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2019-04-05ASoC: intel: skylake: add remove() callback for component driverRanjani Sridharan
Topology is not unloaded in the core during unregister_component() anymore. So, add the remove() callback that will unload the topology. Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-04-05ASoC: cs35l35: Disable regulators on driver removalCharles Keepax
The chips main power supplies VA and VP are enabled during probe but then never disabled, this will cause warnings from the regulator framework on driver removal. Fix this by adding a remove callback and disabling the supplies, whilst doing so follow best practice and put the chip back into reset as well. Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-04-04ALSA: xen-front: Do not use stream buffer size before it is setOleksandr Andrushchenko
This fixes the regression introduced while moving to Xen shared buffer implementation. Fixes: 58f9d806d16a ("ALSA: xen-front: Use Xen common shared buffer implementation") Reviewed-by: Juergen Gross <jgross@suse.com> Signed-off-by: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> Cc: <stable@vger.kernel.org> # v5.0+ Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-04-04ASoC: rockchip: pdm: change dma burst to 8Sugar Zhang
This patch decreases the transfer bursts to avoid the fifo overrun. Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-04-04ASoC: rockchip: pdm: fix regmap_ops hang issueSugar Zhang
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: Mark Brown <broonie@kernel.org>
2019-04-04ASoC: simple-card: don't select DPCM via simple-audio-cardKuninori Morimoto
commit da215354eb55c ("ASoC: simple-card: merge simple-scu-card") merged simple-scu-audio-card which can handle DPCM into simple-audio-card. By this patch, the judgement to select "normal sound card" or "DPCM sound card" is based on its CPU/Codec DAI count. But, because of it, existing "simple-audio-card" user who is assuming "normal sound card" might select DPCM unintentionally. To solve this issue, this patch allows "simple-audio-card" user can select "normal sound card", and "simple-scu-audio-card" user can select both "normal sound card" and "DPCM sound card". This keeps compatibility collectry. Fixes: da215354eb55c ("ASoC: simple-card: merge simple-scu-card") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-04-04ASoC: audio-graph-card: don't select DPCM via audio-graph-cardKuninori Morimoto
commit ae3cb5790906b ("ASoC: audio-graph-card: merge audio-graph-scu-card") merged audio-graph-scu-card which can handle DPCM into audio-graph-card. By this patch, the judgement to select "normal sound card" or "DPCM sound card" is based on its OF-graph endpoint connection. But, because of it, existing "audio-graph-card" user who is assuming "normal sound card" might select DPCM unintentionally. To solve this issue, this patch allows "audio-graph-card" user can select "normal sound card", and "audio-graph-scu-card" user can select both "normal sound card" and "DPCM sound card". This keeps compatibility collectry. Fixes: ae3cb5790906b ("ASoC: audio-graph-card: merge audio-graph-scu-card") Reported-by: Arnaud Pouliquen <arnaud.pouliquen@st.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Arnaud Pouliquen <arnaud.pouliquen@st.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-04-04ASoC: tlv320aic32x4: Change author's nameAnnaliese McDermond
The author of these files has changed her name. Update instances in the code of her dead name to current legal name. Signed-off-by: Annaliese McDermond <nh6z@nh6z.net> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-04-03ALSA: doc: my_chip has no element ioportChristina Quast
chip->ioport is dereferenced in two places, but the struct is defined as follows: struct mychip { struct snd_card *card; struct pci_dev *pci; unsigned long port; int irq; }; Signed-off-by: Christina Quast <cquast@hanoverdisplays.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-04-03ALSA: hda/realtek - Add quirk for Tuxedo XC 1509Richard Sailer
