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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 <[email protected]>
Reported-by: Dean Wallace <[email protected]>
Signed-off-by: Hans de Goede <[email protected]>
Acked-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Signed-off-by: Daniel Drake <[email protected]>
Reviewed-by: Kailang Yang <[email protected]>
Cc: <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Pull yet another ALSA core timer fixes and cleanups.
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Acked-by: Michal Hocko <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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: [email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
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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 <[email protected]>
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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 <[email protected]>
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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 <[email protected]>
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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 <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Acked-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Reviewed-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Cc: <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Acked-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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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: [email protected]
Cc: Jarkko Nikula <[email protected]>
Cc: Curtis Malainey <[email protected]>
Signed-off-by: Guenter Roeck <[email protected]>
Acked-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Reviewed-by: Nathan Chancellor <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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: [email protected]
Cc: <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Signed-off-by: Daniel Drake <[email protected]>
Cc: <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Signed-off-by: Daniel Drake <[email protected]>
Signed-off-by: Jian-Hong Pan <[email protected]>
Cc: <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Signed-off-by: Daniel Drake <[email protected]>
Signed-off-by: Jian-Hong Pan <[email protected]>
Cc: <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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: <[email protected]>
Reviewed-by: Fabio Estevam <[email protected]>
Acked-by: Nicolin Chen <[email protected]>
Signed-off-by: Shengjiu Wang <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Signed-off-by: Shengjiu Wang <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Signed-off-by: Daniel Drake <[email protected]>
Signed-off-by: Jian-Hong Pan <[email protected]>
Cc: <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Signed-off-by: Daniel Drake <[email protected]>
Cc: <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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/[email protected]/
Cc: [email protected]
Signed-off-by: Gustavo A. R. Silva <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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/[email protected]/
Cc: [email protected]
Signed-off-by: Gustavo A. R. Silva <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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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 <[email protected]>
Reviewed-by: Kailang Yang <[email protected]>
Cc: <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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Register platform component with a prefix, to avoid warnings
on debugfs entries creation, due to component name
redundancy.
Signed-off-by: Olivier Moysan <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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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 <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Previously support was added to allow streams to be stopped and
started again without the DSP being power cycled and this was done
by clearing the buffer state in trigger start. Another supported
use-case is using the DSP for a trigger event then opening the
compressed stream later to receive the audio, unfortunately clearing
the buffer state in trigger start destroys the data received
from such a trigger. Correct this issue by moving the call to
wm_adsp_buffer_clear to be in trigger stop instead.
Fixes: 61fc060c40e6 ("ASoC: wm_adsp: Support streams which can start/stop with DSP active")
Signed-off-by: Charles Keepax <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Recently we found the audio jack detection stop working after suspend
on many machines with Realtek codec. Sometimes the audio selection
dialogue didn't show up after users plugged headhphone/headset into
the headset jack, sometimes after uses plugged headphone/headset, then
click the sound icon on the upper-right corner of gnome-desktop, it
also showed the speaker rather than the headphone.
The root cause is that before suspend, the codec already call the
runtime_suspend since this codec is not used by any apps, then in
resume, it will not call runtime_resume for this codec. But for some
realtek codec (so far, alc236, alc255 and alc891) with the specific
BIOS, if it doesn't run runtime_resume after suspend, all codec
functions including jack detection stop working anymore.
This problem existed for a long time, but it was not exposed, that is
because when problem happens, if users play sound or open
sound-setting to check audio device, this will trigger calling to
runtime_resume (via snd_hda_power_up), then the codec starts working
again before users notice this problem.
Since we don't know how many codec and BIOS combinations have this
problem, to fix it, let the driver call runtime_resume for all codecs
in pm_resume, maybe for some codecs, this is not needed, but it is
harmless. After a codec is runtime resumed, if it is not used by any
apps, it will be runtime suspended soon and furthermore we don't run
suspend frequently, this change will not add much power consumption.
Fixes: cc72da7d4d06 ("ALSA: hda - Use standard runtime PM for codec power-save control")
Signed-off-by: Hui Wang <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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The commit 3baffc4a84d7 (ALSA: hda/intel: Refactoring PM code) changed
the behaviour of azx_resume(), it triggers the jackpoll_work after
applying this commit.
This change introduced a new issue, all codecs are runtime active
after S3, and will not call runtime_suspend() automatically.
The root cause is the jackpoll_work calls snd_hda_power_up/down_pm,
and it calls up_pm before snd_hdac_enter_pm is called, while calls
the down_pm in the middle of enter_pm and leave_pm is called. This
makes the dev->power.usage_count unbalanced after S3.
