer' is allocated through kzalloc(). However, if hiface_pcm_init_urb() fails, both 'rt' and 'rt->out_urbs[i].buffer' are not deallocated, leading to memory leak bugs. Also, 'rt->out_urbs[i].buffer' is not deallocated if snd_pcm_new() fails. To fix the above issues, free 'rt' and 'rt->out_urbs[i].buffer'. Fixes: a91c3fb2f842 ("Add M2Tech hiFace USB-SPDIF driver") Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-06ALSA: hda - Don't override global PCM hw info flagTakashi Iwai1-4/+2
The commit bfcba288b97f ("ALSA - hda: Add support for link audio time reporting") introduced the conditional PCM hw info setup, but it overwrites the global azx_pcm_hw object. This will cause a problem if any other HD-audio controller, as it'll inherit the same bit flag although another controller doesn't support that feature. Fix the bug by setting the PCM hw info flag locally. Fixes: bfcba288b97f ("ALSA - hda: Add support for link audio time reporting") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-06ALSA: usb-audio: fix a memory leak bugWenwen Wang1-0/+1
In snd_usb_get_audioformat_uac3(), a structure for channel maps 'chmap' is allocated through kzalloc() before the execution goto 'found_clock'. However, this structure is not deallocated if the memory allocation for 'pd' fails, leading to a memory leak bug. To fix the above issue, free 'fp->chmap' before returning NULL. Fixes: 7edf3b5e6a45 ("ALSA: usb-audio: AudioStreaming Power Domain parsing") Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-08-06ASoC: max98373: Remove executable bitsMark Brown2-0/+0
Signed-off-by: Mark Brown <broonie@kernel.org>
2019-08-02ASoC: amd: acp3x: use dma address for acp3x dma driverVijendar Mukunda1-9/+5
We shouldn't assume CPU physical address we get from page_to_phys() is same as DMA address we get from dma_alloc_coherent(). On x86_64, we won't run into any problem with the assumption when dma_ops is nommu_dma_ops. However, DMA address is IOVA when IOMMU is enabled. And it's most likely different from CPU physical address when AMD IOMMU is not in passthrough mode. This patch fixes page faults when IOMMU is enabled. Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com> Link: https://lore.kernel.org/r/1564753899-17124-2-git-send-email-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-08-02ASoC: amd: acp3x: use dma_ops of parent device for acp3x dma driverVijendar Mukunda1-2/+4
AMD platform device acp3x_rv_i2s created by parent PCI device driver. Pass struct device of the parent to snd_pcm_lib_preallocate_pages() so dma_alloc_coherent() can use correct dma_ops. Otherwise, it will use default dma_ops which is nommu_dma_ops on x86_64 even when IOMMU is enabled and set to non passthrough mode. Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com> Link: https://lore.kernel.org/r/1564753899-17124-1-git-send-email-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-07-31ASoC: max98373: add 88200 and 96000 sampling rate supportfengchunguo2-0/+8
88200 and 96000 sampling rate was not enabled on driver, so can't be played. The error information: max98373 3-0031：rate 96000 not supported max98373 3-0031：ASoC: can't set max98373-aif1 hw params: -22 Signed-off-by: fengchunguo <chunguo.feng@amlogic.com> Link: https://lore.kernel.org/r/20190731074156.5620-1-chunguo.feng@amlogic.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-07-31ASoC: sun4i-i2s: Incorrect SR and WSS computationMarcus Cooper1-2/+2
The A64 audio codec uses the original I2S block but the SR and WSS computation currently assigned is for the newer block. Fixes: 619c15f7fac9 (ASoC: sun4i-i2s: Change SR and WSS computation) Signed-off-by: Marcus Cooper <codekipper@gmail.com> Link: https://lore.kernel.org/r/20190729152130.27955-1-codekipper@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-07-31MAINTAINERS: Update Intel ASoC drivers maintainersCezary Rojewski1-0/+1
Adding myself to Intel ASoC drivers maintainers list. Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20190726181517.27655-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-07-30ALSA: usb-audio: Fix gpf in snd_usb_pipe_sanity_checkHillf Danton1-1/+1
syzbot found the following crash on: general protection fault: 0000 [#1] SMP KASAN RIP: 0010:snd_usb_pipe_sanity_check+0x80/0x130 sound/usb/helper.c:75 Call Trace: snd_usb_motu_microbookii_communicate.constprop.0+0xa0/0x2fb sound/usb/quirks.c:1007 snd_usb_motu_microbookii_boot_quirk sound/usb/quirks.c:1051 [inline] snd_usb_apply_boot_quirk.cold+0x163/0x370 sound/usb/quirks.c:1280 usb_audio_probe+0x2ec/0x2010 sound/usb/card.c:576 usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361 really_probe+0x281/0x650 drivers/base/dd.c:548 .... It was introduced in commit 801ebf1043ae for checking pipe and endpoint types. It is fixed by adding a check of the ep pointer in question. BugLink: https://syzkaller.appspot.com/bug?extid=d59c4387bfb6eced94e2 Reported-by: syzbot <syzbot+d59c4387bfb6eced94e2@syzkaller.appspotmail.com> Fixes: 801ebf1043ae ("ALSA: usb-audio: Sanity checks for each pipe and EP types") Cc: Andrey Konovalov <andreyknvl@google.com> Signed-off-by: Hillf Danton <hdanton@sina.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-07-29ALSA: pcm: fix lost wakeup event scenarios in snd_pcm_drainYuki Tsunashima1-1/+2