This adds a SND_PCI_QUIRK(...) line for the Tuxedo XC 1509. The Tuxedo XC 1509 and the System76 oryp5 are the same barebone notebooks manufactured by Clevo. To name the fixups both use after the actual underlying hardware, this patch also changes System76_orpy5 to clevo_pb51ed in 2 enum symbols and one function name, matching the other pci_quirk entries which are also named after the device ODM. Fixes: 7f665b1c3283 ("ALSA: hda/realtek - Headset microphone and internal speaker support for System76 oryp5") Signed-off-by: Richard Sailer <rs@tuxedocomputers.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-04-03ALSA: hda/realtek - Move to ACT_INIT stateKailang Yang
It will be lose Mic JD state when Chrome OS boot and headset was plugged. Just Implement of reset combo jack JD verb for ACT_PRE_PROBE state. Intel test result was also failed. It test passed until changed the initial state to ACT_INIT. Mic JD will show every time. This patch also changed the model name as 'alc-chrome-book' for application of Chrome OS. Fixes: 10f5b1b85ed1 ("ALSA: hda/realtek - Fixed Headset Mic JD not stable") Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-04-03ASoC: Intel: cht_bsw_max98090_ti: Enable codec clock once and keep it enabledHans de Goede
Users have been seeing sound stability issues with max98090 codecs since: commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") At first that commit broke sound for Chromebook Swanky and Clapper models, the problem was that the machine-driver has been controlling the wrong clock on those models since support for them was added. This was hidden by clk-pmc-atom.c keeping the actual clk on unconditionally. With the machine-driver controlling the proper clock, sound works again but we are seeing bug reports describing it as: low volume, "sounds like played at 10x speed" and instable. When these issues are hit the following message is seen in dmesg: "max98090 i2c-193C9890:00: PLL unlocked". Attempts have been made to fix this by inserting a delay between enabling the clk and enabling and checking the pll, but this has not helped. It seems that at least on boards which use pmc_plt_clk_0 as clock, if we ever disable the clk, the pll looses its lock and after that we get various issues. This commit fixes this by enabling the clock once at probe time on these boards. In essence this restores the old behavior of clk-pmc-atom.c always keeping the clk on on these boards. Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Reported-by: Mogens Jensen <mogens-jensen@protonmail.com> Reported-by: Dean Wallace <duffydack73@gmail.com> Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-04-03ASoC: wm_adsp: Check for buffer in trigger stopCharles Keepax
Trigger stop can be called in situations where trigger start failed and as such it can't be assumed the buffer is already attached to the compressed stream or a NULL pointer may be dereferenced. Fixes: 639e5eb3c7d6 ("ASoC: wm_adsp: Correct handling of compressed streams that restart") Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-04-02ALSA: uapi: #include <time.h> in asound.hDaniel Mentz
The uapi header asound.h defines types based on struct timespec. We need to #include <time.h> to get access to the definition of this struct. Previously, we encountered the following error message when building applications with a clang/bionic toolchain: kernel-headers/sound/asound.h:350:19: error: field has incomplete type 'struct timespec' struct timespec trigger_tstamp; ^ The absence of the time.h #include statement does not cause build errors with glibc, because its version of stdlib.h indirectly includes time.h. Signed-off-by: Daniel Mentz <danielmentz@google.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-04-02ALSA: hda/realtek: Enable headset MIC of Acer TravelMate B114-21 with ALC233Jian-Hong Pan
The Acer TravelMate B114-21 laptop cannot detect and record sound from headset MIC. This patch adds the ALC233_FIXUP_ACER_HEADSET_MIC HDA verb quirk chained with ALC233_FIXUP_ASUS_MIC_NO_PRESENCE pin quirk to fix this issue. [ fixed the missing brace and reordered the entry -- tiwai ] Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Signed-off-by: Daniel Drake <drake@endlessm.com> Reviewed-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-04-02ASoC: dapm: set power_check callback for widgets that shouldnt be always onRanjani Sridharan
