- 16 8月, 2019 1 次提交
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由 Hui Peng 提交于
`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: NHui Peng <benquike@gmail.com> Reported-by: NMathias Payer <mathias.payer@nebelwelt.net> Signed-off-by: NHui Peng <benquike@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 15 8月, 2019 1 次提交
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由 Hui Peng 提交于
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: NHui Peng <benquike@gmail.com> Reported-by: NMathias Payer <mathias.payer@nebelwelt.net> Cc: <stable@vger.kernel.org> Signed-off-by: NHui Peng <benquike@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 14 8月, 2019 2 次提交
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由 Hui Wang 提交于
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: NHui Wang <hui.wang@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Hui Wang 提交于
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: NHui Wang <hui.wang@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 13 8月, 2019 1 次提交
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由 Takashi Iwai 提交于
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: NTakashi Iwai <tiwai@suse.de>
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- 10 8月, 2019 1 次提交
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由 Wenwen Wang 提交于
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: 352f7f91 ("ALSA: hda - Merge Realtek parser code to generic parser") Signed-off-by: NWenwen Wang <wenwen@cs.uga.edu> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 09 8月, 2019 1 次提交
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由 Takashi Iwai 提交于
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: NTakashi Iwai <tiwai@suse.de>
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- 08 8月, 2019 2 次提交
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由 Wenwen Wang 提交于
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: 31ef9134 ("ALSA: add LaCie FireWire Speakers/Griffin FireWave Surround driver") Signed-off-by: NWenwen Wang <wenwen@cs.uga.edu> Reviewed-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp> Cc: <stable@vger.kernel.org> # v2.6.39+ Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Wenwen Wang 提交于
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: NWenwen Wang <wenwen@cs.uga.edu> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 07 8月, 2019 2 次提交
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由 Takashi Iwai 提交于
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: NTakashi Iwai <tiwai@suse.de>
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由 Wenwen Wang 提交于
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: a91c3fb2 ("Add M2Tech hiFace USB-SPDIF driver") Signed-off-by: NWenwen Wang <wenwen@cs.uga.edu> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 06 8月, 2019 5 次提交
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由 Takashi Iwai 提交于
The commit bfcba288 ("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: bfcba288 ("ALSA - hda: Add support for link audio time reporting") Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Wenwen Wang 提交于
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: 7edf3b5e ("ALSA: usb-audio: AudioStreaming Power Domain parsing") Signed-off-by: NWenwen Wang <wenwen@cs.uga.edu> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Merge tag 'asoc-fix-v5.3-rc3-2' of 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.
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由 Takashi Iwai 提交于
Merge tag 'asoc-fix-v5.3-rc3' of 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.
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由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@kernel.org>
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- 03 8月, 2019 2 次提交
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由 Vijendar Mukunda 提交于
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: NVijendar Mukunda <vijendar.mukunda@amd.com> Link: https://lore.kernel.org/r/1564753899-17124-2-git-send-email-Vijendar.Mukunda@amd.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Vijendar Mukunda 提交于
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: NVijendar Mukunda <vijendar.mukunda@amd.com> Link: https://lore.kernel.org/r/1564753899-17124-1-git-send-email-Vijendar.Mukunda@amd.comSigned-off-by: NMark Brown <broonie@kernel.org>
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- 31 7月, 2019 3 次提交
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由 fengchunguo 提交于
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: Nfengchunguo <chunguo.feng@amlogic.com> Link: https://lore.kernel.org/r/20190731074156.5620-1-chunguo.feng@amlogic.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Marcus Cooper 提交于
The A64 audio codec uses the original I2S block but the SR and WSS computation currently assigned is for the newer block. Fixes: 619c15f7 (ASoC: sun4i-i2s: Change SR and WSS computation) Signed-off-by: NMarcus Cooper <codekipper@gmail.com> Link: https://lore.kernel.org/r/20190729152130.27955-1-codekipper@gmail.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Cezary Rojewski 提交于
Adding myself to Intel ASoC drivers maintainers list. Signed-off-by: NCezary Rojewski <cezary.rojewski@intel.com> Acked-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20190726181517.27655-1-cezary.rojewski@intel.comSigned-off-by: NMark Brown <broonie@kernel.org>
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- 30 7月, 2019 2 次提交
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由 Hillf Danton 提交于