lost wakeup can occur after enabling irq, therefore put task into interruptible before enabling interrupts, without this change, task can be put to sleep and snd_pcm_drain will delay Fixes: f2b3614cefb6 ("ALSA: PCM - Don't check DMA time-out too shortly") Signed-off-by: Yuki Tsunashima <ytsunashima@jp.adit-jv.com> Signed-off-by: Suresh Udipi <sudipi@jp.adit-jv.com> [ported from 4.9] Signed-off-by: Adam Miartus <amiartus@de.adit-jv.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-07-27ALSA: hda: Fix 1-minute detection delay when i915 module is not availableSamuel Thibault1-4/+6
Distribution installation images such as Debian include different sets of modules which can be downloaded dynamically. Such images may notably include the hda sound modules but not the i915 DRM module, even if the latter was enabled at build time, as reported on https://bugs.debian.org/931507 In such a case hdac_i915 would be linked in and try to load the i915 module, fail since it is not there, but still wait for a whole minute before giving up binding with it. This fixes such as case by only waiting for the binding if the module was properly loaded (or module support is disabled, in which case i915 is already compiled-in anyway). Fixes: f9b54e1961c7 ("ALSA: hda/i915: Allow delayed i915 audio component binding") Signed-off-by: Samuel Thibault <samuel.thibault@ens-lyon.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-07-26ASoC: ti: davinci-mcasp: Correct slot_width posed constraintPeter Ujfalusi1-9/+34
The slot_width is a property for the bus while the constraint for SNDRV_PCM_HW_PARAM_SAMPLE_BITS is for the in memory format. Applying slot_width constraint to sample_bits works most of the time, but it will blacklist valid formats in some cases. With slot_width 24 we can support S24_3LE and S24_LE formats as they both look the same on the bus, but a a 24 constraint on sample_bits would not allow S24_LE as it is stored in 32bits in memory. Implement a simple hw_rule function to allow all formats which require less or equal number of bits on the bus as slot_width (if configured). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Link: https://lore.kernel.org/r/20190726064244.3762-2-peter.ujfalusi@ti.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-07-26ASoC: rockchip: Fix mono captureCheng-Yi Chiang1-3/+2
This reverts commit db51707b9c9aeedd310ebce60f15d5bb006567e0. Revert "ASoC: rockchip: i2s: Support mono capture" Previous discussion in https://patchwork.kernel.org/patch/10147153/ explains the issue of the patch. While device is configured as 1-ch, hardware is still generating a 2-ch stream. When user space reads the data and assumes it is a 1-ch stream, the rate will be slower by 2x. Revert the change so 1-ch is not supported. User space can selectively take one channel data out of two channel if 1-ch is preferred. Currently, both channels record identical data. Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org> Link: https://lore.kernel.org/r/20190726044202.26866-1-cychiang@chromium.org Signed-off-by: Mark Brown <broonie@kernel.org>
2019-07-26ASoC: Intel: Fix some acpi vs apci typo in somme commentsChristophe JAILLET10-10/+10
Fix some typo to have the filaname given in a comment match the real name of the file. Some 'acpi' have erroneously been written 'apci' Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr> Link: https://lore.kernel.org/r/20190725053523.16542-1-christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown <broonie@kernel.org>
2019-07-26ASoC: ti: davinci-mcasp: Fix clk PDIR handling for i2s master modePeter Ujfalusi1-1/+2
When running McASP as master capture alone will not record any audio unless a parallel playback stream is running. As soon as the playback stops the captured data is going to be silent again. In McASP master mode we need to set the PDIR for the clock pins and fix the mcasp_set_axr_pdir() to skip the bits in the PDIR registers above AMUTE. This went unnoticed as most of the boards uses McASP as slave and neither of these issues are visible (audible) in those setups. Fixes: ca3d9433349e ("ASoC: davinci-mcasp: Update PDIR (pin direction) register handling") Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Link: https://lore.kernel.org/r/20190725083423.7321-1-peter.ujfalusi@ti.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-07-25ALSA: hda - Add a conexant codec entry to let mute led workHui Wang1-0/+1