Currently, buffers, schedulers, src's, encoders, decoders and effect type dapm widgets remain always on as their power_check method is not set. Setting this callback allows these widgets in the audio path to be powered managed properly. Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-04-02ASoC: dpcm: skip missing substream while applying symmetryJerome Brunet
If for any reason, the backend does not have the requested substream (like capture on a playback only backend), the BE will be skipped in dpcm_be_dai_startup(). However, dpcm_apply_symmetry() does not skip those BE and will dereference the be_substream (NULL) pointer anyway. Like in dpcm_be_dai_startup(), just skip those BE. Fixes: 906c7d690c3b ("ASoC: dpcm: Apply symmetry for DPCM") Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-04-01ASoC: tlv320aic32x4: Fix Common PinsAnnaliese McDermond
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: Mark Brown <broonie@kernel.org>
2019-03-28Merge branch 'topic/timer-fixes' into for-nextTakashi Iwai
Pull yet another ALSA core timer fixes and cleanups. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-27ALSA: us122l: Use alloc_pages_exact()Takashi Iwai
alloc_pages_exact() is more suitable choice for allocating the sound buffers, as it doesn't need to align with power-of-two. Along with the conversion, we can drop __GFP_COMP as well. The patch also replace the error messages to be more explicit. Acked-by: Michal Hocko <mhocko@suse.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-27ALSA: Replace snd_malloc_pages() and snd_free_pages() with standard helpers, ↵Takashi Iwai
take#2 snd_malloc_pages() and snd_free_pages() are merely thin wrappers of the standard page allocator / free functions. Even the arguments are compatible with some standard helpers, so there is little merit of keeping these wrappers. This patch replaces the all existing callers of snd_malloc_pages() and snd_free_pages() with the direct calls of the standard helper functions. In this version, we use a recently introduced one, alloc_pages_exact(), which suits better than the old snd_malloc_pages() implementation for our purposes. Then we can avoid the waste of pages by alignment to power-of-two. Since alloc_pages_exact() does split pages, we need no longer __GFP_COMP flag; or better to say, we must not pass __GFP_COMP to alloc_pages_exact(). So the former unconditional addition of __GFP_COMP flag in snd_malloc_pages() is dropped, as well as in most other places. Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Michal Hocko <mhocko@suse.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-27ALSA: timer: Make snd_timer_close() really kill pending actionsTakashi Iwai
snd_timer_close() is supposed to close the timer instance and sync with the deactivation of pending actions. However, there are still some overlooked cases: - It calls snd_timer_stop() at the beginning, but some other might re-trigger the timer right after that. - snd_timer_stop() calls del_timer_sync() only when all belonging instances are closed. If multiple instances were assigned to a timer object and one is closed, the timer is still running. Then the pending action assigned to this timer might be left. Actually either of the above is the likely cause of the reported syzkaller UAF. This patch plug these holes by introducing SNDRV_TIMER_IFLG_DEAD flag. This is set at the beginning of snd_timer_close(), and the flag is checked at snd_timer_start*() and else, so that no longer new action is left after snd_timer_close(). Reported-by: syzbot+d5136d4d3240cbe45a2a@syzkaller.appspotmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-27ALSA: timer: Check ack_list emptiness instead of bit flagTakashi Iwai
For checking the pending timer instance that is still left on the timer object that is being closed, we set/clear a bit flag SNDRV_TIMER_IFLG_CALLBACK around the call of callbacks. This can be simplified by replace with the list_empty() call for ti->ack_list. This covers the existence more comprehensively and safely. A gratis bonus is that we can get rid of SNDRV_TIMER_IFLG_CALLBACK bit flag definition as well. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-27ALSA: timer: Make sure to clear pending ack listTakashi Iwai
