- 22 Jun, 2019 1 commit
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Bard Liao authored
[ Upstream commit fa763f1b ] We observed the same issue as reported by commit a8d7bde2 ("ALSA: hda - Force polling mode on CFL for fixing codec communication") We don't have a better solution. So apply the same workaround to CNL. Signed-off-by:
Bard Liao <yung-chuan.liao@linux.intel.com> Signed-off-by:
Takashi Iwai <tiwai@suse.de> Signed-off-by:
Sasha Levin <sashal@kernel.org>
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- 19 Jun, 2019 8 commits
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Takashi Iwai authored
[ Upstream commit f0654ba9 ] This reverts commit feb68902. The fix attempt was incorrect, leading to the mutex deadlock through the close of OSS sequencer client. The proper fix needs more consideration, so let's revert it now. Fixes: feb68902 ("ALSA: seq: Protect in-kernel ioctl calls with mutex") Reported-by: syzbot+47ded6c0f23016cde310@syzkaller.appspotmail.com Signed-off-by:
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Takashi Iwai authored
[ Upstream commit 2eabc5ec ] The snd_seq_ioctl_get_subscription() retrieves the port subscriber information as a pointer, while the object isn't protected, hence it may be deleted before the actual reference. This race was spotted by syzkaller and may lead to a UAF. The fix is simply copying the data in the lookup function that performs in the rwsem to protect against the deletion. Reported-by: syzbot+9437020c82413d00222d@syzkaller.appspotmail.com Signed-off-by:
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Takashi Iwai authored
[ Upstream commit feb68902 ] ALSA OSS sequencer calls the ioctl function indirectly via snd_seq_kernel_client_ctl(). While we already applied the protection against races between the normal ioctls and writes via the client's ioctl_mutex, this code path was left untouched. And this seems to be the cause of still remaining some rare UAF as spontaneously triggered by syzkaller. For the sake of robustness, wrap the ioctl_mutex also for the call via snd_seq_kernel_client_ctl(), too. Reported-by: syzbot+e4c8abb920efa77bace9@syzkaller.appspotmail.com Signed-off-by:
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S.j. Wang authored
commit b06c58c2 upstream. When the output sample rate is [8kHz, 30kHz], the limitation of the supported ratio range is [1/24, 8]. In the driver we use (8kHz, 30kHz) instead of [8kHz, 30kHz]. So this patch is to fix this issue and the potential rounding issue with divider. Fixes: fff6e03c ("ASoC: fsl_asrc: add support for 8-30kHz output sample rate") Cc: <stable@vger.kernel.org> Signed-off-by:
Shengjiu Wang <shengjiu.wang@nxp.com> Acked-by:
Nicolin Chen <nicoleotsuka@gmail.com> Signed-off-by:
Mark Brown <broonie@kernel.org> Signed-off-by:
Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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S.j. Wang authored
commit ad6eecbf upstream. Add regcache_mark_dirty before regcache_sync for power of codec may be lost at suspend, then all the register need to be reconfigured. Fixes: 0c516b4f ("ASoC: cs42xx8: Add codec driver support for CS42448/CS42888") Cc: <stable@vger.kernel.org> Signed-off-by:
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Takashi Sakamoto authored
commit 0e3fb699 upstream. The data for isochronous resources is not destroyed in expected place. This commit fixes the bug. Cc: <stable@vger.kernel.org> # v4.12+ Fixes: 9b2bb4f2 ("ALSA: firewire-motu: add stream management functionality") Signed-off-by:
Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by:
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Kailang Yang authored
commit 717f43d8 upstream. ALC255 and ALC256 were some difference for hidden register. This update was suitable for ALC256. Fixes: e69e7e03 ("ALSA: hda/realtek - ALC256 speaker noise issue") Signed-off-by:
Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by:
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Takashi Sakamoto authored
commit d8fa87c3 upstream. Stanton SCS.1m can transfer isochronous packet with Multi Bit Linear Audio data channels, therefore it allows software to capture PCM substream. However, ALSA oxfw driver doesn't. This commit changes the driver to add one PCM substream for capture direction. Fixes: de5126cc ("ALSA: oxfw: add stream format quirk for SCS.1 models") Cc: <stable@vger.kernel.org> # v4.5+ Signed-off-by:
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- 15 Jun, 2019 2 commits
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Takashi Iwai authored
commit 7c32ae35 upstream. The call of unsubscribe_port() which manages the group count and module refcount from delete_and_unsubscribe_port() looks racy; it's not covered by the group list lock, and it's likely a cause of the reported unbalance at port deletion. Let's move the call inside the group list_mutex to plug the hole. Reported-by: syzbot+e4c8abb920efa77bace9@syzkaller.appspotmail.com Signed-off-by:
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Takashi Iwai authored
[ Upstream commit f495222e ] Currently the IRQ handler in HD-audio controller driver is registered before the chip initialization. That is, we have some window opened between the azx_acquire_irq() call and the CORB/RIRB setup. If an interrupt is triggered in this small window, the IRQ handler may access to the uninitialized RIRB buffer, which leads to a NULL dereference Oops. This is usually no big problem since most of Intel chips do register the IRQ via MSI, and we've already fixed the order of the IRQ enablement and the CORB/RIRB setup in the former commit b61749a8 ("sound: enable interrupt after dma buffer initialization"), hence the IRQ won't be triggered in that room. However, some platforms use a shared IRQ, and this may allow the IRQ trigger by another source. Another possibility is the kdump environment: a stale interrupt might be present in there, the IRQ handler can be falsely triggered as well. For covering this small race, let's move the azx_acquire_irq() call after hda_intel_init_chip() call. Although this is a bit radical change, it can cover more widely than checking the CORB/RIRB setup locally in the callee side. Reported-by:
Liwei Song <liwei.song@windriver.com> Signed-off-by:
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- 09 Jun, 2019 1 commit
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Kailang Yang authored
commit 317d9313 upstream. I measured power consumption between power_save_node=1 and power_save_node=0. It's almost the same. Codec will enter to runtime suspend and suspend. That pin also will enter to D3. Don't need to enter to D3 by single pin. So, Disable power_save_node as default. It will avoid more issues. Windows Driver also has not this option at runtime PM. Signed-off-by:
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- 31 May, 2019 6 commits
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Arnd Bergmann authored
[ Upstream commit 8ca51047 ] Building with clang shows a variable that is only used by the suspend/resume functions but defined outside of their #ifdef block: sound/soc/ti/davinci-mcasp.c:48:12: error: variable 'context_regs' is not needed and will not be emitted We commonly fix these by marking the PM functions as __maybe_unused, but here that would grow the davinci_mcasp structure, so instead add another #ifdef here. Fixes: 1cc0c054 ("ASoC: davinci-mcasp: Convert the context save/restore to use array") Signed-off-by:
Arnd Bergmann <arnd@arndb.de> Acked-by:
Peter Ujfalusi <peter.ujfalusi@ti.com> Reviewed-by:
Nathan Chancellor <natechancellor@gmail.com> Signed-off-by:
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Wen Yang authored
[ Upstream commit c7052471 ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/fsl_utils.c:74:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 38, but without a corresponding object release within this function. Signed-off-by:
Wen Yang <wen.yang99@zte.com.cn> Cc: Timur Tabi <timur@kernel.org> Cc: Nicolin Chen <nicoleotsuka@gmail.com> Cc: Xiubo Li <Xiubo.Lee@gmail.com> Cc: Fabio Estevam <festevam@gmail.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: linuxppc-dev@lists.ozlabs.org Cc: linux-kernel@vger.kernel.org Signed-off-by:
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Wen Yang authored
[ Upstream commit b820d52e ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/eukrea-tlv320.c:121:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. ./sound/soc/fsl/eukrea-tlv320.c:127:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. Signed-off-by:
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Daniel Baluta authored
[ Upstream commit ddb35114 ] is_slave_mode defaults to false because sai structure that contains it is kzalloc'ed. Anyhow, if we decide to set the following configuration SAI slave -> SAI master, is_slave_mode will remain set on true although SAI being master it should be set to false. Fix this by updating is_slave_mode for each call of fsl_sai_set_dai_fmt. Signed-off-by:
Daniel Baluta <daniel.baluta@nxp.com> Acked-by:
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Arnd Bergmann authored
[ Upstream commit ea751227 ] During randconfig builds, I occasionally run into an invalid configuration of the freescale FIQ sound support: WARNING: unmet direct dependencies detected for SND_SOC_IMX_PCM_FIQ Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=m] Selected by [y]: - SND_SOC_FSL_SPDIF [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=m]!=n && (MXC_TZIC [=n] || MXC_AVIC [=y]) sound/soc/fsl/imx-ssi.o: In function `imx_ssi_remove': imx-ssi.c:(.text+0x28): undefined reference to `imx_pcm_fiq_exit' sound/soc/fsl/imx-ssi.o: In function `imx_ssi_probe': imx-ssi.c:(.text+0xa64): undefined reference to `imx_pcm_fiq_init' The Kconfig warning is a result of the symbol being defined inside of the "if SND_IMX_SOC" block, and is otherwise harmless. The link error is more tricky and happens with SND_SOC_IMX_SSI=y, which may or may not imply FIQ support. However, if SND_SOC_FSL_SSI is set to =m at the same time, that selects SND_SOC_IMX_PCM_FIQ as a loadable module dependency, which then causes a link failure from imx-ssi. The solution here is to make SND_SOC_IMX_PCM_FIQ built-in whenever one of its potential users is built-in. Fixes: ff40260f ("ASoC: fsl: refine DMA/FIQ dependencies") Signed-off-by:
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Jerome Brunet authored
[ Upstream commit 30180e84 ] If the hdmi codec startup fails, it should clear the current_substream pointer to free the device. This is properly done for the audio_startup() callback but for snd_pcm_hw_constraint_eld(). Make sure the pointer cleared if an error is reported. Signed-off-by:
Jerome Brunet <jbrunet@baylibre.com> Signed-off-by:
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- 21 May, 2019 7 commits
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Michał Wadowski authored
commit 56df90b6 upstream. Add patch for realtek codec in Lenovo B50-70 that fixes inverted internal microphone channel. Device IdeaPad Y410P has the same PCI SSID as Lenovo B50-70, but first one is about fix the noise and it didn't seem help in a later kernel version. So I replaced IdeaPad Y410P device description with B50-70 and apply inverted microphone fix. Bugzilla: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1524215 Signed-off-by:
Michał Wadowski <wadosm@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by:
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Curtis Malainey authored
commit a46eb523 upstream. The current algorithm allows 3 types of transfers, 16bit, 32bit and burst. According to Realtek, 16bit transfers have a special restriction in that it is restricted to the memory region of 0x18020000 ~ 0x18021000. This region is the memory location of the I2C registers. The current algorithm does not uphold this restriction and therefore fails to complete writes. Since this has been broken for some time it likely no one is using it. Better to simply disable the 16 bit writes. This will allow users to properly load firmware over SPI without data corruption. Signed-off-by:
Curtis Malainey <cujomalainey@chromium.org> Reviewed-by:
Ben Zhang <benzh@chromium.org> Signed-off-by:
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Jon Hunter authored
commit ecb2795c upstream. The max98090 driver defines 3 DAPM muxes; one for the right line output (LINMOD Mux), one for the left headphone mixer source (MIXHPLSEL Mux) and one for the right headphone mixer source (MIXHPRSEL Mux). The same bit is used for the mux as well as the DAPM enable, and although the mux can be correctly configured, after playback has completed, the mux will be reset during the disable phase. This is preventing the state of these muxes from being saved and restored correctly on system reboot. Fix this by marking these muxes as SND_SOC_NOPM. Note this has been verified this on the Tegra124 Nyan Big which features the MAX98090 codec. Signed-off-by:
Jon Hunter <jonathanh@nvidia.com> Signed-off-by:
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Kailang Yang authored
commit 607ca3bd upstream. Let EAPD turn on after set pin output. [ NOTE: This change is supposed to reduce the possible click noises at (runtime) PM resume. The functionality should be same (i.e. the verbs are executed correctly) no matter which order is, so this should be safe to apply for all codecs -- tiwai ] Signed-off-by:
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Hui Wang authored
commit 7f641e26 upstream. On the machines with AMD GPU or Nvidia GPU, we often meet this issue: after s3, there are 4 HDMI/DP audio devices in the gnome-sound-setting even there is no any monitors plugged. When this problem happens, we check the /proc/asound/cardX/eld#N.M, we will find the monitor_present=1, eld_valid=0. The root cause is BIOS or GPU driver makes the PRESENCE valid even no monitor plugged, and of course the driver will not get the valid eld_data subsequently. In this situation, we should not report the jack_plugged event, to do so, let us change the function hdmi_present_sense_via_verbs(). In this function, it reads the pin_sense via snd_hda_pin_sense(), after calling this function, the jack_dirty is 0, and before exiting via_verbs(), we change the shadow pin_sense according to both monitor_present and eld_valid, then in the snd_hda_jack_report_sync(), since the jack_dirty is still 0, it will report jack event according to this modified shadow pin_sense. After this change, the driver will not report Jack_is_plugged event through hdmi_present_sense_via_verbs() if monitor_present is 1 and eld_valid is 0. Signed-off-by:
Hui Wang <hui.wang@canonical.com> Cc: <stable@vger.kernel.org> Signed-off-by:
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Hui Wang authored
commit 8c2e6728 upstream. The driver will check the monitor presence when resuming from suspend, starting poll or interrupt triggers. In these 3 situations, the jack_dirty will be set to 1 first, then the hda_jack.c reads the pin_sense from register, after reading the register, the jack_dirty will be set to 0. But hdmi_repoll_work() is enabled in these 3 situations, It will read the pin_sense a couple of times subsequently, since the jack_dirty is 0 now, It does not read the register anymore, instead it uses the shadow pin_sense which is read at the first time. It is meaningless to check the shadow pin_sense a couple of times, we need to read the register to check the real plugging state, so we set the jack_dirty to 1 in the hdmi_repoll_work(). Signed-off-by:
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Wenwen Wang authored
commit cb517359 upstream. In parse_audio_selector_unit(), the string array 'namelist' is allocated through kmalloc_array(), and each string pointer in this array, i.e., 'namelist[]', is allocated through kmalloc() in the following for loop. Then, a control instance 'kctl' is created by invoking snd_ctl_new1(). If an error occurs during the creation process, the string array 'namelist', including all string pointers in the array 'namelist[]', should be freed, before the error code ENOMEM is returned. However, the current code does not free 'namelist[]', resulting in memory leaks. To fix the above issue, free all string pointers 'namelist[]' in a loop. Signed-off-by:
Wenwen Wang <wang6495@umn.edu> Cc: <stable@vger.kernel.org> Signed-off-by:
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- 10 May, 2019 11 commits
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Ross Zwisler authored
commit 0efa3334 upstream. Currently in sst_dsp_new() if we get an error return from sst_dma_new() we just print an error message and then still complete the function successfully. This means that we are trying to run without sst->dma properly set up, which will result in NULL pointer dereference when sst->dma is later used. This was happening for me in sst_dsp_dma_get_channel(): struct sst_dma *dma = dsp->dma; ... dma->ch = dma_request_channel(mask, dma_chan_filter, dsp); This resulted in: BUG: unable to handle kernel NULL pointer dereference at 0000000000000018 IP: sst_dsp_dma_get_channel+0x4f/0x125 [snd_soc_sst_firmware] Fix this by adding proper error handling for the case where we fail to set up DMA. This change only affects Haswell and Broadwell systems. Baytrail systems explicilty opt-out of DMA via sst->pdata->resindex_dma_base being set to -1. Signed-off-by:
Ross Zwisler <zwisler@google.com> Cc: stable@vger.kernel.org Acked-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by:
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Tzung-Bi Shih authored
[ Upstream commit d6ba3f81 ] Fix wrong setting on number of channels. The context wants to set constraint to 2 channels instead of 4. Signed-off-by:
Tzung-Bi Shih <tzungbi@google.com> Acked-by:
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Sugar Zhang authored
[ Upstream commit c8506443 ] This is because set_fmt ops maybe called when PD is off, and in such case, regmap_ops will lead system hang. enale PD before doing regmap_ops. Signed-off-by:
Sugar Zhang <sugar.zhang@rock-chips.com> Signed-off-by:
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Annaliese McDermond authored
[ Upstream commit c63adb28 ] The common pins were mistakenly not added to the DAPM graph. Adding these pins will allow valid graphs to be created. Signed-off-by:
Annaliese McDermond <nh6z@nh6z.net> Signed-off-by:
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Daniel Mack authored
[ Upstream commit f0f2338a ] The CS4270 does not by default increment the register address on consecutive writes. During normal operation it doesn't matter as all register accesses are done individually. At resume time after suspend, however, the regcache code gathers the biggest possible block of registers to sync and sends them one on one go. To fix this, set the INCR bit in all cases. Signed-off-by:
Daniel Mack <daniel@zonque.org> Signed-off-by:
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Charles Keepax authored
[ Upstream commit a2225a6d ] Best to lock across handling the bus error to ensure the DSP doesn't change power state as we are reading the status registers. Signed-off-by:
Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by:
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Sylwester Nawrocki authored