This conexant codec isn't in the supported codec list yet, the hda generic driver can drive this codec well, but on a Lenovo machine with mute/mic-mute leds, we need to apply CXT_FIXUP_THINKPAD_ACPI to make the leds work. After adding this codec to the list, the driver patch_conexant.c will apply THINKPAD_ACPI to this machine. Cc: stable@vger.kernel.org Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-07-25ALSA: hda - Fix intermittent CORB/RIRB stall on Intel chipsTakashi Iwai1-3/+2
It turned out that the recent Intel HD-audio controller chips show a significant stall during the system PM resume intermittently. It doesn't happen so often and usually it may read back successfully after one or more seconds, but in some rare worst cases the driver went into fallback mode. After trial-and-error, we found out that the communication stall seems covered by issuing the sync after each verb write, as already done for AMD and other chipsets. So this patch enables the write-sync flag for the recent Intel chips, Skylake and onward, as a workaround. Also, since Broxton and co have the very same driver flags as Skylake, refer to the Skylake driver flags instead of defining the same contents again for simplification. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=201901 Reported-and-tested-by: Todd Brandt <todd.e.brandt@linux.intel.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-07-24ASoC: Fail card instantiation if DAI format setup failsRicard Wanderlof1-2/+5
If the DAI format setup fails, there is no valid communication format between CPU and CODEC, so fail card instantiation, rather than continue with a card that will most likely not function properly. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Link: https://lore.kernel.org/r/alpine.DEB.2.20.1907241132350.6338@lnxricardw1.se.axis.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-07-23ALSA: ac97: Fix double free of ac97_codec_deviceDing Xiang1-9/+4
put_device will call ac97_codec_release to free ac97_codec_device and other resources, so remove the kfree and other redundant code. Fixes: 74426fbff66e ("ALSA: ac97: add an ac97 bus") Signed-off-by: Ding Xiang <dingxiang@cmss.chinamobile.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-07-23ASoC: SOF: Intel: hda: remove misleading error trace from IRQ threadKai Vehmanen2-4/+4
Downgrade "nothing to do in IRQ thread" message from error to a debug message in the IPC interrupt handler thread. The spurious wake-up can happen if a HDA stream interrupt is raised while the IPC interrupt thread is running. IPC functionality is not impacted by this condition, so debug is a more appropriate trace level. Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20190722141402.7194-21-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-07-23ASoC: qcom: apq8016_sbc: Fix oops with multiple DAI linksStephan Gerhold1-8/+8
apq8016_sbc_parse_of() sets up multiple DAI links, depending on the number of nodes in the device tree. However, at the moment CPU and platform components are only allocated for the first link. This causes an oops when more than one link is defined: Internal error: Oops: 96000044 [#1] SMP CPU: 0 PID: 1015 Comm: kworker/0:2 Not tainted 5.3.0-rc1 #4 Call trace: apq8016_sbc_platform_probe+0x1a8/0x3f0 platform_drv_probe+0x50/0xa0 ... Move the allocation inside the loop to ensure that each link is properly initialized. Fixes: 98b232ca9e0e ("ASoC: qcom: apq8016_sbc: use modern dai_link style") Signed-off-by: Stephan Gerhold <stephan@gerhold.net> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/20190722130352.95874-1-stephan@gerhold.net Signed-off-by: Mark Brown <broonie@kernel.org>
2019-07-23ALSA: compress: Be more restrictive about when a drain is allowedCharles Keepax1-0/+6
Draining makes little sense in the situation of hardware overrun, as the hardware will have consumed all its available samples. Additionally, draining whilst the stream is paused would presumably get stuck as no data is being consumed on the DSP side. Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Acked-by: Vinod Koul <vkoul@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-07-23ALSA: compress: Don't allow paritial drain operations on capture streamsCharles Keepax1-0/+8
Partial drain and next track are intended for gapless playback and don't really have an obvious interpretation for a capture stream, so makes sense to not allow those operations on capture streams. Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Acked-by: Vinod Koul <vkoul@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-07-23ALSA: compress: Prevent bypasses of set_paramsCharles Keepax1-6/+24