When a card is under disconnection, we bail out immediately at each timer interrupt or tasklet. This might leave some items left in ack list. For a better integration of the upcoming change to check ack_list emptiness, clear out the whole list upon the emergency exit route. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-27ALSA: timer: Unify timer callback process codeTakashi Iwai
The timer core has two almost identical code for processing callbacks: once in snd_timer_interrupt() for fast callbacks and another in snd_timer_tasklet() for delayed callbacks. Let's unify them. In the new version, the resolution is read from ti->resolution at each call, and this must be fine; ti->resolution is set in the preparation step in snd_timer_interrupt(). Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-26ALSA: emux: Add support of loading GUS-patchTakashi Iwai
It's a feature request for the ancient sutff, but it's still valid; the loading of a GUS-patch isn't available via hwdep device although it's supported over OSS sequencer. The only missing piece is the call of snd_soundfont_load_guspatch() in synth emux hwdep code. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-26ALSA: hda/realtek - Fix speakers on Acer Predator Helios 500 Ryzen laptopsBernhard Rosenkraenzer
On an Acer Predator Helios 500 (Ryzen version), the laptop's speakers don't work out of the box. The problem can be worked around with hdajackretask, remapping the "Black Headphone, Right side" pin (0x21) to the Internal speaker. This patch adds a quirk to change this mapping by default. [ corrected ALC299_FIXUP_PREDATOR_SPK definition and adapted for the latest tree by tiwai ] Signed-off-by: Bernhard Rosenkraenzer <bero@lindev.ch> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-26ASoC: Intel: Skylake: enable S24_LE format supportJenny TC
To enable S24_LE format, sample_type in topology fw has to be set to 1. But sample_type defined in topology firmware configuration is not getting reflected in the dsp param. This patch sets sample_type in base config so that the sample type defined in the topology firmware is reflected in the dsp params. This issues was uncovered while debugging the S24_LE format which require the MSB byte in 32 bit word to be skipped. Setting sample_type in topology firmware to 1 helps to skip MSB byte word. Signed-off-by: Jenny TC <jenny.tc@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-03-25ALSA: aloop: Support S24 sample formatsTimo Wischer
Currently snd_aloop supports only S16 and S32 audio sample formats. With this patch the S24 formats are also supported. Signed-off-by: Timo Wischer <twischer@de.adit-jv.com> Reviewed-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-25ALSA: pcm: Don't suspend stream in unrecoverable PCM stateTakashi Iwai
Currently PCM core sets each opened stream forcibly to SUSPENDED state via snd_pcm_suspend_all() call, and the user-space is responsible for re-triggering the resume manually either via snd_pcm_resume() or prepare call. The scheme works fine usually, but there are corner cases where the stream can't be resumed by that call: the streams still in OPEN state before finishing hw_params. When they are suspended, user-space cannot perform resume or prepare because they haven't been set up yet. The only possible recovery is to re-open the device, which isn't nice at all. Similarly, when a stream is in DISCONNECTED state, it makes no sense to change it to SUSPENDED state. Ditto for in SETUP state; which you can re-prepare directly. So, this patch addresses these issues by filtering the PCM streams to be suspended by checking the PCM state. When a stream is in either OPEN, SETUP or DISCONNECTED as well as already SUSPENDED, the suspend action is skipped. To be noted, this problem was originally reported for the PCM runtime PM on HD-audio. And, the runtime PM problem itself was already addressed (although not intended) by the code refactoring commits 3d21ef0b49f8 ("ALSA: pcm: Suspend streams globally via device type PM ops") and 17bc4815de58 ("ALSA: pci: Remove superfluous snd_pcm_suspend*() calls"). These commits eliminated the snd_pcm_suspend*() calls from the runtime PM suspend callback code path, hence the racy OPEN state won't appear while runtime PM. (FWIW, the race window is between snd_pcm_open_substream() and the first power up in azx_pcm_open().) Although the runtime PM issue was already "fixed", the same problem is still present for the system PM, hence this patch is still needed. And for stable trees, this patch alone should suffice for fixing the runtime PM problem, too. Reported-and-tested-by: Jon Hunter <jonathanh@nvidia.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-25ASoC: dapm: Fix NULL pointer dereference in snd_soc_dapm_free_kcontrolPankaj Bharadiya