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 801ebf10 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=d59c4387bfb6eced94e2Reported-by: Nsyzbot <syzbot+d59c4387bfb6eced94e2@syzkaller.appspotmail.com> Fixes: 801ebf10 ("ALSA: usb-audio: Sanity checks for each pipe and EP types") Cc: Andrey Konovalov <andreyknvl@google.com> Signed-off-by: NHillf Danton <hdanton@sina.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Yuki Tsunashima 提交于
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: f2b3614c ("ALSA: PCM - Don't check DMA time-out too shortly") Signed-off-by: NYuki Tsunashima <ytsunashima@jp.adit-jv.com> Signed-off-by: NSuresh Udipi <sudipi@jp.adit-jv.com> [ported from 4.9] Signed-off-by: NAdam Miartus <amiartus@de.adit-jv.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 27 7月, 2019 1 次提交
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由 Samuel Thibault 提交于
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: f9b54e19 ("ALSA: hda/i915: Allow delayed i915 audio component binding") Signed-off-by: NSamuel Thibault <samuel.thibault@ens-lyon.org> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 26 7月, 2019 4 次提交
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由 Peter Ujfalusi 提交于
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: NPeter Ujfalusi <peter.ujfalusi@ti.com> Link: https://lore.kernel.org/r/20190726064244.3762-2-peter.ujfalusi@ti.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Cheng-Yi Chiang 提交于
This reverts commit db51707b. 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: NCheng-Yi Chiang <cychiang@chromium.org> Link: https://lore.kernel.org/r/20190726044202.26866-1-cychiang@chromium.orgSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Christophe JAILLET 提交于
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: NChristophe JAILLET <christophe.jaillet@wanadoo.fr> Link: https://lore.kernel.org/r/20190725053523.16542-1-christophe.jaillet@wanadoo.frSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Peter Ujfalusi 提交于
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: ca3d9433 ("ASoC: davinci-mcasp: Update PDIR (pin direction) register handling") Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com> Link: https://lore.kernel.org/r/20190725083423.7321-1-peter.ujfalusi@ti.comSigned-off-by: NMark Brown <broonie@kernel.org>
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- 25 7月, 2019 3 次提交
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由 Hui Wang 提交于
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: NHui Wang <hui.wang@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
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=201901Reported-and-tested-by: NTodd Brandt <todd.e.brandt@linux.intel.com> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Ricard Wanderlof 提交于
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: NRicard Wanderlof <ricardw@axis.com> Link: https://lore.kernel.org/r/alpine.DEB.2.20.1907241132350.6338@lnxricardw1.se.axis.comSigned-off-by: NMark Brown <broonie@kernel.org>
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- 23 7月, 2019 8 次提交
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由 Ding Xiang 提交于
put_device will call ac97_codec_release to free ac97_codec_device and other resources, so remove the kfree and other redundant code. Fixes: 74426fbf ("ALSA: ac97: add an ac97 bus") Signed-off-by: NDing Xiang <dingxiang@cmss.chinamobile.com> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Kai Vehmanen 提交于
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: NKai Vehmanen <kai.vehmanen@linux.intel.com> Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20190722141402.7194-21-pierre-louis.bossart@linux.intel.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Stephan Gerhold 提交于
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: 98b232ca ("ASoC: qcom: apq8016_sbc: use modern dai_link style") Signed-off-by: NStephan Gerhold <stephan@gerhold.net> Acked-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/20190722130352.95874-1-stephan@gerhold.netSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Charles Keepax 提交于
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: NCharles Keepax <ckeepax@opensource.cirrus.com> Acked-by: NVinod Koul <vkoul@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Charles Keepax 提交于
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: NCharles Keepax <ckeepax@opensource.cirrus.com> Acked-by: NVinod Koul <vkoul@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Charles Keepax 提交于
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: NCharles Keepax <ckeepax@opensource.cirrus.com> Acked-by: NVinod Koul <vkoul@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Charles Keepax 提交于
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: 4f2ab5e1 ("ALSA: compress: Fix stop handling on compressed capture streams") Signed-off-by: NCharles Keepax <ckeepax@opensource.cirrus.com> Acked-by: NVinod Koul <vkoul@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Wenwen Wang 提交于
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: NWenwen Wang <wenwen@cs.uga.edu> Link: https://lore.kernel.org/r/1563803864-2809-1-git-send-email-wang6495@umn.eduSigned-off-by: NMark Brown <broonie@kernel.org>
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- 22 7月, 2019 1 次提交
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由 Masahiro Yamada 提交于
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: NMasahiro Yamada <yamada.masahiro@socionext.com> Link: https://lore.kernel.org/r/20190721142308.30306-1-yamada.masahiro@socionext.comSigned-off-by: NMark Brown <broonie@kernel.org>
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