Currently, whilst in SNDRV_PCM_STATE_OPEN it is possible to call snd_compr_stop, snd_compr_drain and snd_compr_partial_drain, which allow a transition to SNDRV_PCM_STATE_SETUP. The stream should only be able to move to the setup state once it has received a SNDRV_COMPRESS_SET_PARAMS ioctl. Fix this issue by not allowing those ioctls whilst in the open state. Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Acked-by: Vinod Koul <vkoul@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-07-23ALSA: compress: Fix regression on compressed capture streamsCharles Keepax2-9/+12
A previous fix to the stop handling on compressed capture streams causes some knock on issues. The previous fix updated snd_compr_drain_notify to set the state back to PREPARED for capture streams. This causes some issues however as the handling for snd_compr_poll differs between the two states and some user-space applications were relying on the poll failing after the stream had been stopped. To correct this regression whilst still fixing the original problem the patch was addressing, update the capture handling to skip the PREPARED state rather than skipping the SETUP state as it has done until now. Fixes: 4f2ab5e1d13d ("ALSA: compress: Fix stop handling on compressed capture streams") Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Acked-by: Vinod Koul <vkoul@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-07-22ASoC: dapm: fix a memory leak bugWenwen Wang1-0/+2
In snd_soc_dapm_new_control_unlocked(), a kernel buffer is allocated in dapm_cnew_widget() to hold the new dapm widget. Then, different actions are taken according to the id of the widget, i.e., 'w->id'. If any failure occurs during this process, snd_soc_dapm_new_control_unlocked() should be terminated by going to the 'request_failed' label. However, the allocated kernel buffer is not freed on this code path, leading to a memory leak bug. To fix the above issue, free the buffer before returning from snd_soc_dapm_new_control_unlocked() through the 'request_failed' label. Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu> Link: https://lore.kernel.org/r/1563803864-2809-1-git-send-email-wang6495@umn.edu Signed-off-by: Mark Brown <broonie@kernel.org>
2019-07-22ASoC: SOF: use __u32 instead of uint32_t in uapi headersMasahiro Yamada2-13/+17
When CONFIG_UAPI_HEADER_TEST=y, exported headers are compile-tested to make sure they can be included from user-space. Currently, header.h and fw.h are excluded from the test coverage. To make them join the compile-test, we need to fix the build errors attached below. For a case like this, we decided to use __u{8,16,32,64} variable types in this discussion: https://lkml.org/lkml/2019/6/5/18 Build log: CC usr/include/sound/sof/header.h.s CC usr/include/sound/sof/fw.h.s In file included from <command-line>:32:0: ./usr/include/sound/sof/header.h:19:2: error: unknown type name ‘uint32_t’ uint32_t magic; /**< 'S', 'O', 'F', '\0' */ ^~~~~~~~ ./usr/include/sound/sof/header.h:20:2: error: unknown type name ‘uint32_t’ uint32_t type; /**< component specific type */ ^~~~~~~~ ./usr/include/sound/sof/header.h:21:2: error: unknown type name ‘uint32_t’ uint32_t size; /**< size in bytes of data excl. this struct */ ^~~~~~~~ ./usr/include/sound/sof/header.h:22:2: error: unknown type name ‘uint32_t’ uint32_t abi; /**< SOF ABI version */ ^~~~~~~~ ./usr/include/sound/sof/header.h:23:2: error: unknown type name ‘uint32_t’ uint32_t reserved[4]; /**< reserved for future use */ ^~~~~~~~ ./usr/include/sound/sof/header.h:24:2: error: unknown type name ‘uint32_t’ uint32_t data[0]; /**< Component data - opaque to core */ ^~~~~~~~ In file included from <command-line>:32:0: ./usr/include/sound/sof/fw.h:49:2: error: unknown type name ‘uint32_t’ uint32_t size; /* bytes minus this header */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:50:2: error: unknown type name ‘uint32_t’ uint32_t offset; /* offset from base */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:64:2: error: unknown type name ‘uint32_t’ uint32_t size; /* bytes minus this header */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:65:2: error: unknown type name ‘uint32_t’ uint32_t num_blocks; /* number of blocks */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:73:2: error: unknown type name ‘uint32_t’ uint32_t file_size; /* size of file minus this header */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:74:2: error: unknown type name ‘uint32_t’ uint32_t num_modules; /* number of modules */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:75:2: error: unknown type name ‘uint32_t’ uint32_t abi; /* version of header format */ ^~~~~~~~ Signed-off-by: Masahiro Yamada <yamada.masahiro@socionext.com> Link: https://lore.kernel.org/r/20190721142308.30306-1-yamada.masahiro@socionext.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-07-22SoC: rockchip: rockchip_max98090: Enable MICBIAS for headset keypress detectionEnric Balletbo i Serra1-0/+32