w_text_param can be NULL and it is being dereferenced without checking. Add the missing sanity check to prevent NULL pointer dereference. Signed-off-by: Pankaj Bharadiya <pankaj.laxminarayan.bharadiya@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-03-25ASoC: intel: Fix crash at suspend/resume after failed codec registrationGuenter Roeck
If codec registration fails after the ASoC Intel SST driver has been probed, the kernel will Oops and crash at suspend/resume. general protection fault: 0000 [#1] PREEMPT SMP KASAN PTI CPU: 1 PID: 2811 Comm: cat Tainted: G W 4.19.30 #15 Hardware name: GOOGLE Clapper, BIOS Google_Clapper.5216.199.7 08/22/2014 RIP: 0010:snd_soc_suspend+0x5a/0xd21 Code: 03 80 3c 10 00 49 89 d7 74 0b 48 89 df e8 71 72 c4 fe 4c 89 fa 48 8b 03 48 89 45 d0 48 8d 98 a0 01 00 00 48 89 d8 48 c1 e8 03 <8a> 04 10 84 c0 0f 85 85 0c 00 00 80 3b 00 0f 84 6b 0c 00 00 48 8b RSP: 0018:ffff888035407750 EFLAGS: 00010202 RAX: 0000000000000034 RBX: 00000000000001a0 RCX: 0000000000000000 RDX: dffffc0000000000 RSI: 0000000000000008 RDI: ffff88805c417098 RBP: ffff8880354077b0 R08: dffffc0000000000 R09: ffffed100b975718 R10: 0000000000000001 R11: ffffffff949ea4a3 R12: 1ffff1100b975746 R13: dffffc0000000000 R14: ffff88805cba4588 R15: dffffc0000000000 FS: 0000794a78e91b80(0000) GS:ffff888068d00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007bd5283ccf58 CR3: 000000004b7aa000 CR4: 00000000001006e0 Call Trace: ? dpm_complete+0x67b/0x67b ? i915_gem_suspend+0x14d/0x1ad sst_soc_prepare+0x91/0x1dd ? sst_be_hw_params+0x7e/0x7e dpm_prepare+0x39a/0x88b dpm_suspend_start+0x13/0x9d suspend_devices_and_enter+0x18f/0xbd7 ? arch_suspend_enable_irqs+0x11/0x11 ? printk+0xd9/0x12d ? lock_release+0x95f/0x95f ? log_buf_vmcoreinfo_setup+0x131/0x131 ? rcu_read_lock_sched_held+0x140/0x22a ? __bpf_trace_rcu_utilization+0xa/0xa ? __pm_pr_dbg+0x186/0x190 ? pm_notifier_call_chain+0x39/0x39 ? suspend_test+0x9d/0x9d pm_suspend+0x2f4/0x728 ? trace_suspend_resume+0x3da/0x3da ? lock_release+0x95f/0x95f ? kernfs_fop_write+0x19f/0x32d state_store+0xd8/0x147 ? sysfs_kf_read+0x155/0x155 kernfs_fop_write+0x23e/0x32d __vfs_write+0x108/0x608 ? vfs_read+0x2e9/0x2e9 ? rcu_read_lock_sched_held+0x140/0x22a ? __bpf_trace_rcu_utilization+0xa/0xa ? debug_smp_processor_id+0x10/0x10 ? selinux_file_permission+0x1c5/0x3c8 ? rcu_sync_lockdep_assert+0x6a/0xad ? __sb_start_write+0x129/0x2ac vfs_write+0x1aa/0x434 ksys_write+0xfe/0x1be ? __ia32_sys_read+0x82/0x82 do_syscall_64+0xcd/0x120 entry_SYSCALL_64_after_hwframe+0x49/0xbe In the observed situation, the problem is seen because the codec driver failed to probe due to a hardware problem. max98090 i2c-193C9890:00: Failed to read device revision: -1 max98090 i2c-193C9890:00: ASoC: failed to probe component -1 cht-bsw-max98090 cht-bsw-max98090: ASoC: failed to instantiate card -1 cht-bsw-max98090 cht-bsw-max98090: snd_soc_register_card failed -1 cht-bsw-max98090: probe of cht-bsw-max98090 failed with error -1 The problem is similar to the problem solved with commit 2fc995a87f2e ("ASoC: intel: Fix crash at suspend/resume without card registration"), but codec registration fails at a later point. At that time, the pointer checked with the above mentioned commit is already set, but it is not cleared if the device is subsequently removed. Adding a remove function to clear the pointer fixes the problem. Cc: stable@vger.kernel.org Cc: Jarkko Nikula <jarkko.nikula@linux.intel.com> Cc: Curtis Malainey <cujomalainey@chromium.org> Signed-off-by: Guenter Roeck <linux@roeck-us.net> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-03-22ALSA: hda/ca0132 - Simplify alt firmware loading codeTakashi Iwai