To fix it, let azx_resume() don't trigger jackpoll_work as before
it did.
Fixes: 3baffc4a84d7 ("ALSA: hda/intel: Refactoring PM code")
Signed-off-by: Hui Wang <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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If ASRC turns on, HW will use clk_dac as the reference clock
whether recording or playback.
Both of clk_dac and clk_adc should set proper clock while using ASRC.
Signed-off-by: Shuming Fan <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The jack type detection needs the main bias power of analog.
The modification makes sure the main bias power on/off while jack plug/unplug.
Signed-off-by: Shuming Fan <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The IRQ function may not work when system suspend.
We remove snd_soc_dapm_force_enable_pin function call to
make sure the bias off when idle and run into suspend/resume function.
Signed-off-by: Shuming Fan <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Skip for i2s5 in mck_disable which is also bypassed in mck_enable.
Signed-off-by: Tzung-Bi Shih <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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In kernel API of Linux FireWire subsystem, handlers of isochronous
receive (IR) context can get context headers as an argument of
callback. When 4 byte header is used, the context header includes
isochronous packet header for each packet. When 8 byte header is
used, it includes isochronous cycle as well.
ALSA IEC 61883-1/6 engine uses 4 byte header, and computes isochronous
cycle from the cycle of interrupt. The usage of 8 byte header can
obsolete the computation.
Furthermore, this change works well for a case that a series of
packet in one interrupt includes skipped isochronous cycle,
This commit uses 8 byte header to handle isochronous cycle.
Signed-off-by: Takashi Sakamoto <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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This commit adds support for MOTU 8pre FireWire, which was shipped 2007
and nowadays already discontinued. Userspace applications can transmit
and receive PCM frames and MIDI messages for this model via ALSA PCM
interface and RawMidi/Sequencer interfaces.
Like the other models of MOTU FireWire series, this model has many
quirks in its CIP.
At first, data channels for two pairs of optical interfaces. At lower
sampling transmission frequency, i.e. 44.1 and 48.0 kHz, one pair is
available for ADAT data, thus 8 data chunks are transferred by CIP.
At middle sampling transmission frequency, i.e. 88.2 and 96.0 kHz,
two pairs are available to keep 8 chunks for ADAT data, thus CIP
still includes 8 data chunks.
Apart from data chunks for optical interface, CIP includes fixed number
of data chunks. In tx stream, two chunks for status message, eight
chunks for samples from analog 1-8 input, two chunks for mix-return.
In rx stream, two chunks for control message, two chunks for main 1-2
output, two chunks for phone 1-2 output, two chunks for dummy 1-2.
CIP header in tx stream includes quirks for its dbs and dbc fields.
The value of dbs field is fixed to 0x13, against its actual size.
The value of dbc field is firstly updated to 0x07 from zero, then
it's incremented continuously according to actual number of data h
blocks.
Finally, the model has own bits to disable frame fetch.
This commit uses several options to absorb the above quirks.
$ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom
ROM header and bus information block
-----------------------------------------------------------------
400 0410b57d bus_info_length 4, crc_length 16, crc 46461
404 31333934 bus_name "1394"
408 20001000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 0, max_rec 1 (4)
40c 0001f200 company_id 0001f2 |
410 00083dfb device_id 0000083dfb | EUI-64 0001f20000083dfb
root directory
-----------------------------------------------------------------
414 0004c65c directory_length 4, crc 50780
418 030001f2 vendor
41c 0c0083c0 node capabilities per IEEE 1394
420 8d000006 --> eui-64 leaf at 438
424 d1000001 --> unit directory at 428
unit directory at 428
-----------------------------------------------------------------
428 0003991c directory_length 3, crc 39196
42c 120001f2 specifier id
430 1300000f version
434 17103800 model
eui-64 leaf at 438
-----------------------------------------------------------------
438 00022681 leaf_length 2, crc 9857
43c 0001f200 company_id 0001f2 |
440 00083dfb device_id 0000083dfb | EUI-64 0001f20000083dfb
Signed-off-by: Takashi Sakamoto <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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Back-merge the current devel branch for further development.
Signed-off-by: Takashi Iwai <[email protected]>
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The function snd_opl3_drum_switch declaration in the header file
has the order of the two arguments on_off and vel swapped when
compared to the definition arguments of vel and on_off. Fix this
by swapping them around to match the definition.
This error predates the git history, so no idea when this error
was introduced.
Signed-off-by: Colin Ian King <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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