The TS3A227E says that the headset keypress detection needs the MICBIAS power in order to report the key events to ensure proper operation The headset keypress detection needs the MICBIAS power in order to report the key events all the time as long as MIC is present. So MICBIAS pin is forced on when a MICROPHONE is detected. On Veyron Minnie I observed that if the MICBIAS power is not present and the key press detection is activated (just because it is enabled when you insert a headset), it randomly reports a keypress on insert. E.g. (KEY_PLAYPAUSE) Event: (SW_HEADPHONE_INSERT), value 1 Event: (SW_MICROPHONE_INSERT), value 1 Event: -------------- SYN_REPORT ------------ Event: (KEY_PLAYPAUSE), value 1 Userspace thinks that KEY_PLAYPAUSE is pressed and produces the annoying effect that the media player starts a play/pause loop. Note that, although most of the time the key reported is the one associated with BTN_0, not always this is true. On my tests I also saw different keys reported Signed-off-by: Enric Balletbo i Serra <enric.balletbo@collabora.com> Link: https://lore.kernel.org/r/20190719173929.24065-1-enric.balletbo@collabora.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-07-22ASoC: cs42xx8: Fix MFREQ selection issue for async modeShengjiu Wang1-19/+97
When sample rate of TX is different with sample rate of RX in async mode, the MFreq selection will be wrong. For example, sysclk = 24.576MHz, TX rate = 96000Hz, RX rate = 48000Hz. Then ratio of TX = 256, ratio of RX = 512, For MFreq is shared by TX and RX instance, the correct value of MFreq is 2 for both TX and RX. But original method will cause MFreq = 0 for TX, MFreq = 2 for RX. If TX is started after RX, RX will be impacted, RX work abnormal with MFreq = 0. This patch is to select proper MFreq value according to TX rate and RX rate. Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver support for CS42448/CS42888") Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com> Link: https://lore.kernel.org/r/20190716094547.46787-1-shengjiu.wang@nxp.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-07-22ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walksCharles Keepax1-4/+4
DPCM uses snd_soc_dapm_dai_get_connected_widgets to build a list of the widgets connected to a specific front end DAI so it can search through this list for available back end DAIs. The custom_stop_condition was added to is_connected_ep to facilitate this list not containing more widgets than is necessary. Doing so both speeds up the DPCM handling as less widgets need to be searched and avoids issues with CODEC to CODEC links as these would be confused with back end DAIs if they appeared in the list of available widgets. custom_stop_condition was implemented by aborting the graph walk when the condition is triggered, however there is an issue with this approach. Whilst walking the graph is_connected_ep should update the endpoints cache on each widget, if the walk is aborted the number of attached end points is unknown for that sub-graph. When the stop condition triggered, the original patch ignored the triggering widget and returned zero connected end points; a later patch updated this to set the triggering widget's cache to 1 and return that. Both of these approaches result in inaccurate values being stored in various end point caches as the values propagate back through the graph, which can result in later issues with widgets powering/not powering unexpectedly. As the original goal was to reduce the size of the widget list passed to the DPCM code, the simplest solution is to limit the functionality of the custom_stop_condition to the widget list. This means the rest of the graph will still be processed resulting in correct end point caches, but only widgets up to the stop condition will be added to the returned widget list. Fixes: 6742064aef7f ("ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets") Fixes: 5fdd022c2026 ("ASoC: dpcm: play nice with CODEC<->CODEC links") Fixes: 09464974eaa8 ("ASoC: dapm: Fix to return correct path list in is_connected_ep.") Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20190718084333.15598-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org>
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