ca0132 codec driver loads the firmware selectively depending on the model in addition to the fallback of the default firmware. The code works good, but a minor problem is that the current code seems confusing for Clang where it spews a warning about uninitialized variable. This patch simplifies the code flow for such a false-positive warning. After this refactoring, the ca0132_spec.alt_firmware_present field is no longer used, hence it's eliminated as well. Reported-and-tested-by: Arnd Bergmann <arnd@arndb.de> Reviewed-by: Nathan Chancellor <natechancellor@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-22ALSA: pcm: Fix possible OOB access in PCM oss pluginsTakashi Iwai
The PCM OSS emulation converts and transfers the data on the fly via "plugins". The data is converted over the dynamically allocated buffer for each plugin, and recently syzkaller caught OOB in this flow. Although the bisection by syzbot pointed out to the commit 65766ee0bf7f ("ALSA: oss: Use kvzalloc() for local buffer allocations"), this is merely a commit to replace vmalloc() with kvmalloc(), hence it can't be the cause. The further debug action revealed that this happens in the case where a slave PCM doesn't support only the stereo channels while the OSS stream is set up for a mono channel. Below is a brief explanation: At each OSS parameter change, the driver sets up the PCM hw_params again in snd_pcm_oss_change_params_lock(). This is also the place where plugins are created and local buffers are allocated. The problem is that the plugins are created before the final hw_params is determined. Namely, two snd_pcm_hw_param_near() calls for setting the period size and periods may influence on the final result of channels, rates, etc, too, while the current code has already created plugins beforehand with the premature values. So, the plugin believes that channels=1, while the actual I/O is with channels=2, which makes the driver reading/writing over the allocated buffer size. The fix is simply to move the plugin allocation code after the final hw_params call. Reported-by: syzbot+d4503ae45b65c5bc1194@syzkaller.appspotmail.com Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-22ALSA: hda/realtek: Enable headset MIC of ASUS X430UN and X512DK with ALC256Jian-Hong Pan
The ASUS X430UN and X512DK with ALC256 cannot detect the headset MIC until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Signed-off-by: Daniel Drake <drake@endlessm.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-22ALSA: hda/realtek: Enable headset mic of ASUS P5440FF with ALC256Chris Chiu
The ASUS laptop P5440FF with ALC256 can't detect the headset microphone until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Chris Chiu <chiu@endlessm.com> Signed-off-by: Daniel Drake <drake@endlessm.com> Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-22ALSA: hda/realtek: Enable ASUS X441MB and X705FD headset MIC with ALC256Jian-Hong Pan
The ASUS laptop X441MB and X705FD with ALC256 cannot detect the headset MIC until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Chris Chiu <chiu@endlessm.com> Signed-off-by: Daniel Drake <drake@endlessm.com> Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-21ASoC: fsl_esai: fix channel swap issue when stream startsS.j. Wang
There is very low possibility ( < 0.1% ) that channel swap happened in beginning when multi output/input pin is enabled. The issue is that hardware can't send data to correct pin in the beginning with the normal enable flow. This is hardware issue, but there is no errata, the workaround flow is that: Each time playback/recording, firstly clear the xSMA/xSMB, then enable TE/RE, then enable xSMB and xSMA (xSMB must be enabled before xSMA). Which is to use the xSMA as the trigger start register, previously the xCR_TE or xCR_RE is the bit for starting. Fixes commit 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Cc: <stable@vger.kernel.org> Reviewed-by: Fabio Estevam <festevam@gmail.com> Acked-by: Nicolin Chen <nicoleotsuka@gmail.com> Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-03-21ASoC: fsl_asrc: add constraint for the asrc of older versionS.j. Wang
