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* Merge tag 'sound-5.3-rc7' of ↵Linus Torvalds2019-08-2710-32/+73
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "A collection of small fixes as usual: - More coverage of USB-audio descriptor sanity checks - A fix for mute LED regression on Conexant HD-audio codecs - A few device-specific fixes and quirks for USB-audio and HD-audio - A fix for (die-hard remaining) possible race in sequencer core - FireWire oxfw regression fix that was introduced in 5.3-rc1" * tag 'sound-5.3-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: oxfw: fix to handle correct stream for PCM playback ALSA: seq: Fix potential concurrent access to the deleted pool ALSA: usb-audio: Check mixer unit bitmap yet more strictly ALSA: line6: Fix memory leak at line6_init_pcm() error path ALSA: usb-audio: Fix invalid NULL check in snd_emuusb_set_samplerate() ALSA: hda/ca0132 - Add new SBZ quirk ALSA: usb-audio: Add implicit fb quirk for Behringer UFX1604 ALSA: hda - Fixes inverted Conexant GPIO mic mute led
| * ALSA: oxfw: fix to handle correct stream for PCM playbackTakashi Sakamoto2019-08-261-1/+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>
| * ALSA: seq: Fix potential concurrent access to the deleted poolTakashi Iwai2019-08-253-2/+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>
| * ALSA: usb-audio: Check mixer unit bitmap yet more strictlyTakashi Iwai2019-08-221-8/+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>
| * ALSA: line6: Fix memory leak at line6_init_pcm() error pathTakashi Iwai2019-08-211-9/+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>
| * ALSA: usb-audio: Fix invalid NULL check in snd_emuusb_set_samplerate()Takashi Iwai2019-08-211-4/+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>
| * ALSA: hda/ca0132 - Add new SBZ quirkPaweł Rekowski2019-08-201-0/+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>
| * ALSA: usb-audio: Add implicit fb quirk for Behringer UFX1604Takashi Iwai2019-08-201-0/+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>
| * ALSA: hda - Fixes inverted Conexant GPIO mic mute ledJeronimo Borque2019-08-191-8/+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>
* | Merge tag 'sound-5.3-rc5' of ↵Linus Torvalds2019-08-166-33/+56
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "All small fixes targeted for stable: - Two fixes for USB-audio with malformed descriptor, spotted by fuzzers - Two fixes Conexant HD-audio codec wrt power management - Quirks for HD-audio AMD platform and HP laptop - HD-audio memory leak fix" * tag 'sound-5.3-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: usb-audio: Fix a stack buffer overflow bug in check_input_term ALSA: usb-audio: Fix an OOB bug in parse_audio_mixer_unit ALSA: hda - Add a generic reboot_notify ALSA: hda - Let all conexant codec enter D3 when rebooting ALSA: hda/realtek - Add quirk for HP Envy x360 ALSA: hda - Fix a memory leak bug ALSA: hda - Apply workaround for another AMD chip 1022:1487
| * ALSA: usb-audio: Fix a stack buffer overflow bug in check_input_termHui Peng2019-08-151-8/+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>
| * ALSA: usb-audio: Fix an OOB bug in parse_audio_mixer_unitHui Peng2019-08-141-0/+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>
| * ALSA: hda - Add a generic reboot_notifyHui Wang2019-08-144-15/+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>
| * ALSA: hda - Let all conexant codec enter D3 when rebootingHui Wang2019-08-141-9/+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>
| * ALSA: hda/realtek - Add quirk for HP Envy x360Takashi Iwai2019-08-131-0/+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>
| * ALSA: hda - Fix a memory leak bugWenwen Wang2019-08-101-1/+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>
| * ALSA: hda - Apply workaround for another AMD chip 1022:1487Takashi Iwai2019-08-091-0/+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>
* | Merge tag 'sound-5.3-rc4' of ↵Linus Torvalds2019-08-0943-144/+365
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "Lots of small fixes at this time since we've received the ASoC fix batch now. - Some coverage in ASoC core mostly for minor issues like NULL checks for DPCM and proper error handling in DAI instantiation - A collection of small device-specific changes in various ASoC codec and platform drivers - OF-tree refcount fixes in a few ASoC drivers - Fixes of memory leaks in the error paths of various ASoC / ALSA drivers - A workaround for a long-standing issue on AMD HD-audio device - Updates of MAINTAINERS, mail addresses, file permission fixups" * tag 'sound-5.3-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (38 commits) ALSA: firewire: fix a memory leak bug sound: fix a memory leak bug ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457) ALSA: hiface: fix multiple memory leak bugs ALSA: hda - Don't override global PCM hw info flag ALSA: usb-audio: fix a memory leak bug ASoC: max98373: Remove executable bits ASoC: amd: acp3x: use dma address for acp3x dma driver ASoC: amd: acp3x: use dma_ops of parent device for acp3x dma driver ASoC: max98373: add 88200 and 96000 sampling rate support ASoC: sun4i-i2s: Incorrect SR and WSS computation MAINTAINERS: Update Intel ASoC drivers maintainers ASoC: ti: davinci-mcasp: Correct slot_width posed constraint ASoC: rockchip: Fix mono capture ASoC: Intel: Fix some acpi vs apci typo in somme comments ASoC: ti: davinci-mcasp: Fix clk PDIR handling for i2s master mode ASoC: Fail card instantiation if DAI format setup fails ASoC: SOF: Intel: hda: remove misleading error trace from IRQ thread ASoC: qcom: apq8016_sbc: Fix oops with multiple DAI links ASoC: dapm: fix a memory leak bug ...