[ Upstream commit 2b13bee3 ] After commit fbeec965 ("ASoC: samsung: odroid: Fix 32000 sample rate handling") the audio root clock frequency is configured improperly for 44100 sample rate. Due to clock rate rounding it's 20070401 Hz instead of 22579000 Hz. This results in a too low value of the PSR clock divider in the CPU DAI driver and too fast actual sample rate for fs=44100. E.g. 1 kHz tone has actual 1780 Hz frequency (1 kHz * 20070401/22579000 * 2). Fix this by increasing the correction passed to clk_set_rate() to take into account inaccuracy of the EPLL frequency properly. Fixes: fbeec965 ("ASoC: samsung: odroid: Fix 32000 sample rate handling") Reported-by:
JaeChul Lee <jcsing.lee@samsung.com> Signed-off-by:
Sylwester Nawrocki <s.nawrocki@samsung.com> Signed-off-by:
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John Hsu authored
[ Upstream commit 54d1cf78 ] The driver changes the stream name of DAC and ADC to avoid the issue of widget with prefixed name. When the machine adds prefixed name for codec, the stream name of DAI may not find the widgets. Signed-off-by:
John Hsu <KCHSU0@nuvoton.com> Signed-off-by:
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John Hsu authored
[ Upstream commit 844a4a36 ] The driver has two issues when machine add prefix name for codec. (1)The stream name of DAI can't find the AIF widgets. (2)The drivr can enable/disalbe the MICBIAS and SAR widgets. The patch will fix these issues caused by prefixed name added. Signed-off-by:
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Rander Wang authored
[ Upstream commit 570f18b6 ] On HDaudio platforms, if playback is started when capture is working, there is no audible output. This can be root-caused to the use of the rx|tx_mask to store an HDaudio stream tag. If capture is stared before playback, rx_mask would be non-zero on HDaudio platform, then the channel number of playback, which is in the same codec dai with the capture, would be changed by soc_pcm_codec_params_fixup based on the tx_mask at first, then overwritten by this function based on rx_mask at last. According to the author of tx|rx_mask, tx_mask is for playback and rx_mask is for capture. And stream direction is checked at all other references of tx|rx_mask in ASoC, so here should be an error. This patch checks stream direction for tx|rx_mask for fixup function. This issue would affect not only HDaudio+ASoC, but also I2S codecs if the channel number based on rx_mask is not equal to the one for tx_mask. It could be rarely reproduecd because most drivers in kernel set the same channel number to tx|rx_mask or rx_mask is zero. Tested on all platforms using stream_tag & HDaudio and intel I2S platforms. Signed-off-by:
Rander Wang <rander.wang@linux.intel.com> Acked-by:
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Russell King authored
[ Upstream commit 2e95f984 ] When using the S/PDIF DAI, there is no requirement to call snd_soc_dai_set_fmt() as there is no DAI format definition that defines S/PDIF. In any case, S/PDIF does not have separate clocks, this is embedded into the data stream. Consequently, when attempting to use TDA998x in S/PDIF mode, the attempt to configure TDA998x via the hw_params callback fails as the hdmi_codec_daifmt is left initialised to zero. Since the S/PDIF DAI will only be used by S/PDIF, prepare the hdmi_codec_daifmt structure for this format. Signed-off-by:
Russell King <rmk+kernel@armlinux.org.uk> Reviewed-by:
Jyri Sarha <jsarha@ti.com> Signed-off-by:
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- 08 May, 2019 4 commits
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Arnaud Pouliquen authored
commit 17d3069c upstream. This patch fixes the sai driver structure overwriting which results in a cpu dai name equal NULL. Fixes: 3e086edf ("ASoC: stm32: add SAI driver") Signed-off-by:
Arnaud Pouliquen <arnaud.pouliquen@st.com> Signed-off-by:
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Kailang Yang authored
commit 0700d3d1 upstream. Fixed Dell AIO speaker noise. spec->gen.auto_mute_via_amp = 1, this option was solved speaker white noise at boot. codec->power_save_node = 0, this option was solved speaker noise at resume back. Fixes: 92266651 ("ALSA: hda/realtek - Fix Dell AIO LineOut issue") Signed-off-by:
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Kailang Yang authored
commit 0a29c57b upstream. Add two Dell platform for headset mode. [ Note: this is a further correction / addition of the previous pin-based quirks for Dell machines; another entry for ALC236 with the d-mic pin 0x12 and an entry for ALC295 -- tiwai ] Fixes: b26e36b7 ("ALSA: hda/realtek - add two more pin configuration sets to quirk table") Signed-off-by:
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Greg Kroah-Hartman authored
commit e5c812e8 upstream. The line6 driver uses a lot of USB buffers off of the stack, which is not allowed on many systems, causing the driver to crash on some of them. Fix this up by dynamically allocating the buffers with kmalloc() which allows for proper DMA-able memory. Reported-by:
Christo Gouws <gouws.christo@gmail.com> Reported-by:
Alan Stern <stern@rowland.harvard.edu> Tested-by:
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