There is a constraint for the channel number setting on the asrc of older version (e.g. imx35), the channel number should be even, odd number isn't valid. So add this constraint when the asrc of older version is used. Acked-by: Nicolin Chen <nicoleotsuka@gmail.com> Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-03-21ASoC: cs4270: Set auto-increment bit for register writesDaniel Mack
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: Mark Brown <broonie@kernel.org>
2019-03-21ALSA: hda/realtek - Add support for Acer Aspire E5-523G/ES1-432 headset micChris Chiu
The Acer laptop Aspire E5-523G and ES1-432 with ALC255 can't detect the headset microphone until ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk applied. Signed-off-by: Chris Chiu <chiu@endlessm.com> Signed-off-by: Daniel Drake <drake@endlessm.com> Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-21ALSA: hda/realtek: Enable headset MIC of Acer Aspire Z24-890 with ALC286Jian-Hong Pan
The Acer Aspire Z24-890 cannot detect the headset MIC until ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk applied. Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Signed-off-by: Daniel Drake <drake@endlessm.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-21ALSA: seq: oss: Fix Spectre v1 vulnerabilityGustavo A. R. Silva
dev is indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. This issue was detected with the help of Smatch: sound/core/seq/oss/seq_oss_synth.c:626 snd_seq_oss_synth_make_info() warn: potential spectre issue 'dp->synths' [w] (local cap) Fix this by sanitizing dev before using it to index dp->synths. Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. [1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/ Cc: stable@vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-21ALSA: rawmidi: Fix potential Spectre v1 vulnerabilityGustavo A. R. Silva
info->stream is indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. This issue was detected with the help of Smatch: sound/core/rawmidi.c:604 __snd_rawmidi_info_select() warn: potential spectre issue 'rmidi->streams' [r] (local cap) Fix this by sanitizing info->stream before using it to index rmidi->streams. Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. [1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/ Cc: stable@vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-21ALSA: hda/realtek: Enable headset MIC of Acer AIO with ALC286Jian-Hong Pan
Some Acer AIO desktops like Veriton Z6860G, Z4860G and Z4660G cannot record sound from headset MIC. This patch adds the ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk to fix this issue. Fixes: 9f8aefed9623 ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4660G") Fixes: b72f936f6b32 ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4860G/Z6860G") Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Reviewed-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-03-20ASoC: stm32: dfsdm: fix debugfs warnings on entry creationOlivier Moysan
Register platform component with a prefix, to avoid warnings on debugfs entries creation, due to component name redundancy. Signed-off-by: Olivier Moysan <olivier.moysan@st.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-03-20ASoC: stm32: dfsdm: manage multiple prepareOlivier Moysan
The DFSDM must be stopped when a new setting is applied. restart systematically DFSDM on multiple prepare calls, to apply changes. Signed-off-by: Olivier Moysan <olivier.moysan@st.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-03-19ASoC: wm_adsp: Shutdown any compressed streams on DSP watchdog timeoutCharles Keepax
If a watchdog timeout is received from the DSP it is safe to assume the DSP is not functioning anymore and as such any active compressed streams should be put into an error state. Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-03-19ASoC: wm_adsp: Add locking to wm_adsp2_bus_errorCharles Keepax
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: Mark Brown <broonie@kernel.org>
2019-03-19ASoC: wm_adsp: Correct error messages in wm_adsp_buffer_get_errorCharles Keepax
During recent logging improvements it seems two error messages lost their updates during patch application/rebasing. Add these back in. Fixes: 0d3fba3e7a56 ("ASoC: wm_adsp: Improve logging messages") Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>