| * ALSA: firewire: fix a memory leak bugWenwen Wang2019-08-081-1/+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>
| * sound: fix a memory leak bugWenwen Wang2019-08-081-1/+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>
| * ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)Takashi Iwai2019-08-073-2/+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>
| * ALSA: hiface: fix multiple memory leak bugsWenwen Wang2019-08-071-3/+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>
| * ALSA: hda - Don't override global PCM hw info flagTakashi Iwai2019-08-061-4/+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>
| * ALSA: usb-audio: fix a memory leak bugWenwen Wang2019-08-061-0/+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>
| * Merge tag 'asoc-fix-v5.3-rc3-2' of ↵Takashi Iwai2019-08-062-0/+0
| |\ | | | | | | | | | | | | | | | | | | | | | | | | https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.3 Incremental fix removing executable bits added in a prior patch accidentally.
| | * ASoC: max98373: Remove executable bitsMark Brown2019-08-062-0/+0
| | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@kernel.org>
| * | Merge tag 'asoc-fix-v5.3-rc3' of ↵Takashi Iwai2019-08-0636-133/+281
| |\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.3 A relatively large batch of mostly unremarkable fixes here, a couple of small core fixes for fairly obscure issues, more comment/email updates with no code impact than usual and a bunch of small driver fixes. The support for new sample rates in the max98373 driver is a fix for the fact that the driver declared support for those rates but would in fact return an error if these rates were selected.
| | * ASoC: amd: acp3x: use dma address for acp3x dma driverVijendar Mukunda2019-08-021-9/+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>
| | * ASoC: amd: acp3x: use dma_ops of parent device for acp3x dma driverVijendar Mukunda2019-08-021-2/+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>
| | * ASoC: max98373: add 88200 and 96000 sampling rate supportfengchunguo2019-07-312-0/+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>
| | * ASoC: sun4i-i2s: Incorrect SR and WSS computationMarcus Cooper2019-07-311-2/+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>
| | * ASoC: ti: davinci-mcasp: Correct slot_width posed constraintPeter Ujfalusi2019-07-261-9/+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>
| | * ASoC: rockchip: Fix mono captureCheng-Yi Chiang2019-07-261-3/+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>
| | * ASoC: Intel: Fix some acpi vs apci typo in somme commentsChristophe JAILLET2019-07-2610-10/+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>
| | * ASoC: ti: davinci-mcasp: Fix clk PDIR handling for i2s master modePeter Ujfalusi2019-07-261-1/+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>
| | * ASoC: Fail card instantiation if DAI format setup failsRicard Wanderlof2019-07-241-2/+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>
| | * ASoC: SOF: Intel: hda: remove misleading error trace from IRQ threadKai Vehmanen2019-07-232-4/+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>
| | * ASoC: qcom: apq8016_sbc: Fix oops with multiple DAI linksStephan Gerhold2019-07-231-8/+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>
| | * ASoC: dapm: fix a memory leak bugWenwen Wang2019-07-221-0/+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>
| | * SoC: rockchip: rockchip_max98090: Enable MICBIAS for headset keypress detectionEnric Balletbo i Serra2019-07-221-0/+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>
| | * ASoC: cs42xx8: Fix MFREQ selection issue for async modeShengjiu Wang2019-07-221-19/+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>
| | * ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walksCharles Keepax2019-07-221-4/+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>
| | * ASoC: Intel: bytcht_es8316: Add quirk for Irbis NB41 netbookHans de Goede2019-07-161-0/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The Irbis NB41 netbook has its internal mic on IN2, inverted jack-detect and stereo speakers, add a quirk for this. Cc: russianneuromancer@ya.ru Reported-and-tested-by: russianneuromancer@ya.ru Signed-off-by: Hans de Goede <hdegoede@redhat.com> Link: https://lore.kernel.org/r/20190712112708.25327-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: samsung: odroid: fix a double-free issue for cpu_daiWen Yang2019-07-161-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The cpu_dai variable is still being used after the of_node_put() call, which may result in double-free: of_node_put(cpu_dai); ---> released here ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { ... goto err_put_clk_i2s; --> jump to err_put_clk_i2s ... err_put_clk_i2s: clk_put(priv->clk_i2s_bus); err_put_sclk: clk_put(priv->sclk_i2s); err_put_cpu_dai: of_node_put(cpu_dai); --> double-free here Fixes: d832d2b246c5 ("ASoC: samsung: odroid: Fix of_node refcount unbalance") Signed-off-by: Wen Yang <wen.yang99@zte.com.cn> Cc: Krzysztof Kozlowski <krzk@kernel.org> Cc: Sangbeom Kim <sbkim73@samsung.com> Cc: Sylwester Nawrocki <s.nawrocki@samsung.com> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Mark Brown <broonie@kernel.org> Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.com> Cc: alsa-devel@alsa-project.org Cc: linux-kernel@vger.kernel.org Link: https://lore.kernel.org/r/1562989575-33785-3-git-send-email-wen.yang99@zte.com.cn Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: samsung: odroid: fix an use-after-free issue for codecWen Yang2019-07-161-2/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The codec variable is still being used after the of_node_put() call, which may result in use-after-free. Fixes: bc3cf17b575a ("ASoC: samsung: odroid: Add support for secondary CPU DAI") Signed-off-by: Wen Yang <wen.yang99@zte.com.cn> Cc: Krzysztof Kozlowski <krzk@kernel.org> Cc: Sangbeom Kim <sbkim73@samsung.com> Cc: Sylwester Nawrocki <s.nawrocki@samsung.com> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Mark Brown <broonie@kernel.org> Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.com> Cc: alsa-devel@alsa-project.org Cc: linux-kernel@vger.kernel.org Link: https://lore.kernel.org/r/1562989575-33785-2-git-send-email-wen.yang99@zte.com.cn Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: audio-graph-card: add missing const at graph_get_dai_id()Kuninori Morimoto2019-07-111-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit c152f8491a8d9 ("ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id()") fixups use-after-free issue, but, it need to use "const" for reg. This patch adds it. We will have below without this patch LINUX/sound/soc/generic/audio-graph-card.c: In function 'graph_get_dai_id': LINUX/sound/soc/generic/audio-graph-card.c:87:7: warning: assignment discards\ 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] reg = of_get_property(node, "reg", NULL); Fixes: c152f8491a8d9 ("ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id()") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Wen Yang <wen.yang99@zte.com.cn> Link: https://lore.kernel.org/r/87sgrd43ja.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: rt1011: fix DC calibration offset not applyingShuming Fan2019-07-111-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There are two issues to fix: - DC offset calibration data will be reset after stopping playback. - DC offset calibration data should be applied in the initial setting. Signed-off-by: Shuming Fan <shumingf@realtek.com> Link: https://lore.kernel.org/r/20190711082214.8142-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id()Wen Yang2019-07-101-1/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | After calling of_node_put() on the node variable, it is still being used, which may result in use-after-free. Fix this issue by calling of_node_put() after the last usage. Fixes: a0c426fe1433 ("ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id()") Link: https://lore.kernel.org/r/1562743509-30496-5-git-send-email-wen.yang99@zte.com.cn Signed-off-by: Wen Yang <wen.yang99@zte.com.cn> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: audio-graph-card: fix use-after-free in graph_dai_link_of_dpcm()Wen Yang2019-07-101-13/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | After calling of_node_put() on the ports, port, and node variables, they are still being used, which may result in use-after-free. Fix this issue by calling of_node_put() after the last usage. Fixes: dd98fbc558a0 ("ASoC: audio-graph-card: cleanup DAI link loop method - step1") Link: https://lore.kernel.org/r/1562743509-30496-4-git-send-email-wen.yang99@zte.com.cn Signed-off-by: Wen Yang <wen.yang99@zte.com.cn> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: simple-card: fix an use-after-free in simple_for_each_link()Wen Yang2019-07-101-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The codec variable is still being used after the of_node_put() call, which may result in use-after-free. Fixes: d947cdfd4be2 ("ASoC: simple-card: cleanup DAI link loop method - step1") Link: https://lore.kernel.org/r/1562743509-30496-3-git-send-email-wen.yang99@zte.com.cn Signed-off-by: Wen Yang <wen.yang99@zte.com.cn> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
generated by cgit v1.2.3 (git 2.25.1) at 2024-12-03 03:55:14 +0000